valgrind --suppressions=./contrib/valgrind.supp --log-fd=9 asterisk -vvvvcg 9>valgrind.txt Asterisk 13.14.0, Copyright (C) 1999 - 2014, 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. ========================================================================= [ Initializing Custom Configuration Options ] == Parsing '/etc/asterisk/extconfig.conf': Found == Parsing '/etc/asterisk/asterisk.conf': Found == Manager registered action DBGet == Manager registered action DBPut == Manager registered action DBDel == Manager registered action DBDelTree PBX UUID: fe0fbd83-b10e-4c40-b725-3ddb823d519e == Registered 'audio' codec 'g723' at sample rate '8000' with id '1' == Created cached format with name 'g723' == Registered 'audio' codec 'ulaw' at sample rate '8000' with id '2' == Created cached format with name 'ulaw' == Registered 'audio' codec 'alaw' at sample rate '8000' with id '3' == Created cached format with name 'alaw' == Registered 'audio' codec 'gsm' at sample rate '8000' with id '4' == Created cached format with name 'gsm' == Registered 'audio' codec 'g726' at sample rate '8000' with id '5' == Created cached format with name 'g726' == Registered 'audio' codec 'g726aal2' at sample rate '8000' with id '6' == Created cached format with name 'g726aal2' == Registered 'audio' codec 'adpcm' at sample rate '8000' with id '7' == Created cached format with name 'adpcm' == Registered 'audio' codec 'slin' at sample rate '8000' with id '8' == Created cached format with name 'slin' == Registered 'audio' codec 'slin' at sample rate '12000' with id '9' == Created cached format with name 'slin12' == Registered 'audio' codec 'slin' at sample rate '16000' with id '10' == Created cached format with name 'slin16' == Registered 'audio' codec 'slin' at sample rate '24000' with id '11' == Created cached format with name 'slin24' == Registered 'audio' codec 'slin' at sample rate '32000' with id '12' == Created cached format with name 'slin32' == Registered 'audio' codec 'slin' at sample rate '44100' with id '13' == Created cached format with name 'slin44' == Registered 'audio' codec 'slin' at sample rate '48000' with id '14' == Created cached format with name 'slin48' == Registered 'audio' codec 'slin' at sample rate '96000' with id '15' == Created cached format with name 'slin96' == Registered 'audio' codec 'slin' at sample rate '192000' with id '16' == Created cached format with name 'slin192' == Registered 'audio' codec 'lpc10' at sample rate '8000' with id '17' == Created cached format with name 'lpc10' == Registered 'audio' codec 'g729' at sample rate '8000' with id '18' == Created cached format with name 'g729' == Registered 'audio' codec 'speex' at sample rate '8000' with id '19' == Created cached format with name 'speex' == Registered 'audio' codec 'speex' at sample rate '16000' with id '20' == Created cached format with name 'speex16' == Registered 'audio' codec 'speex' at sample rate '32000' with id '21' == Created cached format with name 'speex32' == Registered 'audio' codec 'ilbc' at sample rate '8000' with id '22' == Created cached format with name 'ilbc' == Registered 'audio' codec 'g722' at sample rate '16000' with id '23' == Created cached format with name 'g722' == Registered 'audio' codec 'siren7' at sample rate '16000' with id '24' == Created cached format with name 'siren7' == Registered 'audio' codec 'siren14' at sample rate '32000' with id '25' == Created cached format with name 'siren14' == Registered 'audio' codec 'testlaw' at sample rate '8000' with id '26' == Created cached format with name 'testlaw' == Registered 'audio' codec 'g719' at sample rate '48000' with id '27' == Created cached format with name 'g719' == Registered 'audio' codec 'opus' at sample rate '48000' with id '28' == Created cached format with name 'opus' == Registered 'image' codec 'jpeg' at sample rate '0' with id '29' == Created cached format with name 'jpeg' == Registered 'image' codec 'png' at sample rate '0' with id '30' == Created cached format with name 'png' == Registered 'video' codec 'h261' at sample rate '1000' with id '31' == Created cached format with name 'h261' == Registered 'video' codec 'h263' at sample rate '1000' with id '32' == Created cached format with name 'h263' == Registered 'video' codec 'h263p' at sample rate '1000' with id '33' == Created cached format with name 'h263p' == Registered 'video' codec 'h264' at sample rate '1000' with id '34' == Created cached format with name 'h264' == Registered 'video' codec 'mpeg4' at sample rate '1000' with id '35' == Created cached format with name 'mpeg4' == Registered 'video' codec 'vp8' at sample rate '1000' with id '36' == Created cached format with name 'vp8' == Registered 'text' codec 'red' at sample rate '0' with id '37' == Created cached format with name 'red' == Registered 'text' codec 't140' at sample rate '0' with id '38' == Created cached format with name 't140' == Registered 'audio' codec 'none' at sample rate '8000' with id '39' == Created cached format with name 'none' == Registered 'audio' codec 'silk' at sample rate '8000' with id '40' == Created cached format with name 'silk8' == Registered 'audio' codec 'silk' at sample rate '12000' with id '41' == Created cached format with name 'silk12' == Registered 'audio' codec 'silk' at sample rate '16000' with id '42' == Created cached format with name 'silk16' == Registered 'audio' codec 'silk' at sample rate '24000' with id '43' == Created cached format with name 'silk24' == Sorcery registered wizard 'bucket' == Sorcery registered wizard 'bucket_file' == Parsing '/etc/asterisk/sorcery.conf': Found == Parsing '/etc/asterisk/stasis.conf': Found == Parsing '/etc/asterisk/logger.conf': Found == Parsing '/etc/asterisk/pjproject.conf': Found == Message handler 'dialplan' registered. == Registered custom function 'MESSAGE' == Registered custom function 'MESSAGE_DATA' == Registered application 'MessageSend' == Manager registered action MessageSend == Manager registered action DataGet == Registered channel type 'Surrogate' (Surrogate channel used to pull channel from an application) == Parsing '/etc/asterisk/codecs.conf': Found == Manager registered action BridgeTechnologyList == Manager registered action BridgeTechnologySuspend == Manager registered action BridgeTechnologyUnsuspend Asterisk Dynamic Loader Starting: == Parsing '/etc/asterisk/modules.conf': Found == Parsing '/etc/asterisk/features.conf': Found == Registered custom function 'FEATURE' == Registered custom function 'FEATUREMAP' == Registered application 'Bridge' == Manager registered action Bridge == Parsing '/etc/asterisk/dnsmgr.conf': Found == Parsing '/etc/asterisk/acl.conf': Found == Parsing '/etc/asterisk/http.conf': Found == 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 'us' == Parsing '/etc/asterisk/cdr.conf': Found [Feb 20 15:22:03] NOTICE[32481]: cdr.c:4180 cdr_toggle_runtime_options: CDR simple logging enabled. [Feb 20 15:22:03] NOTICE[32481]: cdr.c:4180 cdr_toggle_runtime_options: CDR simple logging enabled. == Parsing '/etc/asterisk/dsp.conf': Found == Parsing '/etc/asterisk/udptl.conf': Found Asterisk PBX Core Initializing == Registering builtin functions: == Registered custom function 'EXCEPTION' == Registered custom function 'TESTTIME' == Manager registered action ShowDialPlan == Manager registered action ExtensionStateList == Registered application 'Answer' == Registered application 'BackGround' == Registered application 'Busy' == Registered application 'Congestion' == Registered application 'ExecIfTime' == Registered application 'Goto' == Registered application 'GotoIf' == Registered application 'GotoIfTime' == Registered application 'ImportVar' == Registered application 'Hangup' == Registered application 'Incomplete' == Registered application 'NoOp' == Registered application 'Proceeding' == Registered application 'Progress' == Registered application 'RaiseException' == Registered application 'Ringing' == Registered application 'SayAlpha' == Registered application 'SayAlphaCase' == Registered application 'SayDigits' == Registered application 'SayNumber' == Registered application 'SayPhonetic' == Registered application 'SetAMAFlags' == Registered application 'Wait' == Registered application 'WaitExten' == Registered application 'Set' == Registered application 'MSet' == Registered channel type 'Local' (Local Proxy Channel Driver) == Manager registered action LocalOptimizeAway == Parsing '/etc/asterisk/cel.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 PresenceState == 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 LoggerRotate == Manager registered action CoreShowChannels == Manager registered action ModuleLoad == Manager registered action ModuleCheck == Manager registered action AOCMessage == Manager registered action Filter == Manager registered action BlindTransfer == Registered custom function 'AMI_CLIENT' == Parsing '/etc/asterisk/manager.conf': Found == Parsing '/etc/asterisk/users.conf': Found == Parsing '/etc/asterisk/enum.conf': Found == Registered application 'CallCompletionRequest' == Registered application 'CallCompletionCancel' == Parsing '/etc/asterisk/ccss.conf': Found == Parsing '/etc/asterisk/ccss.conf': Found Asterisk Dynamic Loader Starting: == Parsing '/etc/asterisk/modules.conf': Found [Feb 20 15:22:08] NOTICE[32481]: loader.c:1446 load_modules: 316 modules will be loaded. [Feb 20 15:22:08] NOTICE[32481]: loader.c:1446 load_modules: 316 modules will be loaded. Loading res_statsd.so. == Parsing '/etc/asterisk/statsd.conf': Found == res_statsd.so => (Statsd client support) Loading res_odbc.so. == Parsing '/etc/asterisk/res_odbc.conf': Found [Feb 20 15:22:12] NOTICE[32481]: res_odbc.c:1089 load_module: res_odbc loaded. [Feb 20 15:22:12] NOTICE[32481]: res_odbc.c:1089 load_module: res_odbc loaded. == res_odbc.so => (ODBC resource) Loading res_odbc_transaction.so. == Registered application 'ODBC_Commit' == Registered application 'ODBC_Rollback' == Registered custom function 'ODBC' == res_odbc_transaction.so => (ODBC transaction resource) Loading res_sorcery_config.so. == Sorcery registered wizard 'config' == res_sorcery_config.so => (Sorcery Configuration File Object Wizard) Loading res_sorcery_memory_cache.so. == Sorcery registered wizard 'memory_cache' == Manager registered action SorceryMemoryCacheExpireObject == Manager registered action SorceryMemoryCacheExpire == Manager registered action SorceryMemoryCacheStaleObject == Manager registered action SorceryMemoryCacheStale == Manager registered action SorceryMemoryCachePopulate == res_sorcery_memory_cache.so => (Sorcery Memory Cache Object Wizard) Loading res_sorcery_realtime.so. == Sorcery registered wizard 'realtime' == res_sorcery_realtime.so => (Sorcery Realtime Object Wizard) Loading res_pjsip_config_wizard.so. == res_pjsip_config_wizard.so => (PJSIP Config Wizard) Loading res_sorcery_astdb.so. == Sorcery registered wizard 'astdb' == res_sorcery_astdb.so => (Sorcery Astdb Object Wizard) Loading res_sorcery_memory.so. == Sorcery registered wizard 'memory' == res_sorcery_memory.so => (Sorcery In-Memory Object Wizard) Loading res_format_attr_opus.so. == Registered format interface for codec 'opus' == res_format_attr_opus.so => (Opus Format Attribute Module) Loading codec_opus.so. == Parsing '/etc/asterisk/sorcery.conf': Found == Parsing '/etc/asterisk/codecs.conf': Found == Updated cached format with name 'opus' == Registered translator 'lintoopus' from codec slin to opus, table cost, 600000, computational cost 999999 == Registered translator 'opustolin' from codec opus to slin, table cost, 900000, computational cost 999999 == codec_opus.so => (OPUS Coder/Decoder) Loading res_pjproject.so. == Parsing '/etc/asterisk/sorcery.conf': Found == Parsing '/etc/asterisk/pjproject.conf': Found == res_pjproject.so => (PJPROJECT Log and Utility Support) Loading res_pjsip.so. == Parsing '/etc/asterisk/sorcery.conf': Found == Parsing '/etc/asterisk/pjsip.conf': Found -- Local IPv4 address determined to be: 192.168.41.78 -- Local IPv6 address determined to be: ::1 == Manager registered action PJSIPShowEndpoints == Manager registered action PJSIPShowEndpoint == Parsing '/etc/asterisk/pjsip.conf': Found == Parsing '/etc/asterisk/pjsip_wizard.conf': Found == Parsing '/etc/asterisk/pjsip.conf': Found == Parsing '/etc/asterisk/pjsip_wizard.conf': Found == Parsing '/etc/asterisk/pjsip.conf': Found == Parsing '/etc/asterisk/pjsip.conf': Found == Parsing '/etc/asterisk/pjsip_wizard.conf': Found == Parsing '/etc/asterisk/pjsip.conf': Found == Parsing '/etc/asterisk/pjsip.conf': Found == Parsing '/etc/asterisk/pjsip.conf': Found == Manager registered action PJSIPQualify == res_pjsip.so => (Basic SIP resource) Loading res_crypto.so. == res_crypto.so => (Cryptographic Digital Signatures) Loading res_smdi.so. == Parsing '/etc/asterisk/smdi.conf': Found == Registered custom function 'SMDI_MSG_RETRIEVE' == Registered custom function 'SMDI_MSG' == res_smdi.so => (Simplified Message Desk Interface (SMDI) Resource) Loading res_rtp_multicast.so. == Registered RTP engine 'multicast' == res_rtp_multicast.so => (Multicast RTP Engine) Loading res_monitor.so. == 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) Loading res_stun_monitor.so. == Parsing '/etc/asterisk/res_stun_monitor.conf': Found == res_stun_monitor.so => (STUN Network Monitor) Loading res_pjsip_outbound_publish.so. == Parsing '/etc/asterisk/sorcery.conf': Found == Parsing '/etc/asterisk/pjsip.conf': Found == res_pjsip_outbound_publish.so => (PJSIP Outbound Publish Support) Loading res_xmpp.so. == Parsing '/etc/asterisk/xmpp.conf': Found == Manager registered action JabberSend == Registered application 'JabberSend' == Registered application 'JabberSendGroup' == Registered application 'JabberStatus' == Registered application 'JabberJoin' == Registered application 'JabberLeave' == Registered custom function 'JABBER_STATUS' == Registered custom function 'JABBER_RECEIVE' -- Message technology 'xmpp' registered. == res_xmpp.so => (Asterisk XMPP Interface) Loading res_pjsip_pubsub.so. == Parsing '/etc/asterisk/sorcery.conf': Found == Parsing '/etc/asterisk/pjsip.conf': Found == Parsing '/etc/asterisk/pjsip.conf': Found == Manager registered action PJSIPShowSubscriptionsInbound == Manager registered action PJSIPShowSubscriptionsOutbound == Manager registered action PJSIPShowResourceLists == res_pjsip_pubsub.so => (PJSIP event resource) Loading res_srtp.so. == res_srtp.so => (Secure RTP (SRTP)) Loading res_http_websocket.so. == WebSocket registered sub-protocol 'echo' == res_http_websocket.so => (HTTP WebSocket Support) Loading res_phoneprov.so. == Parsing '/etc/asterisk/phoneprov.conf': Found == Parsing '/etc/asterisk/phoneprov.conf': Found [Feb 20 15:22:24] WARNING[32481]: res_phoneprov.c:1231 get_defaults: Unable to find a valid server address or name. [Feb 20 15:22:24] WARNING[32481]: res_phoneprov.c:1231 get_defaults: Unable to find a valid server address or name. == Parsing '/etc/asterisk/sip.conf': Found == Parsing '/etc/asterisk/users.conf': Found == Parsing '/etc/asterisk/users.conf': Found == Registered custom function 'PP_EACH_USER' == Registered custom function 'PP_EACH_EXTENSION' == res_phoneprov.so => (HTTP Phone Provisioning) Loading res_hep.so. == Parsing '/etc/asterisk/hep.conf': Found == res_hep.so => (HEPv3 API) Loading res_stasis_recording.so. == res_stasis_recording.so => (Stasis application recording support) Loading res_ari.so. == Parsing '/etc/asterisk/ari.conf': Found [Feb 20 15:22:24] ERROR[32481]: ari/config.c:312 process_config: No configured users for ARI [Feb 20 15:22:24] ERROR[32481]: ari/config.c:312 process_config: No configured users for ARI == res_ari.so => (Asterisk RESTful Interface) Loading res_fax.so. == Parsing '/etc/asterisk/res_fax.conf': Found == Registered application 'SendFAX' == Registered application 'ReceiveFAX' == Manager registered action FAXSessions == Manager registered action FAXSession == Manager registered action FAXStats == Registered custom function 'FAXOPT' == res_fax.so => (Generic FAX Applications) Loading res_ari_model.so. == res_ari_model.so => (ARI Model validators) Loading res_agi.so. == 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)) Loading res_pjsip_session.so. == res_pjsip_session.so => (PJSIP Session resource) Loading res_speech.so. == res_speech.so => (Generic Speech Recognition API) Loading res_calendar.so. == 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) Loading res_stasis_device_state.so. == res_stasis_device_state.so => (Stasis application device state support) Loading res_stasis_snoop.so. == res_stasis_snoop.so => (Stasis application snoop support) Loading func_periodic_hook.so. -- Registered extension context '__func_periodic_hook_context__'; registrar: func_periodic_hook -- Added extension 'hook' priority 1 (CID match '') to __func_periodic_hook_context__ -- Added extension 'hook' priority 2 (CID match '') to __func_periodic_hook_context__ -- Added extension 'hook' priority 3 (CID match '') to __func_periodic_hook_context__ -- Added extension 'hook' priority 4 (CID match '') to __func_periodic_hook_context__ -- Added extension 'hook' priority 5 (CID match '') to __func_periodic_hook_context__ -- Added extension 'hook' priority 6 (CID match '') to __func_periodic_hook_context__ -- Added extension 'beep' priority 1 (CID match '') to __func_periodic_hook_context__ -- Added extension 'beep' priority 2 (CID match '') to __func_periodic_hook_context__ == Registered custom function 'PERIODIC_HOOK' == func_periodic_hook.so => (Periodic dialplan hooks.) Loading pbx_lua.so. -- Registered extension context 'default'; registrar: pbx_lua -- Including switch 'Lua/' in context 'default' -- Registered extension context 'public'; registrar: pbx_lua -- Including switch 'Lua/' in context 'public' -- Registered extension context 'demo'; registrar: pbx_lua -- Including switch 'Lua/' in context 'demo' -- Registered extension context 'local'; registrar: pbx_lua -- Including switch 'Lua/' in context 'local' -- Added extension '1000' priority -1 to demo -- Added extension '1001' priority -1 to demo -- Added extension '1234' priority -1 to default -- Registered extension context '__func_periodic_hook_context__'; registrar: func_periodic_hook -- merging incls/swits/igpats from old(__func_periodic_hook_context__) to new(__func_periodic_hook_context__) context, registrar = pbx_lua -- Added extension 'beep' priority 2 (CID match '') to __func_periodic_hook_context__ -- Added extension 'beep' priority 1 (CID match '') to __func_periodic_hook_context__ -- Added extension 'hook' priority 6 (CID match '') to __func_periodic_hook_context__ -- Added extension 'hook' priority 5 (CID match '') to __func_periodic_hook_context__ -- Added extension 'hook' priority 4 (CID match '') to __func_periodic_hook_context__ -- Added extension 'hook' priority 3 (CID match '') to __func_periodic_hook_context__ -- Added extension 'hook' priority 2 (CID match '') to __func_periodic_hook_context__ -- Added extension 'hook' priority 1 (CID match '') to __func_periodic_hook_context__ -- Time to scan old dialplan and merge leftovers back into the new: 0.031306 sec -- Time to restore hints and swap in new dialplan: 0.004289 sec -- Time to delete the old dialplan: 0.006012 sec -- Total time merge_contexts_delete: 0.041607 sec [Feb 20 15:22:25] NOTICE[32481]: pbx_lua.c:1640 load_or_reload_lua_stuff: Lua PBX Switch loaded. [Feb 20 15:22:25] NOTICE[32481]: pbx_lua.c:1640 load_or_reload_lua_stuff: Lua PBX Switch loaded. == pbx_lua.so => (Lua PBX Switch) Loading res_stasis.so. == Message handler 'ari' registered. == res_stasis.so => (Stasis application support) Loading res_ael_share.so. == res_ael_share.so => (share-able code for AEL) Loading res_stasis_playback.so. == res_stasis_playback.so => (Stasis application playback support) Loading res_stasis_answer.so. == res_stasis_answer.so => (Stasis application answer support) Loading res_parking.so. == Parsing '/etc/asterisk/res_parking.conf': Found -- Registered extension context 'parkedcalls'; registrar: res_parking -- Added extension '700' priority 1 to parkedcalls -- Added extension '701' priority 1 to parkedcalls -- Added extension '702' priority 1 to parkedcalls -- Added extension '703' priority 1 to parkedcalls -- Added extension '704' priority 1 to parkedcalls -- Added extension '705' priority 1 to parkedcalls -- Added extension '706' priority 1 to parkedcalls -- Added extension '707' priority 1 to parkedcalls -- Added extension '708' priority 1 to parkedcalls -- Added extension '709' priority 1 to parkedcalls -- Added extension '710' priority 1 to parkedcalls -- Added extension '711' priority 1 to parkedcalls -- Added extension '712' priority 1 to parkedcalls -- Added extension '713' priority 1 to parkedcalls -- Added extension '714' priority 1 to parkedcalls -- Added extension '715' priority 1 to parkedcalls -- Added extension '716' priority 1 to parkedcalls -- Added extension '717' priority 1 to parkedcalls -- Added extension '718' priority 1 to parkedcalls -- Added extension '719' priority 1 to parkedcalls -- Added extension '720' priority 1 to parkedcalls == Registered application 'Park' == Registered application 'ParkedCall' == Registered application 'ParkAndAnnounce' == Manager registered action Parkinglots == Manager registered action ParkedCalls == Manager registered action Park == res_parking.so => (Call Parking Resource) Loading res_curl.so. == res_curl.so => (cURL Resource Module) Loading func_curl.so. == Registered custom function 'CURL' == Registered custom function 'CURLOPT' == func_curl.so => (Load external URL) Loading res_config_pgsql.so. == Parsing '/etc/asterisk/res_pgsql.conf': Found [Feb 20 15:22:31] ERROR[32481]: res_config_pgsql.c:1540 pgsql_reconnect: PostgreSQL RealTime: Failed to connect database asterisk on 127.0.0.1: [Feb 20 15:22:31] ERROR[32481]: res_config_pgsql.c:1540 pgsql_reconnect: PostgreSQL RealTime: Failed to connect database asterisk on 127.0.0.1: [Feb 20 15:22:31] WARNING[32481]: res_config_pgsql.c:1483 parse_config: PostgreSQL RealTime: Couldn't establish connection. Check debug. [Feb 20 15:22:31] WARNING[32481]: res_config_pgsql.c:1483 parse_config: PostgreSQL RealTime: Couldn't establish connection. Check debug. == PostgreSQL RealTime reloaded. == res_config_pgsql.so => (PostgreSQL RealTime Configuration Driver) Loading res_config_odbc.so. == res_config_odbc.so => (Realtime ODBC configuration) Loading res_config_sqlite3.so. == Parsing '/etc/asterisk/res_config_sqlite3.conf': Found == res_config_sqlite3.so => (SQLite 3 realtime config engine) Loading res_config_ldap.so. == Parsing '/etc/asterisk/res_ldap.conf': Found [Feb 20 15:22:31] NOTICE[32481]: res_config_ldap.c:1710 parse_config: No directory user found, anonymous binding as default. [Feb 20 15:22:31] NOTICE[32481]: res_config_ldap.c:1710 parse_config: No directory user found, anonymous binding as default. [Feb 20 15:22:31] ERROR[32481]: res_config_ldap.c:1736 parse_config: No directory URL or host found. [Feb 20 15:22:31] ERROR[32481]: res_config_ldap.c:1736 parse_config: No directory URL or host found. [Feb 20 15:22:31] ERROR[32481]: res_config_ldap.c:1613 load_module: Cannot load LDAP RealTime driver. [Feb 20 15:22:31] ERROR[32481]: res_config_ldap.c:1613 load_module: Cannot load LDAP RealTime driver. == res_config_ldap.so => (LDAP realtime interface) Loading res_config_sqlite.so. == Parsing '/etc/asterisk/res_config_sqlite.conf': Found == res_config_sqlite.so => (Realtime SQLite configuration) Loading res_config_curl.so. == Parsing '/etc/asterisk/res_curl.conf': Found == res_config_curl.so => (Realtime Curl configuration) Loading res_timing_pthread.so. == res_timing_pthread.so => (pthread Timing Interface) Loading res_timing_timerfd.so. == res_timing_timerfd.so => (Timerfd Timing Interface) Loading res_pjsip_authenticator_digest.so. == res_pjsip_authenticator_digest.so => (PJSIP authentication resource) Loading res_pjsip_transport_management.so. == res_pjsip_transport_management.so => (PJSIP Reliable Transport Management) Loading res_pjsip_endpoint_identifier_user.so. == res_pjsip_endpoint_identifier_user.so => (PJSIP username endpoint identifier) Loading res_pjsip_endpoint_identifier_ip.so. == Parsing '/etc/asterisk/sorcery.conf': Found == Parsing '/etc/asterisk/pjsip.conf': Found == Parsing '/etc/asterisk/pjsip_wizard.conf': Found == res_pjsip_endpoint_identifier_ip.so => (PJSIP IP endpoint identifier) Loading res_pjsip_registrar.so. == Manager registered action PJSIPShowRegistrationsInbound == Manager registered action PJSIPShowRegistrationInboundContactStatuses == res_pjsip_registrar.so => (PJSIP Registrar Support) Loading res_rtp_asterisk.so. == Registered RTP engine 'asterisk' == Parsing '/etc/asterisk/rtp.conf': Found == RTP Allocating from port range 10000 -> 20000 == res_rtp_asterisk.so => (Asterisk RTP Stack) Loading res_format_attr_vp8.so. == Registered format interface for codec 'vp8' == res_format_attr_vp8.so => (VP8 Format Attribute Module) Loading res_format_attr_siren14.so. == Registered format interface for codec 'siren14' == res_format_attr_siren14.so => (Siren14 Format Attribute Module) Loading res_pjsip_mwi_body_generator.so. == res_pjsip_mwi_body_generator.so => (PJSIP MWI resource) Loading res_format_attr_siren7.so. == Registered format interface for codec 'siren7' == res_format_attr_siren7.so => (Siren7 Format Attribute Module) Loading res_pjsip_pidf_digium_body_supplement.so. == res_pjsip_pidf_digium_body_supplement.so => (PJSIP PIDF Digium presence supplement) Loading res_format_attr_celt.so. == Registered format interface for codec 'celt' == res_format_attr_celt.so => (CELT Format Attribute Module) Loading res_pjsip_dialog_info_body_generator.so. == res_pjsip_dialog_info_body_generator.so => (PJSIP Extension State Dialog Info+XML Provider) Loading res_format_attr_silk.so. == Registered format interface for codec 'silk' == res_format_attr_silk.so => (SILK Format Attribute Module) Loading res_musiconhold.so. == Parsing '/etc/asterisk/musiconhold.conf': Found == Registered application 'MusicOnHold' == Registered application 'StartMusicOnHold' == Registered application 'StopMusicOnHold' == res_musiconhold.so => (Music On Hold Resource) Loading res_pjsip_outbound_authenticator_digest.so. == res_pjsip_outbound_authenticator_digest.so => (PJSIP authentication resource) Loading res_pjsip_xpidf_body_generator.so. == res_pjsip_xpidf_body_generator.so => (PJSIP Extension State PIDF Provider) Loading res_pjsip_pidf_eyebeam_body_supplement.so. == res_pjsip_pidf_eyebeam_body_supplement.so => (PJSIP PIDF Eyebeam supplement) Loading res_pjsip_pidf_body_generator.so. == res_pjsip_pidf_body_generator.so => (PJSIP Extension State PIDF Provider) Loading res_format_attr_g729.so. == Registered format interface for codec 'g729' == res_format_attr_g729.so => (G.729 Format Attribute Module) Loading res_pjsip_mwi.so. == res_pjsip_mwi.so => (PJSIP MWI resource) Loading res_pjsip_publish_asterisk.so. == Parsing '/etc/asterisk/sorcery.conf': Found == Parsing '/etc/asterisk/pjsip.conf': Found == res_pjsip_publish_asterisk.so => (PJSIP Asterisk Event PUBLISH Support) Loading res_pjsip_exten_state.so. == res_pjsip_exten_state.so => (PJSIP Extension State Notifications) Loading chan_motif.so. == Parsing '/etc/asterisk/motif.conf': Found == Registered RTP glue 'Motif' == Registered channel type 'Motif' (Motif Jingle Channel Driver) == chan_motif.so => (Motif Jingle Channel Driver) Loading res_pjsip_sdp_rtp.so. == res_pjsip_sdp_rtp.so => (PJSIP SDP RTP/AVP stream handler) Loading chan_rtp.so. == Registered channel type 'MulticastRTP' (Multicast RTP Paging Channel Driver) == Registered channel type 'UnicastRTP' (Unicast RTP Media Channel Driver) == chan_rtp.so => (RTP Media Channel) Loading chan_iax2.so. == Parsing '/etc/asterisk/iax.conf': Found == Parsing '/etc/asterisk/users.conf': Found == Binding IAX2 to default address 0.0.0.0:4569 == Registered application 'IAX2Provision' == Registered custom function 'IAXPEER' == Registered custom function 'IAXVAR' == Manager registered action IAXpeers == Manager registered action IAXpeerlist == Manager registered action IAXnetstats == Manager registered action IAXregistry == Registered channel type 'IAX2' (Inter Asterisk eXchange Driver (Ver 2)) == 10 helper threads started == IAX Ready and Listening == Parsing '/etc/asterisk/iaxprov.conf': Found -- Loaded provisioning template 'default' == chan_iax2.so => (Inter Asterisk eXchange (Ver 2)) Loading chan_sip.so. SIP channel loading... == Parsing '/etc/asterisk/sip.conf': Found == Parsing '/etc/asterisk/users.conf': Found == SIP Listening on 0.0.0.0:5060 == Using SIP CoS mark 4 [Feb 20 15:22:34] NOTICE[32481]: chan_sip.c:31359 build_peer: The 'username' field for sip peers has been deprecated in favor of the term 'defaultuser' [Feb 20 15:22:34] NOTICE[32481]: chan_sip.c:31359 build_peer: The 'username' field for sip peers has been deprecated in favor of the term 'defaultuser' == Parsing '/etc/asterisk/sip_notify.conf': Found -- Message technology 'sip' registered. == 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 'CHECKSIPDOMAIN' == Manager registered action SIPpeers == Manager registered action SIPshowpeer == Manager registered action SIPqualifypeer == Manager registered action SIPshowregistry == Manager registered action SIPnotify == Manager registered action SIPpeerstatus == WebSocket registered sub-protocol 'sip' == chan_sip.so => (Session Initiation Protocol (SIP)) Loading chan_skinny.so. [Feb 20 15:22:34] NOTICE[32481]: chan_skinny.c:8418 config_load: Configuring skinny from skinny.conf [Feb 20 15:22:34] NOTICE[32481]: chan_skinny.c:8418 config_load: Configuring skinny from skinny.conf == Parsing '/etc/asterisk/skinny.conf': Found == Skinny listening on 0.0.0.0:2000 == Registered channel type 'Skinny' (Skinny Client Control Protocol (Skinny)) == Registered RTP glue 'Skinny' == Manager registered action SKINNYdevices == Manager registered action SKINNYshowdevice == Manager registered action SKINNYlines == Manager registered action SKINNYshowline == chan_skinny.so => (Skinny Client Control Protocol (Skinny)) Loading res_pjsip_t38.so. == res_pjsip_t38.so => (PJSIP T.38 UDPTL Support) Loading chan_mgcp.so. == Parsing '/etc/asterisk/mgcp.conf': Found == MGCP Listening on 0.0.0.0:2727 == Registered channel type 'MGCP' (Media Gateway Control Protocol (MGCP)) == Registered RTP glue 'MGCP' == chan_mgcp.so => (Media Gateway Control Protocol (MGCP)) Loading res_pjsip_dtmf_info.so. == res_pjsip_dtmf_info.so => (PJSIP DTMF INFO Support) Loading format_pcm.so. == 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) Loading res_pjsip_empty_info.so. == res_pjsip_empty_info.so => (PJSIP Empty INFO Support) Loading format_g726.so. == 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) Loading format_wav.so. == 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)) Loading res_pjsip_caller_id.so. == res_pjsip_caller_id.so => (PJSIP Caller ID Support) Loading format_g729.so. == Registered file format g729, extension(s) g729 == format_g729.so => (Raw G.729 data) Loading res_pjsip_diversion.so. == res_pjsip_diversion.so => (PJSIP Add Diversion Header Support) Loading res_pjsip_one_touch_record_info.so. == res_pjsip_one_touch_record_info.so => (PJSIP INFO One Touch Recording Support) Loading res_pjsip_nat.so. == res_pjsip_nat.so => (PJSIP NAT Support) Loading format_g719.so. == Registered file format g719, extension(s) g719 == format_g719.so => (ITU G.719) Loading res_adsi.so. == Parsing '/etc/asterisk/adsi.conf': Found == res_adsi.so => (ADSI Resource) Loading format_wav_gsm.so. == Registered file format wav49, extension(s) WAV|wav49 == format_wav_gsm.so => (Microsoft WAV format (Proprietary GSM)) Loading res_pjsip_transport_websocket.so. Loading format_h264.so. == Registered file format h264, extension(s) h264 == format_h264.so => (Raw H.264 data) Loading res_pjsip_registrar_expire.so. == res_pjsip_registrar_expire.so => (PJSIP Contact Auto-Expiration) Loading format_ilbc.so. == Registered file format iLBC, extension(s) ilbc == format_ilbc.so => (Raw iLBC data) Loading res_pjsip_history.so. == res_pjsip_history.so => (PJSIP History) Loading format_ogg_opus.so. == Registered file format ogg_opus, extension(s) opus == format_ogg_opus.so => (OGG/Opus audio) Loading res_pjsip_outbound_registration.so. == Parsing '/etc/asterisk/sorcery.conf': Found == Manager registered action PJSIPUnregister == Manager registered action PJSIPRegister == Manager registered action PJSIPShowRegistrationsOutbound == Parsing '/etc/asterisk/pjsip.conf': Found == Parsing '/etc/asterisk/pjsip_wizard.conf': Found == res_pjsip_outbound_registration.so => (PJSIP Outbound Registration Support) Loading res_pjsip_dlg_options.so. == res_pjsip_dlg_options.so => (SIP OPTIONS in dialog handler) Loading format_ogg_vorbis.so. == Registered file format ogg_vorbis, extension(s) ogg == format_ogg_vorbis.so => (OGG/Vorbis audio) Loading format_vox.so. == Registered file format vox, extension(s) vox == format_vox.so => (Dialogic VOX (ADPCM) File Format) Loading format_sln.so. == Registered file format sln, extension(s) sln|raw == Registered file format sln12, extension(s) sln12 == Registered file format sln16, extension(s) sln16 == Registered file format sln24, extension(s) sln24 == Registered file format sln32, extension(s) sln32 == Registered file format sln44, extension(s) sln44 == Registered file format sln48, extension(s) sln48 == Registered file format sln96, extension(s) sln96 == Registered file format sln192, extension(s) sln192 == format_sln.so => (Raw Signed Linear Audio support (SLN) 8khz-192khz) Loading res_pjsip_acl.so. == Parsing '/etc/asterisk/sorcery.conf': Found == Parsing '/etc/asterisk/pjsip.conf': Found == res_pjsip_acl.so => (PJSIP ACL Resource) Loading res_pjsip_rfc3326.so. == res_pjsip_rfc3326.so => (PJSIP RFC3326 Support) Loading app_stack.so. == 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' == Registered custom function 'STACK_PEEK' == app_stack.so => (Dialplan subroutines (Gosub, Return, etc)) Loading format_h263.so. == Registered file format h263, extension(s) h263 == format_h263.so => (Raw H.263 data) Loading res_pjsip_sips_contact.so. == res_pjsip_sips_contact.so => (UAC SIPS Contact support) Loading res_pjsip_notify.so. == Parsing '/etc/asterisk/pjsip_notify.conf': Found == Manager registered action PJSIPNotify == res_pjsip_notify.so => (CLI/AMI PJSIP NOTIFY Support) Loading format_gsm.so. == Registered file format gsm, extension(s) gsm == format_gsm.so => (Raw GSM data) Loading res_pjsip_messaging.so. -- Message technology 'pjsip' registered. == res_pjsip_messaging.so => (PJSIP Messaging Support) Loading func_dialplan.so. == Registered custom function 'DIALPLAN_EXISTS' == Registered custom function 'VALID_EXTEN' == func_dialplan.so => (Dialplan Context/Extension/Priority Checking Functions) Loading res_pjsip_logger.so. == res_pjsip_logger.so => (PJSIP Packet Logger) Loading res_pjsip_refer.so. == res_pjsip_refer.so => (PJSIP Blind and Attended Transfer Support) Loading format_siren7.so. == Registered file format siren7, extension(s) siren7 == format_siren7.so => (ITU G.722.1 (Siren7, licensed from Polycom)) Loading res_pjsip_header_funcs.so. == Registered custom function 'PJSIP_HEADER' == res_pjsip_header_funcs.so => (PJSIP Header Functions) Loading format_jpeg.so. == Registered format 'jpg' (JPEG (Joint Picture Experts Group)) == format_jpeg.so => (jpeg (joint picture experts group) image format) Loading format_g723.so. == Registered file format g723sf, extension(s) g723|g723sf == format_g723.so => (G.723.1 Simple Timestamp File Format) Loading res_pjsip_path.so. == res_pjsip_path.so => (PJSIP Path Header Support) Loading res_pjsip_phoneprov_provider.so. == Parsing '/etc/asterisk/sorcery.conf': Found == Parsing '/etc/asterisk/pjsip.conf': Found == Parsing '/etc/asterisk/pjsip_wizard.conf': Found == res_pjsip_phoneprov_provider.so => (PJSIP Phoneprov Provider) Loading res_pjsip_send_to_voicemail.so. == res_pjsip_send_to_voicemail.so => (PJSIP REFER Send to Voicemail Support) Loading format_siren14.so. == Registered file format siren14, extension(s) siren14 == format_siren14.so => (ITU G.722.1 Annex C (Siren14, licensed from Polycom)) Loading func_presencestate.so. == Registered custom function 'PRESENCE_STATE' == func_presencestate.so => (Gets or sets a presence state in the dialplan) Loading app_agent_pool.so. == Manager registered action Agents == Manager registered action AgentLogoff == Registered custom function 'AGENT' == Registered application 'AgentLogin' == Registered application 'AgentRequest' == Parsing '/etc/asterisk/agents.conf': Found == app_agent_pool.so => (Call center agent pool applications) Loading func_devstate.so. == Registered custom function 'DEVICE_STATE' == Registered custom function 'HINT' == func_devstate.so => (Gets or sets a device state in the dialplan) Loading app_confbridge.so. == Parsing '/etc/asterisk/confbridge.conf': Found [Feb 20 15:22:42] WARNING[32481]: config_options.c:1370 uint_handler_fn: Attempted to set internal_sample_rate=auto, but set it to 0 instead due to default) [Feb 20 15:22:42] WARNING[32481]: config_options.c:1370 uint_handler_fn: Attempted to set internal_sample_rate=auto, but set it to 0 instead due to default) [Feb 20 15:22:42] WARNING[32481]: config_options.c:1370 uint_handler_fn: Attempted to set internal_sample_rate=auto, but set it to 0 instead due to default) [Feb 20 15:22:42] WARNING[32481]: config_options.c:1370 uint_handler_fn: Attempted to set internal_sample_rate=auto, but set it to 0 instead due to default) [Feb 20 15:22:42] NOTICE[32481]: confbridge/conf_config_parser.c:2094 verify_default_profiles: Adding default_menu menu to app_confbridge [Feb 20 15:22:42] NOTICE[32481]: confbridge/conf_config_parser.c:2094 verify_default_profiles: Adding default_menu menu to app_confbridge == Registered channel type 'CBRec' (Conference Bridge Recording Channel) == Registered channel type 'CBAnn' (Conference Bridge Announcing Channel) == Registered application 'ConfBridge' == Registered custom function 'CONFBRIDGE' == Registered custom function 'CONFBRIDGE_INFO' == Manager registered action ConfbridgeList == Manager registered action ConfbridgeListRooms == Manager registered action ConfbridgeMute == Manager registered action ConfbridgeUnmute == Manager registered action ConfbridgeKick == Manager registered action ConfbridgeUnlock == Manager registered action ConfbridgeLock == Manager registered action ConfbridgeStartRecord == Manager registered action ConfbridgeStopRecord == Manager registered action ConfbridgeSetSingleVideoSrc == app_confbridge.so => (Conference Bridge Application) Loading res_calendar_caldav.so. == Registered calendar type 'caldav' (CalDAV calendars) == res_calendar_caldav.so => (Asterisk CalDAV Calendar Integration) Loading res_calendar_exchange.so. == Registered calendar type 'exchange' (MS Exchange calendars) == res_calendar_exchange.so => (Asterisk MS Exchange Calendar Integration) Loading res_calendar_icalendar.so. == Registered calendar type 'ical' (iCalendar .ics calendars) == res_calendar_icalendar.so => (Asterisk iCalendar .ics file integration) Loading res_calendar_ews.so. == Registered calendar type 'ews' (MS Exchange Web Service calendars) == res_calendar_ews.so => (Asterisk MS Exchange Web Service Calendar Integration) Loading cdr_sqlite3_custom.so. == Parsing '/etc/asterisk/cdr_sqlite3_custom.conf': Found Loading cel_sqlite3_custom.so. == Parsing '/etc/asterisk/cel_sqlite3_custom.conf': Found Loading cdr_radius.so. == Parsing '/etc/asterisk/cdr.conf': Found == cdr_radius.so => (RADIUS CDR Backend) Loading cel_radius.so. == Parsing '/etc/asterisk/cel.conf': Found == cel_radius.so => (RADIUS CEL Backend) Loading cdr_custom.so. == Parsing '/etc/asterisk/cdr_custom.conf': Found == cdr_custom.so => (Customizable Comma Separated Values CDR Backend) Loading cel_odbc.so. == Parsing '/etc/asterisk/cel_odbc.conf': Found == cel_odbc.so => (ODBC CEL backend) Loading cdr_manager.so. == Parsing '/etc/asterisk/cdr_manager.conf': Found == cdr_manager.so => (Asterisk Manager Interface CDR Backend) Loading cel_manager.so. == Parsing '/etc/asterisk/cel.conf': Found == cel_manager.so => (Asterisk Manager Interface CEL Backend) Loading cdr_adaptive_odbc.so. == Parsing '/etc/asterisk/cdr_adaptive_odbc.conf': Found == cdr_adaptive_odbc.so => (Adaptive ODBC CDR backend) Loading cdr_csv.so. == Parsing '/etc/asterisk/cdr.conf': Found == cdr_csv.so => (Comma Separated Values CDR Backend) Loading cel_custom.so. == Parsing '/etc/asterisk/cel_custom.conf': Found [Feb 20 15:22:43] NOTICE[32481]: cel_custom.c:97 load_config: No mappings found in cel_custom.conf. Not logging CEL to custom CSVs. [Feb 20 15:22:43] NOTICE[32481]: cel_custom.c:97 load_config: No mappings found in cel_custom.conf. Not logging CEL to custom CSVs. Added CEL CSV mapping for 0 files. == cel_custom.so => (Customizable Comma Separated Values CEL Backend) Loading cel_pgsql.so. == Parsing '/etc/asterisk/cel_pgsql.conf': Found [Feb 20 15:22:43] WARNING[32481]: cel_pgsql.c:441 process_my_load_module: CEL pgsql config file missing global section. [Feb 20 15:22:43] WARNING[32481]: cel_pgsql.c:441 process_my_load_module: CEL pgsql config file missing global section. == cel_pgsql.so => (PostgreSQL CEL Backend) Loading cdr_syslog.so. == Parsing '/etc/asterisk/cdr_syslog.conf': Found Loading cdr_pgsql.so. == Parsing '/etc/asterisk/cdr_pgsql.conf': Found [Feb 20 15:22:43] NOTICE[32481]: cdr_pgsql.c:523 config_module: cdr_pgsql configuration contains no global section, skipping module load. [Feb 20 15:22:43] NOTICE[32481]: cdr_pgsql.c:523 config_module: cdr_pgsql configuration contains no global section, skipping module load. Loading cel_tds.so. == Parsing '/etc/asterisk/cel_tds.conf': Found [Feb 20 15:22:43] NOTICE[32481]: cel_tds.c:452 tds_load_module: cel_tds has no global category, nothing to configure. [Feb 20 15:22:43] NOTICE[32481]: cel_tds.c:452 tds_load_module: cel_tds has no global category, nothing to configure. [Feb 20 15:22:43] WARNING[32481]: cel_tds.c:557 load_module: cel_tds module had config problems; declining load [Feb 20 15:22:43] WARNING[32481]: cel_tds.c:557 load_module: cel_tds module had config problems; declining load Loading cdr_odbc.so. == Parsing '/etc/asterisk/cdr_odbc.conf': Found == cdr_odbc.so => (ODBC CDR Backend) Loading cdr_tds.so. == Parsing '/etc/asterisk/cdr_tds.conf': Found Loading app_url.so. == Registered application 'SendURL' == app_url.so => (Send URL Applications) Loading func_vmcount.so. == Registered custom function 'VMCOUNT' == func_vmcount.so => (Indicator for whether a voice mailbox has messages in a given folder.) Loading codec_g726.so. == Registered translator 'g726tolin' from codec g726 to slin, table cost, 900000, computational cost 145977 == Registered translator 'lintog726' from codec slin to g726, table cost, 600000, computational cost 224966 == Registered translator 'g726aal2tolin' from codec g726aal2 to slin, table cost, 900000, computational cost 118982 == Registered translator 'lintog726aal2' from codec slin to g726aal2, table cost, 600000, computational cost 209969 == codec_g726.so => (ITU G.726-32kbps G726 Transcoder) Loading pbx_dundi.so. == Parsing '/etc/asterisk/dundi.conf': Found == Registered custom function 'DUNDILOOKUP' == Registered custom function 'DUNDIQUERY' == Registered custom function 'DUNDIRESULT' == DUNDi Ready and Listening on 0.0.0.0 port 4520 == pbx_dundi.so => (Distributed Universal Number Discovery (DUNDi)) Loading func_global.so. == Registered custom function 'GLOBAL' == Registered custom function 'SHARED' == func_global.so => (Variable dialplan functions) Loading res_ari_applications.so. == res_ari_applications.so => (RESTful API module - Stasis application resources) Loading app_mp3.so. == Registered application 'MP3Player' == app_mp3.so => (Silly MP3 Application) Loading app_morsecode.so. == Registered application 'Morsecode' == app_morsecode.so => (Morse code) Loading app_getcpeid.so. == Registered application 'GetCPEID' == app_getcpeid.so => (Get ADSI CPE ID) Loading app_waituntil.so. == Registered application 'WaitUntil' == app_waituntil.so => (Wait until specified time) Loading res_realtime.so. == res_realtime.so => (Realtime Data Lookup/Rewrite) Loading func_sha1.so. == Registered custom function 'SHA1' == func_sha1.so => (SHA-1 computation dialplan function) Loading func_sysinfo.so. == Registered custom function 'SYSINFO' == func_sysinfo.so => (System information related functions) Loading res_ari_asterisk.so. == res_ari_asterisk.so => (RESTful API module - Asterisk resources) Loading func_iconv.so. == Registered custom function 'ICONV' == func_iconv.so => (Charset conversions) Loading app_directed_pickup.so. == Registered application 'Pickup' == Registered application 'PickupChan' == app_directed_pickup.so => (Directed Call Pickup Application) Loading chan_unistim.so. == Parsing '/etc/asterisk/unistim.conf': Found == UNISTIM Listening on 0.0.0.0:5000 == Registered channel type 'USTM' (UNISTIM Channel Driver) == Registered RTP glue 'USTM' == chan_unistim.so => (UNISTIM Protocol (USTM)) Loading app_talkdetect.so. == Registered application 'BackgroundDetect' == app_talkdetect.so => (Playback with Talk Detection) Loading func_math.so. == Registered custom function 'MATH' == Registered custom function 'INC' == Registered custom function 'DEC' == func_math.so => (Mathematical dialplan function) Loading res_convert.so. == res_convert.so => (File format conversion CLI command) Loading app_festival.so. == Parsing '/etc/asterisk/festival.conf': Found == Registered application 'Festival' == app_festival.so => (Simple Festival Interface) Loading func_sorcery.so. == Registered custom function 'AST_SORCERY' == func_sorcery.so => (Get a field from a sorcery object) Loading app_channelredirect.so. == Registered application 'ChannelRedirect' == app_channelredirect.so => (Redirects a given channel to a dialplan target) Loading res_format_attr_h263.so. == Registered format interface for codec 'h263' == Registered format interface for codec 'h263p' == res_format_attr_h263.so => (H.263 Format Attribute Module) Loading app_jack.so. == Registered application 'JACK' == Registered custom function 'JACK_HOOK' == app_jack.so => (JACK Interface) Loading chan_oss.so. == Parsing '/etc/asterisk/oss.conf': Found == Registered channel type 'Console' (OSS Console Channel Driver) == chan_oss.so => (OSS Console Channel Driver) Loading func_odbc.so. == 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) Loading app_system.so. == Registered application 'TrySystem' == Registered application 'System' == app_system.so => (Generic System() application) Loading func_shell.so. == Registered custom function 'SHELL' == func_shell.so => (Collects the output generated by a command executed by the system shell) Loading app_senddtmf.so. == Manager registered action PlayDTMF == Registered application 'SendDTMF' == app_senddtmf.so => (Send DTMF digits Application) Loading app_adsiprog.so. == Registered application 'ADSIProg' == app_adsiprog.so => (Asterisk ADSI Programming Application) Loading app_ices.so. == Registered application 'ICES' == app_ices.so => (Encode and Stream via icecast and ices) Loading app_db.so. == Registered application 'DBdel' == Registered application 'DBdeltree' == app_db.so => (Database Access Functions) Loading app_waitforring.so. == Registered application 'WaitForRing' == app_waitforring.so => (Waits until first ring after time) Loading app_forkcdr.so. == Registered application 'ForkCDR' == app_forkcdr.so => (Fork The CDR into 2 separate entities) Loading app_macro.so. == Registered application 'MacroExit' == Registered application 'MacroIf' == Registered application 'MacroExclusive' == Registered application 'Macro' == app_macro.so => (Extension Macros) Loading codec_a_mu.so. == Registered translator 'alawtoulaw' from codec alaw to ulaw, table cost, 915000, computational cost 17996 == Registered translator 'ulawtoalaw' from codec ulaw to alaw, table cost, 915000, computational cost 17997 == codec_a_mu.so => (A-law and Mulaw direct Coder/Decoder) Loading app_playtones.so. == Registered application 'PlayTones' == Registered application 'StopPlayTones' == app_playtones.so => (Playtones Application) Loading codec_alaw.so. == Registered translator 'alawtolin' from codec alaw to slin, table cost, 900000, computational cost 21997 == Registered translator 'lintoalaw' from codec slin to alaw, table cost, 600000, computational cost 20997 == codec_alaw.so => (A-law Coder/Decoder) Loading func_rand.so. == Registered custom function 'RAND' == func_rand.so => (Random number dialplan function) Loading app_page.so. == Registered application 'Page' == app_page.so => (Page Multiple Phones) Loading func_timeout.so. == Registered custom function 'TIMEOUT' == func_timeout.so => (Channel timeout dialplan functions) Loading res_pjsip_endpoint_identifier_anonymous.so. == res_pjsip_endpoint_identifier_anonymous.so => (PJSIP Anonymous endpoint identifier) Loading bridge_builtin_features.so. == bridge_builtin_features.so => (Built in bridging features) Loading res_clioriginate.so. == res_clioriginate.so => (Call origination and redirection from the CLI) Loading codec_g722.so. == Registered translator 'g722tolin' from codec g722 to slin, table cost, 960000, computational cost 167975 == Registered translator 'lintog722' from codec slin to g722, table cost, 825000, computational cost 164975 == Registered translator 'g722tolin16' from codec g722 to slin, table cost, 900000, computational cost 306953 == Registered translator 'lin16tog722' from codec slin to g722, table cost, 600000, computational cost 311953 == codec_g722.so => (ITU G.722-64kbps G722 Transcoder) Loading app_nbscat.so. == Registered application 'NBScat' == app_nbscat.so => (Silly NBS Stream Application) Loading func_version.so. == Registered custom function 'VERSION' == func_version.so => (Get Asterisk Version/Build Info) Loading app_image.so. == Registered application 'SendImage' == app_image.so => (Image Transmission Application) Loading codec_ilbc.so. == Registered translator 'ilbctolin' from codec ilbc to slin, table cost, 900000, computational cost 221966 == Registered translator 'lintoilbc' from codec slin to ilbc, table cost, 600000, computational cost 739889 == codec_ilbc.so => (iLBC Coder/Decoder) Loading app_voicemail.so. == 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 'VoiceMailPlayMsg' == Registered application 'VMSayName' == Registered custom function 'MAILBOX_EXISTS' == Registered custom function 'VM_INFO' == Manager registered action VoicemailUsersList == Manager registered action VoicemailRefresh == app_voicemail.so => (Comedian Mail (Voicemail System)) Loading func_jitterbuffer.so. == Registered custom function 'JITTERBUFFER' == func_jitterbuffer.so => (Jitter buffer for read side of channel.) Loading app_amd.so. == Parsing '/etc/asterisk/amd.conf': Found -- AMD defaults: initialSilence [2500] greeting [1500] afterGreetingSilence [800] totalAnalysisTime [5000] minimumWordLength [100] betweenWordsSilence [50] maximumNumberOfWords [3] silenceThreshold [256] maximumWordLength [5000] == Registered application 'AMD' == app_amd.so => (Answering Machine Detection Application) Loading app_exec.so. == Registered application 'Exec' == Registered application 'TryExec' == Registered application 'ExecIf' == app_exec.so => (Executes dialplan applications) Loading pbx_realtime.so. == pbx_realtime.so => (Realtime Switch) Loading app_chanspy.so. == Registered application 'ChanSpy' == Registered application 'ExtenSpy' == Registered application 'DAHDIScan' == app_chanspy.so => (Listen to the audio of an active channel) Loading app_minivm.so. == Registered application 'MinivmRecord' == Registered application 'MinivmGreet' == Registered application 'MinivmNotify' == Registered application 'MinivmDelete' == Registered application 'MinivmAccMess' == Registered application 'MinivmMWI' == Registered custom function 'MINIVMACCOUNT' == Registered custom function 'MINIVMCOUNTER' == Parsing '/etc/asterisk/minivm.conf': Found == app_minivm.so => (Mini VoiceMail (A minimal Voicemail e-mail System)) Loading app_originate.so. == Registered application 'Originate' == app_originate.so => (Originate call) Loading func_audiohookinherit.so. == Registered custom function 'AUDIOHOOK_INHERIT' == func_audiohookinherit.so => (Audiohook inheritance placeholder function) Loading pbx_ael.so. == Registered application 'AELSub' == Setting global variable 'CONSOLE-AEL' to '"Console/dsp"' == Setting global variable 'IAXINFO-AEL' to 'guest' == Setting global variable 'OUTBOUND-TRUNK' to '"Zap/g2"' == Setting global variable 'OUTBOUND-TRUNKMSD' to '1' -- Registered extension context 'ael-dundi-e164-canonical'; registrar: pbx_ael -- Registered extension context 'ael-dundi-e164-customers'; registrar: pbx_ael -- Registered extension context 'ael-dundi-e164-via-pstn'; registrar: pbx_ael -- Registered extension context 'ael-dundi-e164-local'; registrar: pbx_ael -- Including context 'ael-dundi-e164-canonical' in context 'ael-dundi-e164-local' -- Including context 'ael-dundi-e164-customers' in context 'ael-dundi-e164-local' -- Including context 'ael-dundi-e164-via-pstn' in context 'ael-dundi-e164-local' -- Registered extension context 'ael-dundi-e164-switch'; registrar: pbx_ael -- Including switch 'DUNDi/e164' in context 'ael-dundi-e164-switch' -- Registered extension context 'ael-dundi-e164-lookup'; registrar: pbx_ael -- Including context 'ael-dundi-e164-local' in context 'ael-dundi-e164-lookup' -- Including context 'ael-dundi-e164-switch' in context 'ael-dundi-e164-lookup' -- Registered extension context 'ael-dundi-e164'; registrar: pbx_ael -- Registered extension context 'ael-iaxtel700'; registrar: pbx_ael -- Registered extension context 'ael-iaxprovider'; registrar: pbx_ael -- Registered extension context 'ael-trunkint'; registrar: pbx_ael -- Including context 'ael-dundi-e164-lookup' in context 'ael-trunkint' -- Registered extension context 'ael-trunkld'; registrar: pbx_ael -- Including context 'ael-dundi-e164-lookup' in context 'ael-trunkld' -- Registered extension context 'ael-trunklocal'; registrar: pbx_ael -- Registered extension context 'ael-trunktollfree'; registrar: pbx_ael -- Registered extension context 'ael-international'; registrar: pbx_ael -- Including context 'ael-longdistance' in context 'ael-international' -- Including context 'ael-trunkint' in context 'ael-international' -- Registered extension context 'ael-longdistance'; registrar: pbx_ael -- Including context 'ael-local' in context 'ael-longdistance' -- Including context 'ael-trunkld' in context 'ael-longdistance' -- Registered extension context 'ael-local'; registrar: pbx_ael -- Including context 'ael-default' in context 'ael-local' -- Including context 'ael-trunklocal' in context 'ael-local' -- Including context 'ael-iaxtel700' in context 'ael-local' -- Including context 'ael-trunktollfree' in context 'ael-local' -- Including context 'ael-iaxprovider' in context 'ael-local' -- Registered extension context 'ael-std-exten-ael'; registrar: pbx_ael -- Registered extension context 'ael-demo'; registrar: pbx_ael -- Registered extension context 'ael-default'; registrar: pbx_ael -- Including context 'ael-demo' in context 'ael-default' -- Registered extension context 'ael-builtin-h-bubble'; registrar: pbx_ael -- Including context 'ael-builtin-h-bubble' in context 'ael-dundi-e164' -- Including context 'ael-builtin-h-bubble' in context 'ael-std-exten-ael' -- Added extension '~~s~~' priority 1 to ael-dundi-e164 -- Added extension '~~s~~' priority 2 to ael-dundi-e164 -- Added extension '~~s~~' priority 3 to ael-dundi-e164 -- Added extension '_91700XXXXXXX' priority 1 to ael-iaxtel700 -- Added extension '_9011.' priority 1 to ael-trunkint -- Added extension '_9011.' priority 2 to ael-trunkint -- Added extension '_91NXXNXXXXXX' priority 1 to ael-trunkld -- Added extension '_91NXXNXXXXXX' priority 2 to ael-trunkld -- Added extension '_9NXXXXXX' priority 1 to ael-trunklocal -- Added extension '_91800NXXXXXX' priority 1 to ael-trunktollfree -- Added extension '_91888NXXXXXX' priority 1 to ael-trunktollfree -- Added extension '_91877NXXXXXX' priority 1 to ael-trunktollfree -- Added extension '_91866NXXXXXX' priority 1 to ael-trunktollfree -- Added extension '~~s~~' priority 1 to ael-std-exten-ael -- Added extension '~~s~~' priority 2 to ael-std-exten-ael -- Added extension '~~s~~' priority 3 to ael-std-exten-ael -- Added extension '~~s~~' priority 4 to ael-std-exten-ael -- Added extension '~~s~~' priority 5 to ael-std-exten-ael -- Added extension '~~s~~' priority 6 to ael-std-exten-ael -- Added extension '~~s~~' priority 7 to ael-std-exten-ael -- Added extension '~~s~~' priority 8 to ael-std-exten-ael -- Added extension 'a' priority 1 to ael-std-exten-ael -- Added extension 'a' priority 2 to ael-std-exten-ael -- Added extension '_sw_1_.' priority 10 to ael-std-exten-ael -- Added extension '_sw_1_.' priority 11 to ael-std-exten-ael -- Added extension 'sw_1_' priority 10 to ael-std-exten-ael -- Added extension 'sw_1_BUSY' priority 10 to ael-std-exten-ael -- Added extension 'sw_1_BUSY' priority 11 to ael-std-exten-ael -- Added extension 's' priority 1 to ael-demo -- Added extension 's' priority 2 to ael-demo -- Added extension 's' priority 3 to ael-demo -- Added extension 's' priority 4 to ael-demo -- Added extension 's' priority 5 to ael-demo -- Added extension 's' priority 6 to ael-demo -- Added extension 's' priority 7 to ael-demo -- Added extension 's' priority 8 to ael-demo -- Added extension 's' priority 9 to ael-demo -- Added extension 's' priority 10 to ael-demo -- Added extension 's' priority 11 to ael-demo -- Added extension 's' priority 12 to ael-demo -- Added extension '2' priority 1 to ael-demo -- Added extension '2' priority 2 to ael-demo -- Added extension '3' priority 1 to ael-demo -- Added extension '3' priority 2 to ael-demo -- Added extension '1000' priority 1 to ael-demo -- Added extension '500' priority 1 to ael-demo -- Added extension '500' priority 2 to ael-demo -- Added extension '500' priority 3 to ael-demo -- Added extension '500' priority 4 to ael-demo -- Added extension '600' priority 1 to ael-demo -- Added extension '600' priority 2 to ael-demo -- Added extension '600' priority 3 to ael-demo -- Added extension '600' priority 4 to ael-demo -- Added extension '_1234' priority 1 to ael-demo -- Added extension '8500' priority 1 to ael-demo -- Added extension '8500' priority 2 to ael-demo -- Added extension '#' priority 1 to ael-demo -- Added extension '#' priority 2 to ael-demo -- Added extension 't' priority 1 to ael-demo -- Added extension 'i' priority 1 to ael-demo -- Added extension 'h' priority 1 to ael-builtin-h-bubble -- Added extension 'h' priority 9991 to ael-builtin-h-bubble -- Added extension 'h' priority 9992 to ael-builtin-h-bubble -- Added extension 'h' priority 9993 to ael-builtin-h-bubble -- Added extension 'h' priority 9994 to ael-builtin-h-bubble -- Added extension 'h' priority 9995 to ael-builtin-h-bubble -- Added extension 'h' priority 9996 to ael-builtin-h-bubble -- Registered extension context 'parkedcalls'; registrar: res_parking/default -- merging incls/swits/igpats from old(parkedcalls) to new(parkedcalls) context, registrar = pbx_ael -- Added extension '720' priority 1 to parkedcalls -- Added extension '719' priority 1 to parkedcalls -- Added extension '718' priority 1 to parkedcalls -- Added extension '717' priority 1 to parkedcalls -- Added extension '716' priority 1 to parkedcalls -- Added extension '715' priority 1 to parkedcalls -- Added extension '714' priority 1 to parkedcalls -- Added extension '713' priority 1 to parkedcalls -- Added extension '712' priority 1 to parkedcalls -- Added extension '711' priority 1 to parkedcalls -- Added extension '710' priority 1 to parkedcalls -- Added extension '709' priority 1 to parkedcalls -- Added extension '708' priority 1 to parkedcalls -- Added extension '707' priority 1 to parkedcalls -- Added extension '706' priority 1 to parkedcalls -- Added extension '705' priority 1 to parkedcalls -- Added extension '704' priority 1 to parkedcalls -- Added extension '703' priority 1 to parkedcalls -- Added extension '702' priority 1 to parkedcalls -- Added extension '701' priority 1 to parkedcalls -- Added extension '700' priority 1 to parkedcalls -- Registered extension context '__func_periodic_hook_context__'; registrar: func_periodic_hook -- merging incls/swits/igpats from old(__func_periodic_hook_context__) to new(__func_periodic_hook_context__) context, registrar = pbx_ael -- Added extension 'hook' priority 1 (CID match '') to __func_periodic_hook_context__ -- Added extension 'hook' priority 2 (CID match '') to __func_periodic_hook_context__ -- Added extension 'hook' priority 3 (CID match '') to __func_periodic_hook_context__ -- Added extension 'hook' priority 4 (CID match '') to __func_periodic_hook_context__ -- Added extension 'hook' priority 5 (CID match '') to __func_periodic_hook_context__ -- Added extension 'hook' priority 6 (CID match '') to __func_periodic_hook_context__ -- Added extension 'beep' priority 1 (CID match '') to __func_periodic_hook_context__ -- Added extension 'beep' priority 2 (CID match '') to __func_periodic_hook_context__ -- Registered extension context 'local'; registrar: pbx_lua -- merging incls/swits/igpats from old(local) to new(local) context, registrar = pbx_ael -- Including switch 'Lua/' in context 'local' -- Registered extension context 'demo'; registrar: pbx_lua -- merging incls/swits/igpats from old(demo) to new(demo) context, registrar = pbx_ael -- Including switch 'Lua/' in context 'demo' -- Added extension '1001' priority -1 to demo -- Added extension '1000' priority -1 to demo -- Registered extension context 'public'; registrar: pbx_lua -- merging incls/swits/igpats from old(public) to new(public) context, registrar = pbx_ael -- Including switch 'Lua/' in context 'public' -- Registered extension context 'default'; registrar: pbx_lua -- merging incls/swits/igpats from old(default) to new(default) context, registrar = pbx_ael -- Including switch 'Lua/' in context 'default' -- Added extension '1234' priority -1 to default -- Time to scan old dialplan and merge leftovers back into the new: 0.203204 sec -- Time to restore hints and swap in new dialplan: 0.000607 sec -- Time to delete the old dialplan: 0.030425 sec -- Total time merge_contexts_delete: 0.234236 sec == pbx_ael.so => (Asterisk Extension Language Compiler) Loading pbx_config.so. == Manager registered action DialplanExtensionAdd == Manager registered action DialplanExtensionRemove == Parsing '/etc/asterisk/extensions.conf': Found == Setting global variable 'CONSOLE' to 'Console/dsp' == Setting global variable 'IAXINFO' to 'guest' == Setting global variable 'TRUNK' to 'DAHDI/G2' == Setting global variable 'TRUNKMSD' to '1' -- Registered extension context 'dundi-e164-canonical'; registrar: pbx_config -- Registered extension context 'dundi-e164-customers'; registrar: pbx_config -- Registered extension context 'dundi-e164-via-pstn'; registrar: pbx_config -- Registered extension context 'dundi-e164-local'; registrar: pbx_config -- Including context 'dundi-e164-canonical' in context 'dundi-e164-local' -- Including context 'dundi-e164-customers' in context 'dundi-e164-local' -- Including context 'dundi-e164-via-pstn' in context 'dundi-e164-local' -- Registered extension context 'dundi-e164-switch'; registrar: pbx_config -- Including switch 'DUNDi/e164' in context 'dundi-e164-switch' -- Registered extension context 'dundi-e164-lookup'; registrar: pbx_config -- Including context 'dundi-e164-local' in context 'dundi-e164-lookup' -- Including context 'dundi-e164-switch' in context 'dundi-e164-lookup' -- Registered extension context 'macro-dundi-e164'; registrar: pbx_config -- Added extension 's' priority 1 to macro-dundi-e164 -- Including context 'dundi-e164-lookup' in context 'macro-dundi-e164' -- Registered extension context 'iaxtel700'; registrar: pbx_config -- Added extension '_91700XXXXXXX' priority 1 to iaxtel700 -- Registered extension context 'iaxprovider'; registrar: pbx_config -- Registered extension context 'trunkint'; registrar: pbx_config -- Added extension '_9011.' priority 1 to trunkint -- Added extension '_9011.' priority 2 to trunkint -- Registered extension context 'trunkld'; registrar: pbx_config -- Added extension '_91NXXNXXXXXX' priority 1 to trunkld -- Added extension '_91NXXNXXXXXX' priority 2 to trunkld -- Registered extension context 'trunklocal'; registrar: pbx_config -- Added extension '_9NXXXXXX' priority 1 to trunklocal -- Registered extension context 'trunktollfree'; registrar: pbx_config -- Added extension '_91800NXXXXXX' priority 1 to trunktollfree -- Added extension '_91888NXXXXXX' priority 1 to trunktollfree -- Added extension '_91877NXXXXXX' priority 1 to trunktollfree -- Added extension '_91866NXXXXXX' priority 1 to trunktollfree -- Registered extension context 'international'; registrar: pbx_config -- Including context 'longdistance' in context 'international' -- Including context 'trunkint' in context 'international' -- Registered extension context 'longdistance'; registrar: pbx_config -- Including context 'local' in context 'longdistance' -- Including context 'trunkld' in context 'longdistance' -- Registered extension context 'local'; registrar: pbx_config -- Including context 'default' in context 'local' -- Including context 'trunklocal' in context 'local' -- Including context 'iaxtel700' in context 'local' -- Including context 'trunktollfree' in context 'local' -- Including context 'iaxprovider' in context 'local' -- Including context 'parkedcalls' in context 'local' -- Registered extension context 'outbound-freenum'; registrar: pbx_config -- Added extension '_X*X!' priority 1 to outbound-freenum -- Added extension '_XX*X!' priority 1 to outbound-freenum -- Added extension '_XXX*X!' priority 1 to outbound-freenum -- Added extension '_XXXX*X!' priority 1 to outbound-freenum -- Added extension '_XXXXX*X!' priority 1 to outbound-freenum -- Added extension '_XXXXXX*X!' priority 1 to outbound-freenum -- Registered extension context 'outbound-freenum2'; registrar: pbx_config -- Added extension '_X!' priority 1 to outbound-freenum2 -- Added extension '_X!' priority 2 to outbound-freenum2 -- Added extension '_X!' priority 3 to outbound-freenum2 -- Added extension '_X!' priority 4 to outbound-freenum2 -- Added extension '_X!' priority 5 to outbound-freenum2 -- Added extension '_X!' priority 6 to outbound-freenum2 -- Added extension '_X!' priority 7 to outbound-freenum2 -- Added extension '_X!' priority 8 to outbound-freenum2 -- Added extension '_X!' priority 9 to outbound-freenum2 -- Added extension '_X!' priority 10 to outbound-freenum2 -- Added extension '_X!' priority 11 to outbound-freenum2 -- Added extension '_X!' priority 12 to outbound-freenum2 -- Added extension '_X!' priority 13 to outbound-freenum2 -- Added extension 'fn-BUSY' priority 1 to outbound-freenum2 -- Added extension '_f[n]-.' priority 1 to outbound-freenum2 -- Added extension '_f[n]-.' priority 2 to outbound-freenum2 -- Registered extension context 'macro-trunkdial'; registrar: pbx_config -- Added extension 's' priority 1 to macro-trunkdial -- Added extension 's' priority 2 to macro-trunkdial -- Added extension 's-NOANSWER' priority 1 to macro-trunkdial -- Added extension 's-BUSY' priority 1 to macro-trunkdial -- Added extension '_s-.' priority 1 to macro-trunkdial -- 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-BUSY' priority 1 to stdexten -- Added extension 'stdexten-BUSY' priority 2 to stdexten -- Added extension '_stde[x]te[n]-.' priority 1 to stdexten -- Added extension 'a' priority 1 to stdexten -- Added extension 'a' priority 2 to stdexten -- Registered extension context 'stdPrivacyexten'; registrar: pbx_config -- Added extension '_X.' priority 60000 to stdPrivacyexten -- Added extension '_X.' priority 60001 to stdPrivacyexten -- Added extension '_X.' priority 60002 to stdPrivacyexten -- Added extension '_X.' priority 60003 to stdPrivacyexten -- Added extension '_X.' priority 60004 to stdPrivacyexten -- Added extension '_X.' priority 60005 to stdPrivacyexten -- Added extension '_X.' priority 60006 to stdPrivacyexten -- Added extension '_X.' priority 60007 to stdPrivacyexten -- Added extension '_X.' priority 60008 to stdPrivacyexten -- Added extension 'stdexten-NOANSWER' priority 1 to stdPrivacyexten -- Added extension 'stdexten-NOANSWER' priority 2 to stdPrivacyexten -- Added extension 'stdexten-NOANSWER' priority 3 to stdPrivacyexten -- Added extension 'stdexten-BUSY' priority 1 to stdPrivacyexten -- Added extension 'stdexten-BUSY' priority 2 to stdPrivacyexten -- Added extension 'stdexten-BUSY' priority 3 to stdPrivacyexten -- Added extension 'stdexten-DONTCALL' priority 1 to stdPrivacyexten -- Added extension 'stdexten-TORTURE' priority 1 to stdPrivacyexten -- Added extension '_stde[x]te[n]-.' priority 1 to stdPrivacyexten -- Added extension 'a' priority 1 to stdPrivacyexten -- Added extension 'a' priority 2 to stdPrivacyexten -- Registered extension context 'macro-page'; registrar: pbx_config -- Added extension 's' priority 1 to macro-page -- Added extension 's' priority 2 to macro-page -- Added extension 's' priority 3 to macro-page -- Added extension 's' priority 4 to macro-page -- Added extension 's' priority 5 to macro-page -- Added extension 's' priority 6 to macro-page -- Added extension 's' priority 7 to macro-page -- Registered extension context 'demo'; registrar: pbx_config -- Including context 'stdexten' in context 'demo' -- Added extension 's' priority 1 to demo -- Added extension 's' priority 2 to demo -- Added extension 's' priority 3 to demo -- Added extension 's' priority 4 to demo -- Added extension 's' priority 5 to demo -- Added extension 's' priority 6 to demo -- Added extension 's' priority 7 to demo -- Added extension '2' priority 1 to demo -- Added extension '2' priority 2 to demo -- Added extension '3' priority 1 to demo -- Added extension '3' priority 2 to demo -- Added extension '1000' priority 1 to demo -- Added extension '1234' priority 1 to demo -- Added extension '1234' priority 2 to demo -- Added extension '1234' priority 3 to demo -- Added extension '1235' priority 1 to demo -- Added extension '1236' priority 1 to demo -- Added extension '1236' priority 2 to demo -- Added extension '#' priority 1 to demo -- Added extension '#' priority 2 to demo -- Added extension 't' priority 1 to demo -- Added extension 'i' priority 1 to demo -- Added extension '500' priority 1 to demo -- Added extension '500' priority 2 to demo -- Added extension '500' priority 3 to demo -- Added extension '500' priority 4 to demo -- Added extension '600' priority 1 to demo -- Added extension '600' priority 2 to demo -- Added extension '600' priority 3 to demo -- Added extension '600' priority 4 to demo -- Added extension '76245' priority 1 to demo -- Added extension '_7XXX' priority 1 to demo -- Added extension '7999' priority 1 to demo -- Added extension '7999' priority 2 to demo -- Added extension '8500' priority 1 to demo -- Added extension '8500' priority 2 to demo -- Registered extension context 'page'; registrar: pbx_config -- Added extension '_X.' priority 1 to page -- Registered extension context 'public'; registrar: pbx_config -- Including context 'justek' in context 'public' -- Registered extension context 'default'; registrar: pbx_config -- Including context 'justek' in context 'default' -- Registered extension context 'time'; registrar: pbx_config -- Added extension '_X.' priority 30000 to time -- Added extension '_X.' priority 30001 to time -- Added extension '_X.' priority 30002 to time -- Added extension '_X.' priority 30003 to time -- Added extension '_X.' priority 30004 to time -- Added extension '_X.' priority 30005 to time -- Added extension '_X.' priority 30006 to time -- Added extension '_X.' priority 30007 to time -- Added extension '_X.' priority 30008 to time -- Added extension '_X.' priority 30009 to time -- Added extension '_X.' priority 30010 to time -- Registered extension context 'ani'; registrar: pbx_config -- Added extension '_X.' priority 40000 to ani -- Added extension '_X.' priority 40001 to ani -- Added extension '_X.' priority 40002 to ani -- Added extension '_X.' priority 40003 to ani -- Added extension '_X.' priority 40004 to ani -- Added extension '_X.' priority 40005 to ani -- Added extension '_X.' priority 40006 to ani -- Added extension '_X.' priority 40007 to ani -- Registered extension context 'justek'; registrar: pbx_config -- Added extension '_50XXX' priority 1 to justek -- Added extension '_50XXX' priority 2 to justek -- Added extension '_50XXX' priority 3 to justek -- Added extension '_350XXX' priority 1 to justek -- Added extension '_350XXX' priority 2 to justek -- Added extension '_350XXX' priority 3 to justek -- Added extension '_350XXX' priority 4 to justek -- Added extension '_350XXX' priority 5 to justek -- Added extension '_350XXX' priority 6 to justek -- Added extension '_350XXX' priority 7 to justek -- Added extension '_350XXX' priority 8 to justek -- Added extension '_350XXX' priority 9 to justek == Parsing '/etc/asterisk/users.conf': Found -- merging incls/swits/igpats from old(default) to new(default) context, registrar = pbx_config -- Including switch 'Lua/' in context 'default' -- Added extension '1234' priority -1 to default -- merging incls/swits/igpats from old(public) to new(public) context, registrar = pbx_config -- Including switch 'Lua/' in context 'public' -- merging incls/swits/igpats from old(demo) to new(demo) context, registrar = pbx_config -- Including switch 'Lua/' in context 'demo' -- Added extension '1000' priority -1 to demo -- Added extension '1001' priority -1 to demo -- merging incls/swits/igpats from old(local) to new(local) context, registrar = pbx_config -- Including switch 'Lua/' in context 'local' -- Registered extension context '__func_periodic_hook_context__'; registrar: func_periodic_hook -- merging incls/swits/igpats from old(__func_periodic_hook_context__) to new(__func_periodic_hook_context__) context, registrar = pbx_config -- Added extension 'beep' priority 2 (CID match '') to __func_periodic_hook_context__ -- Added extension 'beep' priority 1 (CID match '') to __func_periodic_hook_context__ -- Added extension 'hook' priority 6 (CID match '') to __func_periodic_hook_context__ -- Added extension 'hook' priority 5 (CID match '') to __func_periodic_hook_context__ -- Added extension 'hook' priority 4 (CID match '') to __func_periodic_hook_context__ -- Added extension 'hook' priority 3 (CID match '') to __func_periodic_hook_context__ -- Added extension 'hook' priority 2 (CID match '') to __func_periodic_hook_context__ -- Added extension 'hook' priority 1 (CID match '') to __func_periodic_hook_context__ -- Registered extension context 'parkedcalls'; registrar: res_parking -- merging incls/swits/igpats from old(parkedcalls) to new(parkedcalls) context, registrar = pbx_config -- Added extension '700' priority 1 to parkedcalls -- Added extension '701' priority 1 to parkedcalls -- Added extension '702' priority 1 to parkedcalls -- Added extension '703' priority 1 to parkedcalls -- Added extension '704' priority 1 to parkedcalls -- Added extension '705' priority 1 to parkedcalls -- Added extension '706' priority 1 to parkedcalls -- Added extension '707' priority 1 to parkedcalls -- Added extension '708' priority 1 to parkedcalls -- Added extension '709' priority 1 to parkedcalls -- Added extension '710' priority 1 to parkedcalls -- Added extension '711' priority 1 to parkedcalls -- Added extension '712' priority 1 to parkedcalls -- Added extension '713' priority 1 to parkedcalls -- Added extension '714' priority 1 to parkedcalls -- Added extension '715' priority 1 to parkedcalls -- Added extension '716' priority 1 to parkedcalls -- Added extension '717' priority 1 to parkedcalls -- Added extension '718' priority 1 to parkedcalls -- Added extension '719' priority 1 to parkedcalls -- Added extension '720' priority 1 to parkedcalls -- Registered extension context 'ael-builtin-h-bubble'; registrar: pbx_ael -- merging incls/swits/igpats from old(ael-builtin-h-bubble) to new(ael-builtin-h-bubble) context, registrar = pbx_config -- Added extension 'h' priority 9996 to ael-builtin-h-bubble -- Added extension 'h' priority 9995 to ael-builtin-h-bubble -- Added extension 'h' priority 9994 to ael-builtin-h-bubble -- Added extension 'h' priority 9993 to ael-builtin-h-bubble -- Added extension 'h' priority 9992 to ael-builtin-h-bubble -- Added extension 'h' priority 9991 to ael-builtin-h-bubble -- Added extension 'h' priority 1 to ael-builtin-h-bubble -- Registered extension context 'ael-default'; registrar: pbx_ael -- merging incls/swits/igpats from old(ael-default) to new(ael-default) context, registrar = pbx_config -- Including context 'ael-demo' in context 'ael-default' -- Registered extension context 'ael-demo'; registrar: pbx_ael -- merging incls/swits/igpats from old(ael-demo) to new(ael-demo) context, registrar = pbx_config -- Added extension 'i' priority 1 to ael-demo -- Added extension 't' priority 1 to ael-demo -- Added extension '#' priority 2 to ael-demo -- Added extension '#' priority 1 to ael-demo -- Added extension '8500' priority 2 to ael-demo -- Added extension '8500' priority 1 to ael-demo -- Added extension '_1234' priority 1 to ael-demo -- Added extension '600' priority 4 to ael-demo -- Added extension '600' priority 3 to ael-demo -- Added extension '600' priority 2 to ael-demo -- Added extension '600' priority 1 to ael-demo -- Added extension '500' priority 4 to ael-demo -- Added extension '500' priority 3 to ael-demo -- Added extension '500' priority 2 to ael-demo -- Added extension '500' priority 1 to ael-demo -- Added extension '1000' priority 1 to ael-demo -- Added extension '3' priority 2 to ael-demo -- Added extension '3' priority 1 to ael-demo -- Added extension '2' priority 2 to ael-demo -- Added extension '2' priority 1 to ael-demo -- Added extension 's' priority 12 to ael-demo -- Added extension 's' priority 11 to ael-demo -- Added extension 's' priority 10 to ael-demo -- Added extension 's' priority 9 to ael-demo -- Added extension 's' priority 8 to ael-demo -- Added extension 's' priority 7 to ael-demo -- Added extension 's' priority 6 to ael-demo -- Added extension 's' priority 5 to ael-demo -- Added extension 's' priority 4 to ael-demo -- Added extension 's' priority 3 to ael-demo -- Added extension 's' priority 2 to ael-demo -- Added extension 's' priority 1 to ael-demo -- Registered extension context 'ael-std-exten-ael'; registrar: pbx_ael -- merging incls/swits/igpats from old(ael-std-exten-ael) to new(ael-std-exten-ael) context, registrar = pbx_config -- Including context 'ael-builtin-h-bubble' in context 'ael-std-exten-ael' -- Added extension 'sw_1_BUSY' priority 11 to ael-std-exten-ael -- Added extension 'sw_1_BUSY' priority 10 to ael-std-exten-ael -- Added extension 'sw_1_' priority 10 to ael-std-exten-ael -- Added extension '_sw_1_.' priority 11 to ael-std-exten-ael -- Added extension '_sw_1_.' priority 10 to ael-std-exten-ael -- Added extension 'a' priority 2 to ael-std-exten-ael -- Added extension 'a' priority 1 to ael-std-exten-ael -- Added extension '~~s~~' priority 8 to ael-std-exten-ael -- Added extension '~~s~~' priority 7 to ael-std-exten-ael -- Added extension '~~s~~' priority 6 to ael-std-exten-ael -- Added extension '~~s~~' priority 5 to ael-std-exten-ael -- Added extension '~~s~~' priority 4 to ael-std-exten-ael -- Added extension '~~s~~' priority 3 to ael-std-exten-ael -- Added extension '~~s~~' priority 2 to ael-std-exten-ael -- Added extension '~~s~~' priority 1 to ael-std-exten-ael -- Registered extension context 'ael-local'; registrar: pbx_ael -- merging incls/swits/igpats from old(ael-local) to new(ael-local) context, registrar = pbx_config -- Including context 'ael-default' in context 'ael-local' -- Including context 'ael-trunklocal' in context 'ael-local' -- Including context 'ael-iaxtel700' in context 'ael-local' -- Including context 'ael-trunktollfree' in context 'ael-local' -- Including context 'ael-iaxprovider' in context 'ael-local' -- Registered extension context 'ael-longdistance'; registrar: pbx_ael -- merging incls/swits/igpats from old(ael-longdistance) to new(ael-longdistance) context, registrar = pbx_config -- Including context 'ael-local' in context 'ael-longdistance' -- Including context 'ael-trunkld' in context 'ael-longdistance' -- Registered extension context 'ael-international'; registrar: pbx_ael -- merging incls/swits/igpats from old(ael-international) to new(ael-international) context, registrar = pbx_config -- Including context 'ael-longdistance' in context 'ael-international' -- Including context 'ael-trunkint' in context 'ael-international' -- Registered extension context 'ael-trunktollfree'; registrar: pbx_ael -- merging incls/swits/igpats from old(ael-trunktollfree) to new(ael-trunktollfree) context, registrar = pbx_config -- Added extension '_91866NXXXXXX' priority 1 to ael-trunktollfree -- Added extension '_91877NXXXXXX' priority 1 to ael-trunktollfree -- Added extension '_91888NXXXXXX' priority 1 to ael-trunktollfree -- Added extension '_91800NXXXXXX' priority 1 to ael-trunktollfree -- Registered extension context 'ael-trunklocal'; registrar: pbx_ael -- merging incls/swits/igpats from old(ael-trunklocal) to new(ael-trunklocal) context, registrar = pbx_config -- Added extension '_9NXXXXXX' priority 1 to ael-trunklocal -- Registered extension context 'ael-trunkld'; registrar: pbx_ael -- merging incls/swits/igpats from old(ael-trunkld) to new(ael-trunkld) context, registrar = pbx_config -- Including context 'ael-dundi-e164-lookup' in context 'ael-trunkld' -- Added extension '_91NXXNXXXXXX' priority 2 to ael-trunkld -- Added extension '_91NXXNXXXXXX' priority 1 to ael-trunkld -- Registered extension context 'ael-trunkint'; registrar: pbx_ael -- merging incls/swits/igpats from old(ael-trunkint) to new(ael-trunkint) context, registrar = pbx_config -- Including context 'ael-dundi-e164-lookup' in context 'ael-trunkint' -- Added extension '_9011.' priority 2 to ael-trunkint -- Added extension '_9011.' priority 1 to ael-trunkint -- Registered extension context 'ael-iaxprovider'; registrar: pbx_ael -- merging incls/swits/igpats from old(ael-iaxprovider) to new(ael-iaxprovider) context, registrar = pbx_config -- Registered extension context 'ael-iaxtel700'; registrar: pbx_ael -- merging incls/swits/igpats from old(ael-iaxtel700) to new(ael-iaxtel700) context, registrar = pbx_config -- Added extension '_91700XXXXXXX' priority 1 to ael-iaxtel700 -- Registered extension context 'ael-dundi-e164'; registrar: pbx_ael -- merging incls/swits/igpats from old(ael-dundi-e164) to new(ael-dundi-e164) context, registrar = pbx_config -- Including context 'ael-builtin-h-bubble' in context 'ael-dundi-e164' -- Added extension '~~s~~' priority 3 to ael-dundi-e164 -- Added extension '~~s~~' priority 2 to ael-dundi-e164 -- Added extension '~~s~~' priority 1 to ael-dundi-e164 -- Registered extension context 'ael-dundi-e164-lookup'; registrar: pbx_ael -- merging incls/swits/igpats from old(ael-dundi-e164-lookup) to new(ael-dundi-e164-lookup) context, registrar = pbx_config -- Including context 'ael-dundi-e164-local' in context 'ael-dundi-e164-lookup' -- Including context 'ael-dundi-e164-switch' in context 'ael-dundi-e164-lookup' -- Registered extension context 'ael-dundi-e164-switch'; registrar: pbx_ael -- merging incls/swits/igpats from old(ael-dundi-e164-switch) to new(ael-dundi-e164-switch) context, registrar = pbx_config -- Including switch 'DUNDi/e164' in context 'ael-dundi-e164-switch' -- Registered extension context 'ael-dundi-e164-local'; registrar: pbx_ael -- merging incls/swits/igpats from old(ael-dundi-e164-local) to new(ael-dundi-e164-local) context, registrar = pbx_config -- Including context 'ael-dundi-e164-canonical' in context 'ael-dundi-e164-local' -- Including context 'ael-dundi-e164-customers' in context 'ael-dundi-e164-local' -- Including context 'ael-dundi-e164-via-pstn' in context 'ael-dundi-e164-local' -- Registered extension context 'ael-dundi-e164-via-pstn'; registrar: pbx_ael -- merging incls/swits/igpats from old(ael-dundi-e164-via-pstn) to new(ael-dundi-e164-via-pstn) context, registrar = pbx_config -- Registered extension context 'ael-dundi-e164-customers'; registrar: pbx_ael -- merging incls/swits/igpats from old(ael-dundi-e164-customers) to new(ael-dundi-e164-customers) context, registrar = pbx_config -- Registered extension context 'ael-dundi-e164-canonical'; registrar: pbx_ael -- merging incls/swits/igpats from old(ael-dundi-e164-canonical) to new(ael-dundi-e164-canonical) context, registrar = pbx_config -- Time to scan old dialplan and merge leftovers back into the new: 0.515559 sec -- Time to restore hints and swap in new dialplan: 0.000683 sec -- Time to delete the old dialplan: 0.017101 sec -- Total time merge_contexts_delete: 0.533343 sec == pbx_config.so => (Text Extension Configuration) Loading pbx_spool.so. == pbx_spool.so => (Outgoing Spool Support) Loading app_zapateller.so. == Registered application 'Zapateller' == app_zapateller.so => (Block Telemarketers with Special Information Tone) Loading app_readexten.so. == Registered application 'ReadExten' == app_readexten.so => (Read and evaluate extension validity) Loading res_ari_events.so. == WebSocket registered sub-protocol 'ari' == res_ari_events.so => (RESTful API module - WebSocket resource) Loading func_pjsip_aor.so. == Registered custom function 'PJSIP_AOR' == func_pjsip_aor.so => (Get information about a PJSIP AOR) Loading func_aes.so. == Registered custom function 'AES_DECRYPT' == Registered custom function 'AES_ENCRYPT' == func_aes.so => (AES dialplan functions) Loading func_channel.so. == Registered custom function 'CHANNEL' == Registered custom function 'CHANNELS' == Registered custom function 'MASTER_CHANNEL' == func_channel.so => (Channel information dialplan functions) Loading res_ari_playbacks.so. == res_ari_playbacks.so => (RESTful API module - Playback control resources) Loading app_disa.so. == Registered application 'DISA' == app_disa.so => (DISA (Direct Inward System Access) Application) Loading func_volume.so. == Registered custom function 'VOLUME' == func_volume.so => (Technology independent volume control) Loading app_waitforsilence.so. == Registered application 'WaitForSilence' == Registered application 'WaitForNoise' == app_waitforsilence.so => (Wait For Silence) Loading app_dictate.so. == Registered application 'Dictate' == app_dictate.so => (Virtual Dictation Machine) Loading codec_lpc10.so. == Registered translator 'lpc10tolin' from codec lpc10 to slin, table cost, 900000, computational cost 142979 == Registered translator 'lintolpc10' from codec slin to lpc10, table cost, 600000, computational cost 264960 == codec_lpc10.so => (LPC10 2.4kbps Coder/Decoder) Loading res_mutestream.so. == Registered custom function 'MUTEAUDIO' == Manager registered action MuteAudio == res_mutestream.so => (Mute audio stream resources) Loading func_groupcount.so. == 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) Loading chan_phone.so. == Parsing '/etc/asterisk/phone.conf': Found == Registered channel type 'Phone' (Standard Linux Telephony API Driver) == chan_phone.so => (Linux Telephony API Support) Loading codec_ulaw.so. == Registered translator 'ulawtolin' from codec ulaw to slin, table cost, 900000, computational cost 22996 == Registered translator 'lintoulaw' from codec slin to ulaw, table cost, 600000, computational cost 20997 == Registered translator 'lintotestlaw' from codec slin to testlaw, table cost, 600000, computational cost 19997 == Registered translator 'testlawtolin' from codec testlaw to slin, table cost, 900000, computational cost 18998 == codec_ulaw.so => (mu-Law Coder/Decoder) Loading func_config.so. == Registered custom function 'AST_CONFIG' == func_config.so => (Asterisk configuration file variable access) Loading func_frame_trace.so. == Registered custom function 'FRAME_TRACE' == func_frame_trace.so => (Frame Trace for internal ast_frame debugging.) Loading func_env.so. == 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) Loading func_blacklist.so. == Registered custom function 'BLACKLIST' == func_blacklist.so => (Look up Caller*ID name/number from blacklist database) Loading app_while.so. == Registered application 'While' == Registered application 'EndWhile' == Registered application 'ExitWhile' == Registered application 'ContinueWhile' == app_while.so => (While Loops and Conditional Execution) Loading res_limit.so. == res_limit.so => (Resource limits) Loading app_dial.so. == Registered application 'Dial' == Registered application 'RetryDial' == app_dial.so => (Dialing Application) Loading func_pjsip_contact.so. == Registered custom function 'PJSIP_CONTACT' == func_pjsip_contact.so => (Get information about a PJSIP contact) Loading app_speech_utils.so. == 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) Loading app_sayunixtime.so. == Registered application 'SayUnixTime' == Registered application 'DateTime' == app_sayunixtime.so => (Say time) Loading func_hangupcause.so. == Registered custom function 'HANGUPCAUSE' == Registered custom function 'HANGUPCAUSE_KEYS' == Registered application 'HangupCauseClear' == func_hangupcause.so => (HANGUPCAUSE related functions and applications) Loading func_pitchshift.so. == Registered custom function 'PITCH_SHIFT' == func_pitchshift.so => (Audio Effects Dialplan Functions) Loading func_enum.so. == Registered custom function 'ENUMRESULT' == Registered custom function 'ENUMQUERY' == Registered custom function 'ENUMLOOKUP' == Registered custom function 'TXTCIDNAME' == func_enum.so => (ENUM related dialplan functions) Loading func_callcompletion.so. == Registered custom function 'CALLCOMPLETION' == func_callcompletion.so => (Call Control Configuration Function) Loading app_bridgewait.so. == Registered application 'BridgeWait' == app_bridgewait.so => (Place the channel into a holding bridge application) Loading app_sendtext.so. == Registered application 'SendText' == app_sendtext.so => (Send Text Applications) Loading func_srv.so. == Registered custom function 'SRVQUERY' == Registered custom function 'SRVRESULT' == func_srv.so => (SRV related dialplan functions) Loading app_externalivr.so. == Registered application 'ExternalIVR' == app_externalivr.so => (External IVR Interface Application) Loading app_sms.so. == Registered application 'SMS' == app_sms.so => (SMS/PSTN handler) Loading func_pjsip_endpoint.so. == Registered custom function 'PJSIP_ENDPOINT' == func_pjsip_endpoint.so => (Get information about a PJSIP endpoint) Loading func_base64.so. == Registered custom function 'BASE64_ENCODE' == Registered custom function 'BASE64_DECODE' == func_base64.so => (base64 encode/decode dialplan functions) Loading func_module.so. == Registered custom function 'IFMODULE' == func_module.so => (Checks if Asterisk module is loaded in memory) Loading bridge_native_rtp.so. == Registered bridge technology native_rtp == bridge_native_rtp.so => (Native RTP bridging module) Loading res_hep_pjsip.so. == res_hep_pjsip.so => (PJSIP HEPv3 Logger) Loading res_ari_sounds.so. == res_ari_sounds.so => (RESTful API module - Sound resources) Loading app_stasis.so. == Registered application 'Stasis' == app_stasis.so => (Stasis dialplan application) Loading func_callerid.so. == 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)) Loading func_md5.so. == Registered custom function 'MD5' == func_md5.so => (MD5 digest dialplan functions) Loading func_realtime.so. == 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) Loading func_speex.so. == Registered custom function 'AGC' == Registered custom function 'DENOISE' == func_speex.so => (Noise reduction and Automatic Gain Control (AGC)) Loading func_logic.so. == 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) Loading func_talkdetect.so. == Registered custom function 'TALK_DETECT' == func_talkdetect.so => (Talk detection dialplan function) Loading app_verbose.so. == Registered application 'Log' == Registered application 'Verbose' == app_verbose.so => (Send verbose output) Loading codec_speex.so. == Parsing '/etc/asterisk/codecs.conf': Found -- CODEC SPEEX: Setting Quality to 3 -- CODEC SPEEX: Setting Complexity to 2 -- CODEC SPEEX: Perceptual Enhancement Mode. [on] -- CODEC SPEEX: VAD Mode. [on] -- CODEC SPEEX: VBR Mode. [on] -- CODEC SPEEX: Disabling ABR -- CODEC SPEEX: Setting VBR Quality to 4.000000 -- CODEC SPEEX: DTX Mode. [off] -- CODEC SPEEX: Preprocessing. [off] -- CODEC SPEEX: Preprocessor VAD. [off] -- CODEC SPEEX: Preprocessor AGC. [off] -- CODEC SPEEX: Setting preprocessor AGC Level to 8000.000000 -- CODEC SPEEX: Preprocessor Denoise. [off] -- CODEC SPEEX: Preprocessor Dereverb. [off] -- CODEC SPEEX: Setting preprocessor Dereverb Decay to 0.400000 -- CODEC SPEEX: Setting preprocessor Dereverb Level to 0.300000 == Registered translator 'speextolin' from codec speex to slin, table cost, 900000, computational cost 180972 == Registered translator 'lintospeex' from codec slin to speex, table cost, 600000, computational cost 411937 == Registered translator 'speexwbtolin16' from codec speex to slin, table cost, 900000, computational cost 269959 == Registered translator 'lin16tospeexwb' from codec slin to speex, table cost, 600000, computational cost 460931 == Registered translator 'speexuwbtolin32' from codec speex to slin, table cost, 900000, computational cost 999999 == Registered translator 'lin32tospeexuwb' from codec slin to speex, table cost, 600000, computational cost 999999 == codec_speex.so => (Speex Coder/Decoder) Loading res_ari_endpoints.so. == res_ari_endpoints.so => (RESTful API module - Endpoint resources) Loading bridge_simple.so. == Registered bridge technology simple_bridge == bridge_simple.so => (Simple two channel bridging module) Loading func_uri.so. == Registered custom function 'URIDECODE' == Registered custom function 'URIENCODE' == func_uri.so => (URI encode/decode dialplan functions) Loading res_fax_spandsp.so. -- Registered handler for 'Spandsp' (Spandsp FAX Driver) == res_fax_spandsp.so => (Spandsp G.711 and T.38 FAX Technologies) Loading func_dialgroup.so. == Registered custom function 'DIALGROUP' == func_dialgroup.so => (Dialgroup dialplan function) Loading bridge_holding.so. == Registered bridge technology holding_bridge == bridge_holding.so => (Holding bridge module) Loading app_followme.so. == Parsing '/etc/asterisk/followme.conf': Found == Registered application 'FollowMe' == app_followme.so => (Find-Me/Follow-Me Application) Loading app_playback.so. == Parsing '/etc/asterisk/say.conf': Found == Registered application 'Playback' == app_playback.so => (Sound File Playback Application) Loading bridge_softmix.so. == Registered bridge technology softmix == bridge_softmix.so => (Multi-party software based channel mixing) Loading func_cdr.so. == Registered custom function 'CDR' == Registered custom function 'CDR_PROP' == func_cdr.so => (Call Detail Record (CDR) dialplan functions) Loading app_test.so. == Registered application 'TestClient' == Registered application 'TestServer' == app_test.so => (Interface Test Application) Loading codec_gsm.so. == Registered translator 'gsmtolin' from codec gsm to slin, table cost, 900000, computational cost 50992 == Registered translator 'lintogsm' from codec slin to gsm, table cost, 600000, computational cost 106984 == codec_gsm.so => (GSM Coder/Decoder) Loading func_sprintf.so. == Registered custom function 'SPRINTF' == func_sprintf.so => (SPRINTF dialplan function) Loading app_milliwatt.so. == Registered application 'Milliwatt' == app_milliwatt.so => (Digital Milliwatt (mu-law) Test Application) Loading res_format_attr_h264.so. == Registered format interface for codec 'h264' == res_format_attr_h264.so => (H.264 Format Attribute Module) Loading func_holdintercept.so. == Registered custom function 'HOLD_INTERCEPT' == func_holdintercept.so => (Hold interception dialplan function) Loading app_controlplayback.so. == Registered application 'ControlPlayback' == Manager registered action ControlPlayback == app_controlplayback.so => (Control Playback Application) Loading func_cut.so. == Registered custom function 'CUT' == Registered custom function 'SORT' == func_cut.so => (Cut out information from a string) Loading app_record.so. == Registered application 'Record' == app_record.so => (Trivial Record Application) Loading app_chanisavail.so. == Registered application 'ChanIsAvail' == app_chanisavail.so => (Check channel availability) Loading app_dumpchan.so. == Registered application 'DumpChan' == app_dumpchan.so => (Dump Info About The Calling Channel) Loading func_extstate.so. == Registered custom function 'EXTENSION_STATE' == func_extstate.so => (Gets an extension's state in the dialplan) Loading func_strings.so. == Registered custom function 'FIELDQTY' == Registered custom function 'FIELDNUM' == Registered custom function 'FILTER' == Registered custom function 'REPLACE' == Registered custom function 'STRREPLACE' == 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) Loading res_ari_recordings.so. == res_ari_recordings.so => (RESTful API module - Recording resources) Loading app_echo.so. == Registered application 'Echo' == app_echo.so => (Simple Echo Application) Loading app_transfer.so. == Registered application 'Transfer' == app_transfer.so => (Transfers a caller to another extension) Loading func_db.so. == Registered custom function 'DB' == Registered custom function 'DB_EXISTS' == Registered custom function 'DB_DELETE' == Registered custom function 'DB_KEYS' == func_db.so => (Database (astdb) related dialplan functions) Loading res_hep_rtcp.so. == res_hep_rtcp.so => (RTCP HEPv3 Logger) Loading res_ari_channels.so. == res_ari_channels.so => (RESTful API module - Channel resources) Loading bridge_builtin_interval_features.so. == bridge_builtin_interval_features.so => (Built in bridging interval features) Loading codec_adpcm.so. == Registered translator 'adpcmtolin' from codec adpcm to slin, table cost, 900000, computational cost 34995 == Registered translator 'lintoadpcm' from codec slin to adpcm, table cost, 600000, computational cost 40994 == codec_adpcm.so => (Adaptive Differential PCM Coder/Decoder) Loading res_security_log.so. -- Security Logging Enabled == res_security_log.so => (Security Event Logging) Loading res_clialiases.so. == 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 intense span' == Aliased CLI command 'reload' to 'module reload' == Aliased CLI command 'pjsip reload' to 'module reload res_pjsip.so res_pjsip_authenticator_digest.so res_pjsip_endpoint_identifier_ip.so res_pjsip_mwi.so res_pjsip_notify.so res_pjsip_outbound_publish.so res_pjsip_publish_asterisk.so res_pjsip_outbound_registration.so' == res_clialiases.so => (CLI Aliases) Loading app_celgenuserevent.so. == Registered application 'CELGenUserEvent' == app_celgenuserevent.so => (Generate an User-Defined CEL event) Loading pbx_loopback.so. == pbx_loopback.so => (Loopback Switch) Loading res_ari_bridges.so. == res_ari_bridges.so => (RESTful API module - Bridge resources) Loading app_cdr.so. == Registered application 'NoCDR' == Registered application 'ResetCDR' == app_cdr.so => (Tell Asterisk to not maintain a CDR for the current call) Loading res_ari_device_states.so. == res_ari_device_states.so => (RESTful API module - Device state resources) Loading app_softhangup.so. == Registered application 'SoftHangup' == app_softhangup.so => (Hangs up the requested channel) Loading app_alarmreceiver.so. == Parsing '/etc/asterisk/alarmreceiver.conf': Found == Registered application 'AlarmReceiver' == app_alarmreceiver.so => (Alarm Receiver for Asterisk) Loading app_mixmonitor.so. == Registered application 'MixMonitor' == Registered application 'StopMixMonitor' == Manager registered action MixMonitorMute == Manager registered action MixMonitor == Manager registered action StopMixMonitor == Registered custom function 'MIXMONITOR' == app_mixmonitor.so => (Mixed Audio Monitoring Application) Loading app_authenticate.so. == Registered application 'Authenticate' == app_authenticate.so => (Authentication Application) Loading app_userevent.so. == Registered application 'UserEvent' == app_userevent.so => (Custom User Event Application) Loading app_privacy.so. == Registered application 'PrivacyManager' == app_privacy.so => (Require phone number to be entered, if no CallerID sent) Loading chan_bridge_media.so. == Registered channel type 'Announcer' (Bridge Media Announcing Channel Driver) == Registered channel type 'Recorder' (Bridge Media Recording Channel Driver) == chan_bridge_media.so => (Bridge Media Channel Driver) Loading app_directory.so. == Registered application 'Directory' == app_directory.so => (Extension Directory) Loading func_lock.so. == Registered custom function 'LOCK' == Registered custom function 'TRYLOCK' == Registered custom function 'UNLOCK' == func_lock.so => (Dialplan mutexes) Loading app_read.so. == Registered application 'Read' == app_read.so => (Read Variable Application) Loading codec_resample.so. == Registered translator 'slin 8000khz -> 12000khz' from codec slin to slin, table cost, 800000, computational cost 999999 == Registered translator 'slin 8000khz -> 16000khz' from codec slin to slin, table cost, 800000, computational cost 999999 == Registered translator 'slin 8000khz -> 24000khz' from codec slin to slin, table cost, 800000, computational cost 999999 == Registered translator 'slin 8000khz -> 32000khz' from codec slin to slin, table cost, 800000, computational cost 999999 == Registered translator 'slin 8000khz -> 44100khz' from codec slin to slin, table cost, 800000, computational cost 999999 == Registered translator 'slin 8000khz -> 48000khz' from codec slin to slin, table cost, 800000, computational cost 999999 == Registered translator 'slin 8000khz -> 96000khz' from codec slin to slin, table cost, 800000, computational cost 999999 == Registered translator 'slin 8000khz -> 192000khz' from codec slin to slin, table cost, 800000, computational cost 999999 == Registered translator 'slin 12000khz -> 8000khz' from codec slin to slin, table cost, 850000, computational cost 999999 == Registered translator 'slin 12000khz -> 16000khz' from codec slin to slin, table cost, 800000, computational cost 999999 == Registered translator 'slin 12000khz -> 24000khz' from codec slin to slin, table cost, 800000, computational cost 999999 == Registered translator 'slin 12000khz -> 32000khz' from codec slin to slin, table cost, 800000, computational cost 999999 == Registered translator 'slin 12000khz -> 44100khz' from codec slin to slin, table cost, 800000, computational cost 999999 == Registered translator 'slin 12000khz -> 48000khz' from codec slin to slin, table cost, 800000, computational cost 999999 == Registered translator 'slin 12000khz -> 96000khz' from codec slin to slin, table cost, 800000, computational cost 999999 == Registered translator 'slin 12000khz -> 192000khz' from codec slin to slin, table cost, 800000, computational cost 999999 == Registered translator 'slin 16000khz -> 8000khz' from codec slin to slin, table cost, 850000, computational cost 999999 == Registered translator 'slin 16000khz -> 12000khz' from codec slin to slin, table cost, 850000, computational cost 999999 == Registered translator 'slin 16000khz -> 24000khz' from codec slin to slin, table cost, 800000, computational cost 999999 == Registered translator 'slin 16000khz -> 32000khz' from codec slin to slin, table cost, 800000, computational cost 999999 == Registered translator 'slin 16000khz -> 44100khz' from codec slin to slin, table cost, 800000, computational cost 999999 == Registered translator 'slin 16000khz -> 48000khz' from codec slin to slin, table cost, 800000, computational cost 999999 == Registered translator 'slin 16000khz -> 96000khz' from codec slin to slin, table cost, 800000, computational cost 999999 == Registered translator 'slin 16000khz -> 192000khz' from codec slin to slin, table cost, 800000, computational cost 999999 == Registered translator 'slin 24000khz -> 8000khz' from codec slin to slin, table cost, 850000, computational cost 999999 == Registered translator 'slin 24000khz -> 12000khz' from codec slin to slin, table cost, 850000, computational cost 999999 == Registered translator 'slin 24000khz -> 16000khz' from codec slin to slin, table cost, 850000, computational cost 999999 == Registered translator 'slin 24000khz -> 32000khz' from codec slin to slin, table cost, 800000, computational cost 999999 == Registered translator 'slin 24000khz -> 44100khz' from codec slin to slin, table cost, 800000, computational cost 999999 == Registered translator 'slin 24000khz -> 48000khz' from codec slin to slin, table cost, 800000, computational cost 999999 == Registered translator 'slin 24000khz -> 96000khz' from codec slin to slin, table cost, 800000, computational cost 999999 == Registered translator 'slin 24000khz -> 192000khz' from codec slin to slin, table cost, 800000, computational cost 999999 == Registered translator 'slin 32000khz -> 8000khz' from codec slin to slin, table cost, 850000, computational cost 999999 == Registered translator 'slin 32000khz -> 12000khz' from codec slin to slin, table cost, 850000, computational cost 999999 == Registered translator 'slin 32000khz -> 16000khz' from codec slin to slin, table cost, 850000, computational cost 999999 == Registered translator 'slin 32000khz -> 24000khz' from codec slin to slin, table cost, 850000, computational cost 999999 == Registered translator 'slin 32000khz -> 44100khz' from codec slin to slin, table cost, 800000, computational cost 999999 == Registered translator 'slin 32000khz -> 48000khz' from codec slin to slin, table cost, 800000, computational cost 999999 == Registered translator 'slin 32000khz -> 96000khz' from codec slin to slin, table cost, 800000, computational cost 999999 == Registered translator 'slin 32000khz -> 192000khz' from codec slin to slin, table cost, 800000, computational cost 999999 == Registered translator 'slin 44100khz -> 8000khz' from codec slin to slin, table cost, 850000, computational cost 999999 == Registered translator 'slin 44100khz -> 12000khz' from codec slin to slin, table cost, 850000, computational cost 999999 == Registered translator 'slin 44100khz -> 16000khz' from codec slin to slin, table cost, 850000, computational cost 999999 == Registered translator 'slin 44100khz -> 24000khz' from codec slin to slin, table cost, 850000, computational cost 999999 == Registered translator 'slin 44100khz -> 32000khz' from codec slin to slin, table cost, 850000, computational cost 999999 == Registered translator 'slin 44100khz -> 48000khz' from codec slin to slin, table cost, 800000, computational cost 999999 == Registered translator 'slin 44100khz -> 96000khz' from codec slin to slin, table cost, 800000, computational cost 999999 == Registered translator 'slin 44100khz -> 192000khz' from codec slin to slin, table cost, 800000, computational cost 999999 == Registered translator 'slin 48000khz -> 8000khz' from codec slin to slin, table cost, 850000, computational cost 999999 == Registered translator 'slin 48000khz -> 12000khz' from codec slin to slin, table cost, 850000, computational cost 999999 == Registered translator 'slin 48000khz -> 16000khz' from codec slin to slin, table cost, 850000, computational cost 999999 == Registered translator 'slin 48000khz -> 24000khz' from codec slin to slin, table cost, 850000, computational cost 999999 == Registered translator 'slin 48000khz -> 32000khz' from codec slin to slin, table cost, 850000, computational cost 999999 == Registered translator 'slin 48000khz -> 44100khz' from codec slin to slin, table cost, 850000, computational cost 999999 == Registered translator 'slin 48000khz -> 96000khz' from codec slin to slin, table cost, 800000, computational cost 999999 == Registered translator 'slin 48000khz -> 192000khz' from codec slin to slin, table cost, 800000, computational cost 999999 == Registered translator 'slin 96000khz -> 8000khz' from codec slin to slin, table cost, 850000, computational cost 999999 == Registered translator 'slin 96000khz -> 12000khz' from codec slin to slin, table cost, 850000, computational cost 999999 == Registered translator 'slin 96000khz -> 16000khz' from codec slin to slin, table cost, 850000, computational cost 999999 == Registered translator 'slin 96000khz -> 24000khz' from codec slin to slin, table cost, 850000, computational cost 999999 == Registered translator 'slin 96000khz -> 32000khz' from codec slin to slin, table cost, 850000, computational cost 999999 == Registered translator 'slin 96000khz -> 44100khz' from codec slin to slin, table cost, 850000, computational cost 999999 == Registered translator 'slin 96000khz -> 48000khz' from codec slin to slin, table cost, 850000, computational cost 999999 == Registered translator 'slin 96000khz -> 192000khz' from codec slin to slin, table cost, 800000, computational cost 999999 == Registered translator 'slin 192000khz -> 8000khz' from codec slin to slin, table cost, 850000, computational cost 999999 == Registered translator 'slin 192000khz -> 12000khz' from codec slin to slin, table cost, 850000, computational cost 999999 == Registered translator 'slin 192000khz -> 16000khz' from codec slin to slin, table cost, 850000, computational cost 999999 == Registered translator 'slin 192000khz -> 24000khz' from codec slin to slin, table cost, 850000, computational cost 999999 == Registered translator 'slin 192000khz -> 32000khz' from codec slin to slin, table cost, 850000, computational cost 999999 == Registered translator 'slin 192000khz -> 44100khz' from codec slin to slin, table cost, 850000, computational cost 999999 == Registered translator 'slin 192000khz -> 48000khz' from codec slin to slin, table cost, 850000, computational cost 999999 == Registered translator 'slin 192000khz -> 96000khz' from codec slin to slin, table cost, 850000, computational cost 999999 == codec_resample.so => (SLIN Resampling Codec) Loading res_snmp.so. == Parsing '/etc/asterisk/res_snmp.conf': Found Loading [Sub]Agent Module == res_snmp.so => (SNMP [Sub]Agent for Asterisk) Loading app_queue.so. == Parsing '/etc/asterisk/queuerules.conf': Found == Parsing '/etc/asterisk/queues.conf': Found == Registered application 'Queue' == Registered application 'AddQueueMember' == Registered application 'RemoveQueueMember' == Registered application 'PauseQueueMember' == Registered application 'UnpauseQueueMember' == Registered application 'QueueLog' == Manager registered action Queues == Manager registered action QueueStatus == Manager registered action QueueSummary == Manager registered action QueueAdd == Manager registered action QueueRemove == Manager registered action QueuePause == Manager registered action QueueLog == Manager registered action QueuePenalty == Manager registered action QueueMemberRingInUse == Manager registered action QueueRule == Manager registered action QueueReload == Manager registered action QueueReset == Registered custom function 'QUEUE_VARIABLES' == Registered custom function 'QUEUE_EXISTS' == 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) Loading res_manager_devicestate.so. == Manager registered action DeviceStateList == res_manager_devicestate.so => (Manager Device State Topic Forwarder) Loading res_manager_presencestate.so. == Manager registered action PresenceStateList == res_manager_presencestate.so => (Manager Presence State Topic Forwarder) == Parsing '/etc/asterisk/cli_permissions.conf': Found == Parsing '/etc/asterisk/cli.conf': Found Asterisk Ready. *CLI> == Using SIP VIDEO CoS mark 6 == Using SIP RTP CoS mark 5 -- Executing [350001@default:1] Playback("SIP/50001-00000000", "beep") in new stack > 0x47ddd9a0 -- Probation passed - setting RTP source address to 192.168.1.147:10593 -- Playing 'beep.gsm' (language 'en') -- Executing [350001@default:2] Answer("SIP/50001-00000000", "") in new stack > 0x47ddd9a0 -- Probation passed - setting RTP source address to 192.168.1.147:10593 -- Executing [350001@default:3] Set("SIP/50001-00000000", "CID=350001") in new stack -- Executing [350001@default:4] GotoIf("SIP/50001-00000000", "1?cme") in new stack -- Goto (default,350001,8) -- Executing [350001@default:8] ConfBridge("SIP/50001-00000000", "350001,default_bridge_32,default_useradmin,sample_user_menu") in new stack -- Channel CBAnn/350001-00000000;2 joined 'softmix' base-bridge -- Channel SIP/50001-00000000 joined 'softmix' base-bridge -- Playing 'confbridge-join.gsm' (language 'en') *CLI> -- Channel SIP/50001-00000000 left 'softmix' base-bridge *CLI> -- Playing 'confbridge-leave.gsm' (language 'en') -- Channel CBAnn/350001-00000000;2 left 'softmix' base-bridge *CLI> sip set debug on SIP Debugging enabled *CLI> core set verbose 10 <--- SIP read from UDP:192.168.1.147:43521 ---> <-------------> Console verbose was 4 and is now 10. *CLI> core set debug 10 Core debug was OFF and is now 10. *CLI> <--- SIP read from UDP:192.168.1.147:23681 ---> <-------------> [Feb 20 15:23:31] DEBUG[32544]: chan_sip.c:9893 parse_request: Header 0 [ 0]: <--- SIP read from UDP:192.168.1.147:43521 ---> INVITE sip:350001@192.168.1.227 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.147:43521;branch=z9hG4bK.9vxdzeByK;rport From: ;tag=cdYXYwnFg To: sip:350001@192.168.1.227 CSeq: 20 INVITE Call-ID: dV~eC~AMO2 Max-Forwards: 70 Supported: outbound Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO, UPDATE Content-Type: application/sdp Content-Length: 335 Contact: ;+sip.instance="" User-Agent: Linphone/3.9.1 (belle-sip/1.4.2) v=0 o=50001 4045 4054 IN IP4 192.168.41.102 s=Talk c=IN IP4 192.168.41.102 b=AS:200 t=0 0 a=rtcp-xr:rcvr-rtt=all:10000 stat-summary=loss,dup,jitt,TTL voip-metrics m=audio 7080 RTP/AVP 96 101 a=rtpmap:96 opus/48000/2 a=fmtp:96 useinbandfec=1 a=rtpmap:101 telephone-event/48000 m=video 9078 RTP/AVP 96 a=rtpmap:96 VP8/90000 <-------------> [Feb 20 15:23:35] DEBUG[32544]: chan_sip.c:9893 parse_request: Header 0 [ 40]: INVITE sip:350001@192.168.1.227 SIP/2.0 [Feb 20 15:23:35] DEBUG[32544]: chan_sip.c:9893 parse_request: Header 1 [ 69]: Via: SIP/2.0/UDP 192.168.1.147:43521;branch=z9hG4bK.9vxdzeByK;rport [Feb 20 15:23:35] DEBUG[32544]: chan_sip.c:9893 parse_request: Header 2 [ 46]: From: ;tag=cdYXYwnFg [Feb 20 15:23:35] DEBUG[32544]: chan_sip.c:9893 parse_request: Header 3 [ 29]: To: sip:350001@192.168.1.227 [Feb 20 15:23:35] DEBUG[32544]: chan_sip.c:9893 parse_request: Header 4 [ 15]: CSeq: 20 INVITE [Feb 20 15:23:35] DEBUG[32544]: chan_sip.c:9893 parse_request: Header 5 [ 19]: Call-ID: dV~eC~AMO2 [Feb 20 15:23:35] DEBUG[32544]: chan_sip.c:9893 parse_request: Header 6 [ 16]: Max-Forwards: 70 [Feb 20 15:23:35] DEBUG[32544]: chan_sip.c:9893 parse_request: Header 7 [ 19]: Supported: outbound [Feb 20 15:23:35] DEBUG[32544]: chan_sip.c:9893 parse_request: Header 8 [ 89]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO, UPDATE [Feb 20 15:23:35] DEBUG[32544]: chan_sip.c:9893 parse_request: Header 9 [ 29]: Content-Type: application/sdp [Feb 20 15:23:35] DEBUG[32544]: chan_sip.c:9893 parse_request: Header 10 [ 19]: Content-Length: 335 [Feb 20 15:23:35] DEBUG[32544]: chan_sip.c:9893 parse_request: Header 11 [106]: Contact: ;+sip.instance="" [Feb 20 15:23:35] DEBUG[32544]: chan_sip.c:9893 parse_request: Header 12 [ 44]: User-Agent: Linphone/3.9.1 (belle-sip/1.4.2) [Feb 20 15:23:35] DEBUG[32544]: chan_sip.c:9893 parse_request: Header 13 [ 0]: [Feb 20 15:23:35] DEBUG[32544]: chan_sip.c:9893 parse_request: Body 0 [ 3]: v=0 [Feb 20 15:23:35] DEBUG[32544]: chan_sip.c:9893 parse_request: Body 1 [ 39]: o=50001 4045 4054 IN IP4 192.168.41.102 [Feb 20 15:23:35] DEBUG[32544]: chan_sip.c:9893 parse_request: Body 2 [ 6]: s=Talk [Feb 20 15:23:35] DEBUG[32544]: chan_sip.c:9893 parse_request: Body 3 [ 23]: c=IN IP4 192.168.41.102 [Feb 20 15:23:35] DEBUG[32544]: chan_sip.c:9893 parse_request: Body 4 [ 8]: b=AS:200 [Feb 20 15:23:35] DEBUG[32544]: chan_sip.c:9893 parse_request: Body 5 [ 5]: t=0 0 [Feb 20 15:23:35] DEBUG[32544]: chan_sip.c:9893 parse_request: Body 6 [ 72]: a=rtcp-xr:rcvr-rtt=all:10000 stat-summary=loss,dup,jitt,TTL voip-metrics [Feb 20 15:23:35] DEBUG[32544]: chan_sip.c:9893 parse_request: Body 7 [ 27]: m=audio 7080 RTP/AVP 96 101 [Feb 20 15:23:35] DEBUG[32544]: chan_sip.c:9893 parse_request: Body 8 [ 24]: a=rtpmap:96 opus/48000/2 [Feb 20 15:23:35] DEBUG[32544]: chan_sip.c:9893 parse_request: Body 9 [ 24]: a=fmtp:96 useinbandfec=1 [Feb 20 15:23:35] DEBUG[32544]: chan_sip.c:9893 parse_request: Body 10 [ 34]: a=rtpmap:101 telephone-event/48000 [Feb 20 15:23:35] DEBUG[32544]: chan_sip.c:9893 parse_request: Body 11 [ 23]: m=video 9078 RTP/AVP 96 [Feb 20 15:23:35] DEBUG[32544]: chan_sip.c:9930 parse_request: Body 12 [ 21]: a=rtpmap:96 VP8/90000 --- (13 headers 13 lines) --- [Feb 20 15:23:35] DEBUG[32544]: chan_sip.c:9419 __find_call: = Looking for Call ID: dV~eC~AMO2 (Checking From) --From tag cdYXYwnFg --To-tag [Feb 20 15:23:35] DEBUG[32544]: acl.c:957 ast_ouraddrfor: For destination '192.168.1.147', our source address is '192.168.41.78'. [Feb 20 15:23:35] DEBUG[32544]: chan_sip.c:3877 ast_sip_ouraddrfor: Target address 192.168.1.147:43521 is not local, substituting externaddr [Feb 20 15:23:35] DEBUG[32544]: chan_sip.c:3910 ast_sip_ouraddrfor: Setting AST_TRANSPORT_UDP with address 192.168.1.227:5060 [Feb 20 15:23:35] DEBUG[32544]: netsock2.c:172 ast_sockaddr_split_hostport: Splitting '192.168.1.147:43521' into... [Feb 20 15:23:35] DEBUG[32544]: netsock2.c:226 ast_sockaddr_split_hostport: ...host '192.168.1.147' and port '43521'. Sending to 192.168.1.147:43521 (no NAT) [Feb 20 15:23:35] DEBUG[32544]: chan_sip.c:9006 __sip_alloc: Allocating new SIP dialog for dV~eC~AMO2 - INVITE (No RTP) [Feb 20 15:23:35] DEBUG[32544][C-00000001]: chan_sip.c:28652 handle_incoming: **** Received INVITE (5) - Command in SIP INVITE [Feb 20 15:23:35] DEBUG[32544][C-00000001]: sip/reqresp_parser.c:1711 parse_sip_options: Begin: parsing SIP "Supported: outbound" [Feb 20 15:23:35] DEBUG[32544][C-00000001]: sip/reqresp_parser.c:1726 parse_sip_options: Found SIP option: -outbound- [Feb 20 15:23:35] DEBUG[32544][C-00000001]: sip/reqresp_parser.c:1734 parse_sip_options: Matched SIP option: outbound [Feb 20 15:23:35] DEBUG[32544][C-00000001]: netsock2.c:172 ast_sockaddr_split_hostport: Splitting '192.168.1.147:43521' into... [Feb 20 15:23:35] DEBUG[32544][C-00000001]: netsock2.c:226 ast_sockaddr_split_hostport: ...host '192.168.1.147' and port '43521'. Sending to 192.168.1.147:43521 (no NAT) [Feb 20 15:23:35] DEBUG[32544][C-00000001]: chan_sip.c:26104 handle_request_invite: Initializing initreq for method INVITE - callid dV~eC~AMO2 Using INVITE request as basis request - dV~eC~AMO2 [Feb 20 15:23:35] DEBUG[32544][C-00000001]: netsock2.c:172 ast_sockaddr_split_hostport: Splitting '192.168.1.227' into... [Feb 20 15:23:35] DEBUG[32544][C-00000001]: netsock2.c:226 ast_sockaddr_split_hostport: ...host '192.168.1.227' and port ''. Found peer '50001' for '50001' from 192.168.1.147:43521 <--- Reliably Transmitting (no NAT) to 192.168.1.147:43521 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.1.147:43521;branch=z9hG4bK.9vxdzeByK;received=192.168.1.147;rport=43521 From: ;tag=cdYXYwnFg To: sip:350001@192.168.1.227;tag=as48abe0a9 Call-ID: dV~eC~AMO2 CSeq: 20 INVITE Server: Asterisk PBX 13.14.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="2bc3b02a" Content-Length: 0 <------------> [Feb 20 15:23:35] DEBUG[32544][C-00000001]: chan_sip.c:4266 __sip_reliable_xmit: *** SIP TIMER: Initializing retransmit timer on packet: Id #3 [Feb 20 15:23:35] DEBUG[32544][C-00000001]: chan_sip.c:3753 __sip_xmit: Trying to put 'SIP/2.0 401' onto UDP socket destined for 192.168.1.147:43521 Scheduling destruction of SIP dialog 'dV~eC~AMO2' in 32000 ms (Method: INVITE) <--- SIP read from UDP:192.168.1.147:43521 ---> ACK sip:350001@192.168.1.227 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.147:43521;branch=z9hG4bK.9vxdzeByK;rport Call-ID: dV~eC~AMO2 From: ;tag=cdYXYwnFg To: ;tag=as48abe0a9 Contact: ;+sip.instance="" Max-Forwards: 70 CSeq: 20 ACK <-------------> [Feb 20 15:23:35] DEBUG[32544]: chan_sip.c:9893 parse_request: Header 0 [ 37]: ACK sip:350001@192.168.1.227 SIP/2.0 [Feb 20 15:23:35] DEBUG[32544]: chan_sip.c:9893 parse_request: Header 1 [ 69]: Via: SIP/2.0/UDP 192.168.1.147:43521;branch=z9hG4bK.9vxdzeByK;rport [Feb 20 15:23:35] DEBUG[32544]: chan_sip.c:9893 parse_request: Header 2 [ 19]: Call-ID: dV~eC~AMO2 [Feb 20 15:23:35] DEBUG[32544]: chan_sip.c:9893 parse_request: Header 3 [ 46]: From: ;tag=cdYXYwnFg [Feb 20 15:23:35] DEBUG[32544]: chan_sip.c:9893 parse_request: Header 4 [ 46]: To: ;tag=as48abe0a9 [Feb 20 15:23:35] DEBUG[32544]: chan_sip.c:9893 parse_request: Header 5 [106]: Contact: ;+sip.instance="" [Feb 20 15:23:35] DEBUG[32544]: chan_sip.c:9893 parse_request: Header 6 [ 16]: Max-Forwards: 70 [Feb 20 15:23:35] DEBUG[32544]: chan_sip.c:9893 parse_request: Header 7 [ 12]: CSeq: 20 ACK --- (8 headers 0 lines) --- [Feb 20 15:23:35] DEBUG[32544]: chan_sip.c:9419 __find_call: = Looking for Call ID: dV~eC~AMO2 (Checking From) --From tag cdYXYwnFg --To-tag as48abe0a9 [Feb 20 15:23:35] DEBUG[32544][C-00000001]: chan_sip.c:28652 handle_incoming: **** Received ACK (6) - Command in SIP ACK [Feb 20 15:23:35] DEBUG[32544][C-00000001]: chan_sip.c:4526 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #3 [Feb 20 15:23:35] DEBUG[32544][C-00000001]: chan_sip.c:4537 __sip_ack: Stopping retransmission on 'dV~eC~AMO2' of Response 20: Match Found <--- SIP read from UDP:192.168.1.147:43521 ---> INVITE sip:350001@192.168.1.227 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.147:43521;branch=z9hG4bK.InpsD7vqg;rport From: ;tag=cdYXYwnFg To: sip:350001@192.168.1.227 CSeq: 21 INVITE Call-ID: dV~eC~AMO2 Max-Forwards: 70 Supported: outbound Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO, UPDATE Content-Type: application/sdp Content-Length: 335 Contact: ;+sip.instance="" User-Agent: Linphone/3.9.1 (belle-sip/1.4.2) Authorization: Digest realm="asterisk", nonce="2bc3b02a", algorithm=MD5, username="50001", uri="sip:350001@192.168.1.227", response="62bca94952d214be245885afcee9edab" v=0 o=50001 4045 4054 IN IP4 192.168.41.102 s=Talk c=IN IP4 192.168.41.102 b=AS:200 t=0 0 a=rtcp-xr:rcvr-rtt=all:10000 stat-summary=loss,dup,jitt,TTL voip-metrics m=audio 7080 RTP/AVP 96 101 a=rtpmap:96 opus/48000/2 a=fmtp:96 useinbandfec=1 a=rtpmap:101 telephone-event/48000 m=video 9078 RTP/AVP 96 a=rtpmap:96 VP8/90000 <-------------> [Feb 20 15:23:35] DEBUG[32544]: chan_sip.c:9893 parse_request: Header 0 [ 40]: INVITE sip:350001@192.168.1.227 SIP/2.0 [Feb 20 15:23:35] DEBUG[32544]: chan_sip.c:9893 parse_request: Header 1 [ 69]: Via: SIP/2.0/UDP 192.168.1.147:43521;branch=z9hG4bK.InpsD7vqg;rport [Feb 20 15:23:35] DEBUG[32544]: chan_sip.c:9893 parse_request: Header 2 [ 46]: From: ;tag=cdYXYwnFg [Feb 20 15:23:35] DEBUG[32544]: chan_sip.c:9893 parse_request: Header 3 [ 29]: To: sip:350001@192.168.1.227 [Feb 20 15:23:35] DEBUG[32544]: chan_sip.c:9893 parse_request: Header 4 [ 15]: CSeq: 21 INVITE [Feb 20 15:23:35] DEBUG[32544]: chan_sip.c:9893 parse_request: Header 5 [ 19]: Call-ID: dV~eC~AMO2 [Feb 20 15:23:35] DEBUG[32544]: chan_sip.c:9893 parse_request: Header 6 [ 16]: Max-Forwards: 70 [Feb 20 15:23:35] DEBUG[32544]: chan_sip.c:9893 parse_request: Header 7 [ 19]: Supported: outbound [Feb 20 15:23:35] DEBUG[32544]: chan_sip.c:9893 parse_request: Header 8 [ 89]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO, UPDATE [Feb 20 15:23:35] DEBUG[32544]: chan_sip.c:9893 parse_request: Header 9 [ 29]: Content-Type: application/sdp [Feb 20 15:23:35] DEBUG[32544]: chan_sip.c:9893 parse_request: Header 10 [ 19]: Content-Length: 335 [Feb 20 15:23:35] DEBUG[32544]: chan_sip.c:9893 parse_request: Header 11 [106]: Contact: ;+sip.instance="" [Feb 20 15:23:35] DEBUG[32544]: chan_sip.c:9893 parse_request: Header 12 [ 44]: User-Agent: Linphone/3.9.1 (belle-sip/1.4.2) [Feb 20 15:23:35] DEBUG[32550]: res_pjsip_registrar_expire.c:78 check_expiration_thread: Woke up at 1487575415 Interval: 30 [Feb 20 15:23:35] DEBUG[32550]: res_pjsip_registrar_expire.c:85 check_expiration_thread: Expiring 0 contacts [Feb 20 15:23:35] DEBUG[32544]: chan_sip.c:9893 parse_request: Header 13 [167]: Authorization: Digest realm="asterisk", nonce="2bc3b02a", algorithm=MD5, username="50001", uri="sip:350001@192.168.1.227", response="62bca94952d214be245885afcee9edab" [Feb 20 15:23:35] DEBUG[32544]: chan_sip.c:9893 parse_request: Header 14 [ 0]: [Feb 20 15:23:35] DEBUG[32544]: chan_sip.c:9893 parse_request: Body 0 [ 3]: v=0 [Feb 20 15:23:35] DEBUG[32544]: chan_sip.c:9893 parse_request: Body 1 [ 39]: o=50001 4045 4054 IN IP4 192.168.41.102 [Feb 20 15:23:35] DEBUG[32544]: chan_sip.c:9893 parse_request: Body 2 [ 6]: s=Talk [Feb 20 15:23:35] DEBUG[32544]: chan_sip.c:9893 parse_request: Body 3 [ 23]: c=IN IP4 192.168.41.102 [Feb 20 15:23:35] DEBUG[32544]: chan_sip.c:9893 parse_request: Body 4 [ 8]: b=AS:200 [Feb 20 15:23:35] DEBUG[32544]: chan_sip.c:9893 parse_request: Body 5 [ 5]: t=0 0 [Feb 20 15:23:35] DEBUG[32544]: chan_sip.c:9893 parse_request: Body 6 [ 72]: a=rtcp-xr:rcvr-rtt=all:10000 stat-summary=loss,dup,jitt,TTL voip-metrics [Feb 20 15:23:35] DEBUG[32544]: chan_sip.c:9893 parse_request: Body 7 [ 27]: m=audio 7080 RTP/AVP 96 101 [Feb 20 15:23:35] DEBUG[32544]: chan_sip.c:9893 parse_request: Body 8 [ 24]: a=rtpmap:96 opus/48000/2 [Feb 20 15:23:35] DEBUG[32544]: chan_sip.c:9893 parse_request: Body 9 [ 24]: a=fmtp:96 useinbandfec=1 [Feb 20 15:23:35] DEBUG[32544]: chan_sip.c:9893 parse_request: Body 10 [ 34]: a=rtpmap:101 telephone-event/48000 [Feb 20 15:23:35] DEBUG[32544]: chan_sip.c:9893 parse_request: Body 11 [ 23]: m=video 9078 RTP/AVP 96 [Feb 20 15:23:35] DEBUG[32544]: chan_sip.c:9930 parse_request: Body 12 [ 21]: a=rtpmap:96 VP8/90000 --- (14 headers 13 lines) --- [Feb 20 15:23:35] DEBUG[32544]: chan_sip.c:9419 __find_call: = Looking for Call ID: dV~eC~AMO2 (Checking From) --From tag cdYXYwnFg --To-tag [Feb 20 15:23:35] DEBUG[32544]: netsock2.c:172 ast_sockaddr_split_hostport: Splitting '192.168.1.227' into... [Feb 20 15:23:35] DEBUG[32544]: netsock2.c:226 ast_sockaddr_split_hostport: ...host '192.168.1.227' and port ''. [Feb 20 15:23:35] DEBUG[32544]: netsock2.c:172 ast_sockaddr_split_hostport: Splitting '192.168.1.227' into... [Feb 20 15:23:35] DEBUG[32544]: netsock2.c:226 ast_sockaddr_split_hostport: ...host '192.168.1.227' and port ''. [Feb 20 15:23:35] DEBUG[32544][C-00000001]: chan_sip.c:28652 handle_incoming: **** Received INVITE (5) - Command in SIP INVITE [Feb 20 15:23:35] DEBUG[32544][C-00000001]: netsock2.c:172 ast_sockaddr_split_hostport: Splitting '192.168.1.147:43521' into... [Feb 20 15:23:35] DEBUG[32544][C-00000001]: netsock2.c:226 ast_sockaddr_split_hostport: ...host '192.168.1.147' and port '43521'. Sending to 192.168.1.147:43521 (no NAT) [Feb 20 15:23:35] DEBUG[32544][C-00000001]: chan_sip.c:26104 handle_request_invite: Initializing initreq for method INVITE - callid dV~eC~AMO2 Using INVITE request as basis request - dV~eC~AMO2 [Feb 20 15:23:35] DEBUG[32544][C-00000001]: netsock2.c:172 ast_sockaddr_split_hostport: Splitting '192.168.1.227' into... [Feb 20 15:23:35] DEBUG[32544][C-00000001]: netsock2.c:226 ast_sockaddr_split_hostport: ...host '192.168.1.227' and port ''. Found peer '50001' for '50001' from 192.168.1.147:43521 [Feb 20 15:23:35] DEBUG[32544][C-00000001]: rtp_engine.c:454 ast_rtp_instance_new: Using engine 'asterisk' for RTP instance '0xa02e108' [Feb 20 15:23:35] DEBUG[32544][C-00000001]: res_rtp_asterisk.c:2681 ast_rtp_new: Allocated port 14090 for RTP instance '0xa02e108' [Feb 20 15:23:35] DEBUG[32544][C-00000001]: res_rtp_asterisk.c:2708 ast_rtp_new: Creating ICE session 0.0.0.0:14090 (14090) for RTP instance '0xa02e108' [Feb 20 15:23:35] DEBUG[32544][C-00000001]: netsock2.c:172 ast_sockaddr_split_hostport: Splitting '192.168.41.78' into... [Feb 20 15:23:35] DEBUG[32544][C-00000001]: netsock2.c:226 ast_sockaddr_split_hostport: ...host '192.168.41.78' and port ''. [Feb 20 15:23:35] DEBUG[32544][C-00000001]: netsock2.c:172 ast_sockaddr_split_hostport: Splitting '192.168.41.78' into... [Feb 20 15:23:35] DEBUG[32544][C-00000001]: netsock2.c:226 ast_sockaddr_split_hostport: ...host '192.168.41.78' and port ''. [Feb 20 15:23:35] DEBUG[32544][C-00000001]: rtp_engine.c:463 ast_rtp_instance_new: RTP instance '0xa02e108' is setup and ready to go [Feb 20 15:23:35] DEBUG[32544][C-00000001]: rtp_engine.c:454 ast_rtp_instance_new: Using engine 'asterisk' for RTP instance '0x9ee14c8' [Feb 20 15:23:35] DEBUG[32544][C-00000001]: res_rtp_asterisk.c:2681 ast_rtp_new: Allocated port 16634 for RTP instance '0x9ee14c8' [Feb 20 15:23:35] DEBUG[32544][C-00000001]: res_rtp_asterisk.c:2708 ast_rtp_new: Creating ICE session 0.0.0.0:16634 (16634) for RTP instance '0x9ee14c8' [Feb 20 15:23:35] DEBUG[32544][C-00000001]: netsock2.c:172 ast_sockaddr_split_hostport: Splitting '192.168.41.78' into... [Feb 20 15:23:35] DEBUG[32544][C-00000001]: netsock2.c:226 ast_sockaddr_split_hostport: ...host '192.168.41.78' and port ''. [Feb 20 15:23:35] DEBUG[32544][C-00000001]: netsock2.c:172 ast_sockaddr_split_hostport: Splitting '192.168.41.78' into... [Feb 20 15:23:35] DEBUG[32544][C-00000001]: netsock2.c:226 ast_sockaddr_split_hostport: ...host '192.168.41.78' and port ''. [Feb 20 15:23:35] DEBUG[32544][C-00000001]: rtp_engine.c:463 ast_rtp_instance_new: RTP instance '0x9ee14c8' is setup and ready to go [Feb 20 15:23:35] DEBUG[32544][C-00000001]: netsock2.c:172 ast_sockaddr_split_hostport: Splitting 'iZ94rhgsz5zZ' into... [Feb 20 15:23:35] DEBUG[32544][C-00000001]: netsock2.c:226 ast_sockaddr_split_hostport: ...host 'iZ94rhgsz5zZ' and port ''. [Feb 20 15:23:35] DEBUG[32544][C-00000001]: res_rtp_asterisk.c:4967 ast_rtp_prop_set: Setup RTCP on RTP instance '0x9ee14c8' == Using SIP VIDEO CoS mark 6 [Feb 20 15:23:35] DEBUG[32544][C-00000001]: netsock2.c:172 ast_sockaddr_split_hostport: Splitting 'iZ94rhgsz5zZ' into... [Feb 20 15:23:35] DEBUG[32544][C-00000001]: netsock2.c:226 ast_sockaddr_split_hostport: ...host 'iZ94rhgsz5zZ' and port ''. [Feb 20 15:23:35] DEBUG[32544][C-00000001]: res_rtp_asterisk.c:4967 ast_rtp_prop_set: Setup RTCP on RTP instance '0xa02e108' == Using SIP RTP CoS mark 5 [Feb 20 15:23:35] DEBUG[32544][C-00000001]: chan_sip.c:5798 do_setnat: Setting NAT on RTP to On [Feb 20 15:23:35] DEBUG[32544][C-00000001]: chan_sip.c:5802 do_setnat: Setting NAT on VRTP to On [Feb 20 15:23:35] DEBUG[32544][C-00000001]: chan_sip.c:10271 process_sdp: Processing session-level SDP v=0... UNSUPPORTED OR FAILED. [Feb 20 15:23:35] DEBUG[32544][C-00000001]: chan_sip.c:10271 process_sdp: Processing session-level SDP o=50001 4045 4054 IN IP4 192.168.41.102... OK. [Feb 20 15:23:35] DEBUG[32544][C-00000001]: chan_sip.c:10271 process_sdp: Processing session-level SDP s=Talk... UNSUPPORTED OR FAILED. [Feb 20 15:23:35] DEBUG[32544][C-00000001]: netsock2.c:172 ast_sockaddr_split_hostport: Splitting '192.168.41.102' into... [Feb 20 15:23:35] DEBUG[32544][C-00000001]: netsock2.c:226 ast_sockaddr_split_hostport: ...host '192.168.41.102' and port ''. [Feb 20 15:23:35] DEBUG[32544][C-00000001]: chan_sip.c:10271 process_sdp: Processing session-level SDP c=IN IP4 192.168.41.102... OK. [Feb 20 15:23:35] DEBUG[32544][C-00000001]: chan_sip.c:10271 process_sdp: Processing session-level SDP b=AS:200... UNSUPPORTED OR FAILED. [Feb 20 15:23:35] DEBUG[32544][C-00000001]: chan_sip.c:10271 process_sdp: Processing session-level SDP t=0 0... UNSUPPORTED OR FAILED. [Feb 20 15:23:35] DEBUG[32544][C-00000001]: chan_sip.c:10271 process_sdp: Processing session-level SDP a=rtcp-xr:rcvr-rtt=all:10000 stat-summary=loss,dup,jitt,TTL voip-metrics... UNSUPPORTED OR FAILED. Found RTP audio format 96 [Feb 20 15:23:35] DEBUG[32544][C-00000001]: rtp_engine.c:684 ast_rtp_codecs_payloads_set_m_type: Don't have a default tx payload type 96 format for m type on 0x47403080 Found RTP audio format 101 [Feb 20 15:23:35] DEBUG[32544][C-00000001]: rtp_engine.c:689 ast_rtp_codecs_payloads_set_m_type: Setting tx payload type 101 based on m type on 0x47403080 Found audio description format opus for ID 96 [Feb 20 15:23:35] DEBUG[32544][C-00000001]: chan_sip.c:10726 process_sdp: Processing media-level (audio) SDP a=rtpmap:96 opus/48000/2... OK. [Feb 20 15:23:35] DEBUG[32544][C-00000001]: chan_sip.c:10726 process_sdp: Processing media-level (audio) SDP a=fmtp:96 useinbandfec=1... OK. [Feb 20 15:23:35] DEBUG[32544][C-00000001]: rtp_engine.c:791 ast_rtp_codecs_payloads_unset: Unsetting payload 101 on 0x47403080 Found unknown media description format telephone-event for ID 101 [Feb 20 15:23:35] DEBUG[32544][C-00000001]: chan_sip.c:10726 process_sdp: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/48000... UNSUPPORTED OR FAILED. Found RTP video format 96 [Feb 20 15:23:35] DEBUG[32544][C-00000001]: rtp_engine.c:684 ast_rtp_codecs_payloads_set_m_type: Don't have a default tx payload type 96 format for m type on 0x47403010 Found video description format VP8 for ID 96 [Feb 20 15:23:35] DEBUG[32544][C-00000001]: chan_sip.c:10726 process_sdp: Processing media-level (video) SDP a=rtpmap:96 VP8/90000... OK. Capabilities: us - (opus|vp8), peer - audio=(opus)/video=(vp8)/text=(nothing), combined - (opus|vp8) Non-codec capabilities (dtmf): us - 0x0 (nothing), peer - 0x0 (nothing), combined - 0x0 (nothing) [Feb 20 15:23:35] DEBUG[32544][C-00000001]: acl.c:957 ast_ouraddrfor: For destination '192.168.41.102', our source address is '192.168.41.78'. [Feb 20 15:23:35] DEBUG[32544][C-00000001]: res_rtp_asterisk.c:5036 ast_rtp_remote_address_set: Setting RTCP address on RTP instance '0xa02e108' Peer audio RTP is at port 192.168.41.102:7080 [Feb 20 15:23:35] DEBUG[32544][C-00000001]: rtp_engine.c:658 ast_rtp_codecs_payloads_copy: Copying payload 96 (0xce980a0) from 0x47403080 to 0xa02e2d0 [Feb 20 15:23:35] DEBUG[32544][C-00000001]: res_rtp_asterisk.c:4916 ast_rtp_prop_set: Ignoring duplicate RTCP property on RTP instance '0xa02e108' [Feb 20 15:23:35] DEBUG[32544][C-00000001]: res_rtp_asterisk.c:853 ast_rtp_ice_set_role: Set role to CONTROLLED (0x9ee14c8) [Feb 20 15:23:35] DEBUG[32544][C-00000001]: res_rtp_asterisk.c:856 ast_rtp_ice_set_role: Set role failed; no ice instance (0x9ee14c8) [Feb 20 15:23:35] DEBUG[32544][C-00000001]: acl.c:957 ast_ouraddrfor: For destination '192.168.41.102', our source address is '192.168.41.78'. [Feb 20 15:23:35] DEBUG[32544][C-00000001]: res_rtp_asterisk.c:5036 ast_rtp_remote_address_set: Setting RTCP address on RTP instance '0x9ee14c8' Peer video RTP is at port 192.168.41.102:9078 [Feb 20 15:23:35] DEBUG[32544][C-00000001]: rtp_engine.c:658 ast_rtp_codecs_payloads_copy: Copying payload 96 (0xd91bce0) from 0x47403010 to 0x9ee1690 [Feb 20 15:23:35] DEBUG[32544][C-00000001]: chan_sip.c:11022 process_sdp: We're settling with these formats: (opus|vp8) [Feb 20 15:23:35] DEBUG[32544][C-00000001]: chan_sip.c:26236 handle_request_invite: Checking SIP call limits for device 50001 [Feb 20 15:23:35] DEBUG[32544][C-00000001]: chan_sip.c:6767 update_call_counter: Updating call counter for incoming call [Feb 20 15:23:35] DEBUG[32544][C-00000001]: netsock2.c:172 ast_sockaddr_split_hostport: Splitting '192.168.1.227' into... [Feb 20 15:23:35] DEBUG[32544][C-00000001]: netsock2.c:226 ast_sockaddr_split_hostport: ...host '192.168.1.227' and port ''. [Feb 20 15:23:35] DEBUG[32544][C-00000001]: netsock2.c:172 ast_sockaddr_split_hostport: Splitting '192.168.1.227' into... [Feb 20 15:23:35] DEBUG[32544][C-00000001]: netsock2.c:226 ast_sockaddr_split_hostport: ...host '192.168.1.227' and port ''. Looking for 350001 in default (domain 192.168.1.227) [Feb 20 15:23:35] DEBUG[32544][C-00000001]: pbx_lua.c:1480 lua_find_extension: Looking up 350001@default:1 [Feb 20 15:23:35] DEBUG[32544][C-00000001]: pbx_lua.c:1480 lua_find_extension: Looking up 350001@demo:1 [Feb 20 15:23:35] DEBUG[32544][C-00000001]: chan_sip.c:8153 sip_new: *** Our native formats are (vp8|opus) [Feb 20 15:23:35] DEBUG[32544][C-00000001]: chan_sip.c:8154 sip_new: *** Joint capabilities are (opus|vp8) [Feb 20 15:23:35] DEBUG[32544][C-00000001]: chan_sip.c:8155 sip_new: *** Our capabilities are (opus|vp8) [Feb 20 15:23:35] DEBUG[32544][C-00000001]: chan_sip.c:8156 sip_new: *** Prefcaps capabilities are (nothing) [Feb 20 15:23:35] DEBUG[32544][C-00000001]: chan_sip.c:8157 sip_new: *** AST_CODEC_CHOOSE formats are opus [Feb 20 15:23:35] DEBUG[32544][C-00000001]: chan_sip.c:8169 sip_new: *** ************* SIP_PAGE2_VIDEOSUPPORT_ALWAYS *********** [Feb 20 15:23:35] DEBUG[32544][C-00000001]: chan_sip.c:8197 sip_new: This channel can handle video! HOLLYWOOD next! [Feb 20 15:23:35] DEBUG[32544][C-00000001]: pbx_lua.c:1480 lua_find_extension: Looking up 350001@default:1 [Feb 20 15:23:35] DEBUG[32544][C-00000001]: pbx_lua.c:1480 lua_find_extension: Looking up 350001@demo:1 sip_route_dump: route/path hop: [Feb 20 15:23:35] DEBUG[32544][C-00000001]: chan_sip.c:26440 handle_request_invite: SIP/50001-00000001: New call is still down.... Trying... <--- Transmitting (no NAT) to 192.168.1.147:43521 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.1.147:43521;branch=z9hG4bK.InpsD7vqg;received=192.168.1.147;rport=43521 From: ;tag=cdYXYwnFg To: sip:350001@192.168.1.227 Call-ID: dV~eC~AMO2 CSeq: 21 INVITE Server: Asterisk PBX 13.14.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Contact: Content-Length: 0 <------------> [Feb 20 15:23:35] DEBUG[32544][C-00000001]: chan_sip.c:3753 __sip_xmit: Trying to put 'SIP/2.0 100' onto UDP socket destined for 192.168.1.147:43521 [Feb 20 15:23:35] DEBUG[32495]: devicestate.c:369 _ast_device_state: No provider found, checking channel drivers for SIP - 50001 [Feb 20 15:23:35] DEBUG[32495]: chan_sip.c:30178 sip_devicestate: Checking device state for peer 50001 [Feb 20 15:23:35] DEBUG[32495]: devicestate.c:474 do_state_change: Changing state for SIP/50001 - state 1 (Not in use) <--- SIP read from UDP:192.168.1.147:43521 ---> INVITE sip:350001@192.168.1.227 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.147:43521;branch=z9hG4bK.InpsD7vqg;rport From: ;tag=cdYXYwnFg To: sip:350001@192.168.1.227 CSeq: 21 INVITE Call-ID: dV~eC~AMO2 Max-Forwards: 70 Supported: outbound Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO, UPDATE Content-Type: application/sdp Content-Length: 335 Contact: ;+sip.instance="" User-Agent: Linphone/3.9.1 (belle-sip/1.4.2) Authorization: Digest realm="asterisk", nonce="2bc3b02a", algorithm=MD5, username="50001", uri="sip:350001@192.168.1.227", response="62bca94952d214be245885afcee9edab" v=0 o=50001 4045 4054 IN IP4 192.168.41.102 s=Talk c=IN IP4 192.168.41.102 b=AS:200 t=0 0 a=rtcp-xr:rcvr-rtt=all:10000 stat-summary=loss,dup,jitt,TTL voip-metrics m=audio 7080 RTP/AVP 96 101 a=rtpmap:96 opus/48000/2 a=fmtp:96 useinbandfec=1 a=rtpmap:101 telephone-event/48000 m=video 9078 RTP/AVP 96 a=rtpmap:96 VP8/90000 <-------------> [Feb 20 15:23:35] DEBUG[32544]: chan_sip.c:9893 parse_request: Header 0 [ 40]: INVITE sip:350001@192.168.1.227 SIP/2.0 [Feb 20 15:23:35] DEBUG[32544]: chan_sip.c:9893 parse_request: Header 1 [ 69]: Via: SIP/2.0/UDP 192.168.1.147:43521;branch=z9hG4bK.InpsD7vqg;rport [Feb 20 15:23:35] DEBUG[32544]: chan_sip.c:9893 parse_request: Header 2 [ 46]: From: ;tag=cdYXYwnFg [Feb 20 15:23:35] DEBUG[32544]: chan_sip.c:9893 parse_request: Header 3 [ 29]: To: sip:350001@192.168.1.227 [Feb 20 15:23:35] DEBUG[32580][C-00000001]: pbx_lua.c:1480 lua_find_extension: Looking up 350001@default:1 [Feb 20 15:23:35] DEBUG[32544]: chan_sip.c:9893 parse_request: Header 4 [ 15]: CSeq: 21 INVITE [Feb 20 15:23:35] DEBUG[32544]: chan_sip.c:9893 parse_request: Header 5 [ 19]: Call-ID: dV~eC~AMO2 [Feb 20 15:23:35] DEBUG[32544]: chan_sip.c:9893 parse_request: Header 6 [ 16]: Max-Forwards: 70 [Feb 20 15:23:35] DEBUG[32580][C-00000001]: pbx_lua.c:1480 lua_find_extension: Looking up 350001@demo:1 [Feb 20 15:23:35] DEBUG[32544]: chan_sip.c:9893 parse_request: Header 7 [ 19]: Supported: outbound [Feb 20 15:23:35] DEBUG[32544]: chan_sip.c:9893 parse_request: Header 8 [ 89]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO, UPDATE [Feb 20 15:23:35] DEBUG[32544]: chan_sip.c:9893 parse_request: Header 9 [ 29]: Content-Type: application/sdp [Feb 20 15:23:35] DEBUG[32544]: chan_sip.c:9893 parse_request: Header 10 [ 19]: Content-Length: 335 [Feb 20 15:23:35] DEBUG[32544]: chan_sip.c:9893 parse_request: Header 11 [106]: Contact: ;+sip.instance="" [Feb 20 15:23:35] DEBUG[32544]: chan_sip.c:9893 parse_request: Header 12 [ 44]: User-Agent: Linphone/3.9.1 (belle-sip/1.4.2) [Feb 20 15:23:35] DEBUG[32544]: chan_sip.c:9893 parse_request: Header 13 [167]: Authorization: Digest realm="asterisk", nonce="2bc3b02a", algorithm=MD5, username="50001", uri="sip:350001@192.168.1.227", response="62bca94952d214be245885afcee9edab" [Feb 20 15:23:35] DEBUG[32544]: chan_sip.c:9893 parse_request: Header 14 [ 0]: [Feb 20 15:23:36] DEBUG[32580][C-00000001]: pbx.c:2875 pbx_extension_helper: Launching 'Playback' -- Executing [350001@default:1] Playback("SIP/50001-00000001", "beep") in new stack [Feb 20 15:23:35] DEBUG[32544]: chan_sip.c:9893 parse_request: Body 0 [ 3]: v=0 [Feb 20 15:23:36] DEBUG[32544]: chan_sip.c:9893 parse_request: Body 1 [ 39]: o=50001 4045 4054 IN IP4 192.168.41.102 [Feb 20 15:23:36] DEBUG[32544]: chan_sip.c:9893 parse_request: Body 2 [ 6]: s=Talk [Feb 20 15:23:36] DEBUG[32544]: chan_sip.c:9893 parse_request: Body 3 [ 23]: c=IN IP4 192.168.41.102 [Feb 20 15:23:36] DEBUG[32544]: chan_sip.c:9893 parse_request: Body 4 [ 8]: b=AS:200 [Feb 20 15:23:36] DEBUG[32544]: chan_sip.c:9893 parse_request: Body 5 [ 5]: t=0 0 [Feb 20 15:23:36] DEBUG[32544]: chan_sip.c:9893 parse_request: Body 6 [ 72]: a=rtcp-xr:rcvr-rtt=all:10000 stat-summary=loss,dup,jitt,TTL voip-metrics [Feb 20 15:23:36] DEBUG[32544]: chan_sip.c:9893 parse_request: Body 7 [ 27]: m=audio 7080 RTP/AVP 96 101 [Feb 20 15:23:36] DEBUG[32544]: chan_sip.c:9893 parse_request: Body 8 [ 24]: a=rtpmap:96 opus/48000/2 [Feb 20 15:23:36] DEBUG[32544]: chan_sip.c:9893 parse_request: Body 9 [ 24]: a=fmtp:96 useinbandfec=1 [Feb 20 15:23:36] DEBUG[32544]: chan_sip.c:9893 parse_request: Body 10 [ 34]: a=rtpmap:101 telephone-event/48000 [Feb 20 15:23:36] DEBUG[32544]: chan_sip.c:9893 parse_request: Body 11 [ 23]: m=video 9078 RTP/AVP 96 [Feb 20 15:23:36] DEBUG[32544]: chan_sip.c:9930 parse_request: Body 12 [ 21]: a=rtpmap:96 VP8/90000 --- (14 headers 13 lines) --- [Feb 20 15:23:36] DEBUG[32544]: chan_sip.c:9419 __find_call: = Looking for Call ID: dV~eC~AMO2 (Checking From) --From tag cdYXYwnFg --To-tag [Feb 20 15:23:36] DEBUG[32544]: netsock2.c:172 ast_sockaddr_split_hostport: Splitting '192.168.1.227' into... [Feb 20 15:23:36] DEBUG[32544]: netsock2.c:226 ast_sockaddr_split_hostport: ...host '192.168.1.227' and port ''. [Feb 20 15:23:36] DEBUG[32544]: netsock2.c:172 ast_sockaddr_split_hostport: Splitting '192.168.1.227' into... [Feb 20 15:23:36] DEBUG[32544]: netsock2.c:226 ast_sockaddr_split_hostport: ...host '192.168.1.227' and port ''. [Feb 20 15:23:36] DEBUG[32580][C-00000001]: chan_sip.c:7412 sip_answer: SIP answering channel: SIP/50001-00000001 [Feb 20 15:23:36] DEBUG[32495]: devicestate.c:369 _ast_device_state: No provider found, checking channel drivers for SIP - 50001 [Feb 20 15:23:36] DEBUG[32495]: chan_sip.c:30178 sip_devicestate: Checking device state for peer 50001 [Feb 20 15:23:36] DEBUG[32495]: devicestate.c:474 do_state_change: Changing state for SIP/50001 - state 1 (Not in use) [Feb 20 15:23:36] DEBUG[32580][C-00000001]: res_rtp_asterisk.c:3052 ast_rtp_update_source: Setting the marker bit due to a source update [Feb 20 15:23:36] DEBUG[32580][C-00000001]: chan_sip.c:13392 add_sdp: This call needs video offers! [Feb 20 15:23:36] DEBUG[32580][C-00000001]: chan_sip.c:13416 add_sdp: ** Our capability: (opus|vp8) Video flag: False Text flag: True [Feb 20 15:23:36] DEBUG[32580][C-00000001]: chan_sip.c:13417 add_sdp: ** Our prefcodec: (nothing) Audio is at 14090 Video is at 192.168.1.227:16634 Adding codec opus to SDP Adding video codec vp8 to SDP [Feb 20 15:23:36] DEBUG[32580][C-00000001]: chan_sip.c:13587 add_sdp: -- Done with adding codecs to SDP [Feb 20 15:23:36] DEBUG[32580][C-00000001]: chan_sip.c:13612 add_sdp: Setting framing on incoming call: 0 [Feb 20 15:23:36] DEBUG[32580][C-00000001]: chan_sip.c:13800 add_sdp: Done building SDP. Settling with this capability: (opus|vp8) <--- Reliably Transmitting (no NAT) to 192.168.1.147:43521 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.147:43521;branch=z9hG4bK.InpsD7vqg;received=192.168.1.147;rport=43521 From: ;tag=cdYXYwnFg To: sip:350001@192.168.1.227;tag=as413e790c Call-ID: dV~eC~AMO2 CSeq: 21 INVITE Server: Asterisk PBX 13.14.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Contact: Content-Type: application/sdp Content-Length: 284 v=0 o=root 1515163174 1515163174 IN IP4 192.168.1.227 s=Asterisk PBX 13.14.0 c=IN IP4 192.168.1.227 b=CT:384 t=0 0 m=audio 14090 RTP/AVP 96 a=rtpmap:96 opus/48000/2 a=maxptime:20 a=sendrecv m=video 16634 RTP/AVP 96 a=rtpmap:96 VP8/90000 a=rtcp-fb:* ccm fir a=sendrecv <------------> [Feb 20 15:23:36] DEBUG[32580][C-00000001]: chan_sip.c:4266 __sip_reliable_xmit: *** SIP TIMER: Initializing retransmit timer on packet: Id #14 [Feb 20 15:23:36] DEBUG[32580][C-00000001]: chan_sip.c:3753 __sip_xmit: Trying to put 'SIP/2.0 200' onto UDP socket destined for 192.168.1.147:43521 [Feb 20 15:23:36] DEBUG[32544][C-00000001]: chan_sip.c:28652 handle_incoming: **** Received INVITE (5) - Command in SIP INVITE [Feb 20 15:23:36] DEBUG[32544][C-00000001]: chan_sip.c:28677 handle_incoming: Ignoring SIP message because of retransmit (INVITE Seqno 21, ours 21) Ignoring this INVITE request [Feb 20 15:23:36] DEBUG[32544][C-00000001]: chan_sip.c:26344 handle_request_invite: Got a SIP re-transmit of INVITE for call dV~eC~AMO2 [Feb 20 15:23:36] DEBUG[32544][C-00000001]: chan_sip.c:26506 handle_request_invite: SIP/50001-00000001: This call is UP.... <--- Transmitting (no NAT) to 192.168.1.147:43521 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.1.147:43521;branch=z9hG4bK.InpsD7vqg;received=192.168.1.147;rport=43521 From: ;tag=cdYXYwnFg To: sip:350001@192.168.1.227 Call-ID: dV~eC~AMO2 CSeq: 21 INVITE Server: Asterisk PBX 13.14.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Contact: Content-Length: 0 <------------> [Feb 20 15:23:36] DEBUG[32544][C-00000001]: chan_sip.c:3753 __sip_xmit: Trying to put 'SIP/2.0 100' onto UDP socket destined for 192.168.1.147:43521 [Feb 20 15:23:36] DEBUG[32544][C-00000001]: chan_sip.c:13392 add_sdp: This call needs video offers! [Feb 20 15:23:36] DEBUG[32544][C-00000001]: chan_sip.c:13416 add_sdp: ** Our capability: (opus|vp8) Video flag: False Text flag: True [Feb 20 15:23:36] DEBUG[32544][C-00000001]: chan_sip.c:13417 add_sdp: ** Our prefcodec: (nothing) Audio is at 14090 Video is at 192.168.1.227:16634 Adding codec opus to SDP Adding video codec vp8 to SDP [Feb 20 15:23:36] DEBUG[32544][C-00000001]: chan_sip.c:13587 add_sdp: -- Done with adding codecs to SDP [Feb 20 15:23:36] DEBUG[32544][C-00000001]: chan_sip.c:13612 add_sdp: Setting framing on incoming call: 0 [Feb 20 15:23:36] DEBUG[32544][C-00000001]: chan_sip.c:13800 add_sdp: Done building SDP. Settling with this capability: (opus|vp8) <--- Transmitting (no NAT) to 192.168.1.147:43521 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.147:43521;branch=z9hG4bK.InpsD7vqg;received=192.168.1.147;rport=43521 From: ;tag=cdYXYwnFg To: sip:350001@192.168.1.227;tag=as413e790c Call-ID: dV~eC~AMO2 CSeq: 21 INVITE Server: Asterisk PBX 13.14.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Contact: Content-Type: application/sdp Content-Length: 284 v=0 o=root 1515163174 1515163175 IN IP4 192.168.1.227 s=Asterisk PBX 13.14.0 c=IN IP4 192.168.1.227 b=CT:384 t=0 0 m=audio 14090 RTP/AVP 96 a=rtpmap:96 opus/48000/2 a=maxptime:20 a=sendrecv m=video 16634 RTP/AVP 96 a=rtpmap:96 VP8/90000 a=rtcp-fb:* ccm fir a=sendrecv <------------> [Feb 20 15:23:36] DEBUG[32544][C-00000001]: chan_sip.c:3753 __sip_xmit: Trying to put 'SIP/2.0 200' onto UDP socket destined for 192.168.1.147:43521 <--- SIP read from UDP:192.168.1.147:43521 ---> <-------------> [Feb 20 15:23:36] DEBUG[32544]: chan_sip.c:9893 parse_request: Header 0 [ 0]: <--- SIP read from UDP:192.168.1.147:43521 ---> ACK sip:350001@192.168.1.227:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.147:43521;rport;branch=z9hG4bK.ewQhLOacV From: ;tag=cdYXYwnFg To: ;tag=as413e790c CSeq: 21 ACK Call-ID: dV~eC~AMO2 Max-Forwards: 70 Authorization: Digest realm="asterisk", nonce="2bc3b02a", algorithm=MD5, username="50001", uri="sip:350001@192.168.1.227", response="62bca94952d214be245885afcee9edab" <-------------> [Feb 20 15:23:36] DEBUG[32544]: chan_sip.c:9893 parse_request: Header 0 [ 42]: ACK sip:350001@192.168.1.227:5060 SIP/2.0 [Feb 20 15:23:36] DEBUG[32544]: chan_sip.c:9893 parse_request: Header 1 [ 69]: Via: SIP/2.0/UDP 192.168.1.147:43521;rport;branch=z9hG4bK.ewQhLOacV [Feb 20 15:23:36] DEBUG[32544]: chan_sip.c:9893 parse_request: Header 2 [ 46]: From: ;tag=cdYXYwnFg [Feb 20 15:23:36] DEBUG[32544]: chan_sip.c:9893 parse_request: Header 3 [ 46]: To: ;tag=as413e790c [Feb 20 15:23:36] DEBUG[32544]: chan_sip.c:9893 parse_request: Header 4 [ 12]: CSeq: 21 ACK [Feb 20 15:23:36] DEBUG[32544]: chan_sip.c:9893 parse_request: Header 5 [ 19]: Call-ID: dV~eC~AMO2 [Feb 20 15:23:36] DEBUG[32544]: chan_sip.c:9893 parse_request: Header 6 [ 16]: Max-Forwards: 70 [Feb 20 15:23:36] DEBUG[32544]: chan_sip.c:9893 parse_request: Header 7 [167]: Authorization: Digest realm="asterisk", nonce="2bc3b02a", algorithm=MD5, username="50001", uri="sip:350001@192.168.1.227", response="62bca94952d214be245885afcee9edab" --- (8 headers 0 lines) --- [Feb 20 15:23:36] DEBUG[32544]: chan_sip.c:9419 __find_call: = Looking for Call ID: dV~eC~AMO2 (Checking From) --From tag cdYXYwnFg --To-tag as413e790c [Feb 20 15:23:36] DEBUG[32580][C-00000001]: res_rtp_asterisk.c:4564 ast_rtp_read: 0x934bce0 -- Probation learning mode pass with source address 192.168.1.147:14337 > 0x934bce0 -- Probation passed - setting RTP source address to 192.168.1.147:14337 [Feb 20 15:23:36] DEBUG[32580][C-00000001]: acl.c:957 ast_ouraddrfor: For destination '192.168.1.147', our source address is '192.168.41.78'. [Feb 20 15:23:36] DEBUG[32580][C-00000001]: res_rtp_asterisk.c:5036 ast_rtp_remote_address_set: Setting RTCP address on RTP instance '0xa02e108' [Feb 20 15:23:36] DEBUG[32544][C-00000001]: chan_sip.c:28652 handle_incoming: **** Received ACK (6) - Command in SIP ACK [Feb 20 15:23:36] DEBUG[32544][C-00000001]: chan_sip.c:4526 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #14 [Feb 20 15:23:36] DEBUG[32544][C-00000001]: chan_sip.c:4537 __sip_ack: Stopping retransmission on 'dV~eC~AMO2' of Response 21: Match Found [Feb 20 15:23:36] DEBUG[32580][C-00000001]: channel.c:5680 set_format: Channel SIP/50001-00000001 setting write format path: gsm -> opus [Feb 20 15:23:36] DEBUG[32580][C-00000001]: res_rtp_asterisk.c:3625 ast_rtp_write: Ooh, format changed from none to opus [Feb 20 15:23:36] DEBUG[32580][C-00000001]: res_rtp_asterisk.c:3393 ast_rtp_raw_write: Difference is 672, ms is 6 [Feb 20 15:23:36] DEBUG[32580][C-00000001]: channel.c:3469 ast_settimeout_full: Scheduling timer at (50 requested / 50 actual) timer ticks per second -- Playing 'beep.gsm' (language 'en') [Feb 20 15:23:36] DEBUG[32580][C-00000001]: res_rtp_asterisk.c:4564 ast_rtp_read: 0x934bce0 -- Probation learning mode pass with source address 192.168.1.147:14337 > 0x934bce0 -- Probation passed - setting RTP source address to 192.168.1.147:14337 <--- SIP read from UDP:192.168.1.147:43521 ---> ACK sip:350001@192.168.1.227:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.147:43521;branch=z9hG4bK.ewQhLOacV;rport From: ;tag=cdYXYwnFg To: ;tag=as413e790c CSeq: 21 ACK Call-ID: dV~eC~AMO2 Max-Forwards: 70 Authorization: Digest realm="asterisk", nonce="2bc3b02a", algorithm=MD5, username="50001", uri="sip:350001@192.168.1.227", response="62bca94952d214be245885afcee9edab" <-------------> [Feb 20 15:23:36] DEBUG[32544]: chan_sip.c:9893 parse_request: Header 0 [ 42]: ACK sip:350001@192.168.1.227:5060 SIP/2.0 [Feb 20 15:23:36] DEBUG[32544]: chan_sip.c:9893 parse_request: Header 1 [ 69]: Via: SIP/2.0/UDP 192.168.1.147:43521;branch=z9hG4bK.ewQhLOacV;rport [Feb 20 15:23:36] DEBUG[32544]: chan_sip.c:9893 parse_request: Header 2 [ 46]: From: ;tag=cdYXYwnFg [Feb 20 15:23:36] DEBUG[32544]: chan_sip.c:9893 parse_request: Header 3 [ 46]: To: ;tag=as413e790c [Feb 20 15:23:36] DEBUG[32544]: chan_sip.c:9893 parse_request: Header 4 [ 12]: CSeq: 21 ACK [Feb 20 15:23:36] DEBUG[32544]: chan_sip.c:9893 parse_request: Header 5 [ 19]: Call-ID: dV~eC~AMO2 [Feb 20 15:23:36] DEBUG[32544]: chan_sip.c:9893 parse_request: Header 6 [ 16]: Max-Forwards: 70 [Feb 20 15:23:36] DEBUG[32544]: chan_sip.c:9893 parse_request: Header 7 [167]: Authorization: Digest realm="asterisk", nonce="2bc3b02a", algorithm=MD5, username="50001", uri="sip:350001@192.168.1.227", response="62bca94952d214be245885afcee9edab" --- (8 headers 0 lines) --- [Feb 20 15:23:36] DEBUG[32544]: chan_sip.c:9419 __find_call: = Looking for Call ID: dV~eC~AMO2 (Checking From) --From tag cdYXYwnFg --To-tag as413e790c [Feb 20 15:23:36] DEBUG[32580][C-00000001]: res_rtp_asterisk.c:3393 ast_rtp_raw_write: Difference is 3024, ms is 83 [Feb 20 15:23:36] DEBUG[32513]: threadpool.c:1137 worker_idle: Worker thread idle timeout reached. Dying. [Feb 20 15:23:36] DEBUG[32483]: threadpool.c:996 worker_thread_destroy: Destroying worker thread 10 [Feb 20 15:23:36] DEBUG[32580][C-00000001]: res_timing_timerfd.c:167 timerfd_timer_ack: Expected to acknowledge 1 ticks but got 3 instead [Feb 20 15:23:36] DEBUG[32544][C-00000001]: chan_sip.c:28652 handle_incoming: **** Received ACK (6) - Command in SIP ACK [Feb 20 15:23:36] DEBUG[32580][C-00000001]: res_rtp_asterisk.c:3393 ast_rtp_raw_write: Difference is 1968, ms is 61 [Feb 20 15:23:36] DEBUG[32580][C-00000001]: res_timing_timerfd.c:167 timerfd_timer_ack: Expected to acknowledge 1 ticks but got 3 instead [Feb 20 15:23:36] DEBUG[32580][C-00000001]: res_rtp_asterisk.c:3393 ast_rtp_raw_write: Difference is 1200, ms is 45 [Feb 20 15:23:36] DEBUG[32580][C-00000001]: res_timing_timerfd.c:167 timerfd_timer_ack: Expected to acknowledge 1 ticks but got 2 instead [Feb 20 15:23:36] DEBUG[32580][C-00000001]: res_rtp_asterisk.c:3393 ast_rtp_raw_write: Difference is 1104, ms is 43 [Feb 20 15:23:36] DEBUG[32580][C-00000001]: res_timing_timerfd.c:167 timerfd_timer_ack: Expected to acknowledge 1 ticks but got 2 instead [Feb 20 15:23:36] DEBUG[32580][C-00000001]: res_rtp_asterisk.c:3393 ast_rtp_raw_write: Difference is 1680, ms is 55 [Feb 20 15:23:36] DEBUG[32580][C-00000001]: res_timing_timerfd.c:167 timerfd_timer_ack: Expected to acknowledge 1 ticks but got 3 instead [Feb 20 15:23:36] DEBUG[32580][C-00000001]: res_rtp_asterisk.c:3393 ast_rtp_raw_write: Difference is 1344, ms is 48 [Feb 20 15:23:36] DEBUG[32580][C-00000001]: res_timing_timerfd.c:167 timerfd_timer_ack: Expected to acknowledge 1 ticks but got 2 instead [Feb 20 15:23:36] DEBUG[32580][C-00000001]: res_rtp_asterisk.c:3393 ast_rtp_raw_write: Difference is 1440, ms is 50 [Feb 20 15:23:36] DEBUG[32580][C-00000001]: res_timing_timerfd.c:167 timerfd_timer_ack: Expected to acknowledge 1 ticks but got 3 instead [Feb 20 15:23:36] DEBUG[32580][C-00000001]: res_rtp_asterisk.c:3393 ast_rtp_raw_write: Difference is 1152, ms is 44 [Feb 20 15:23:36] DEBUG[32580][C-00000001]: res_timing_timerfd.c:167 timerfd_timer_ack: Expected to acknowledge 1 ticks but got 2 instead [Feb 20 15:23:36] DEBUG[32580][C-00000001]: res_rtp_asterisk.c:3393 ast_rtp_raw_write: Difference is 1584, ms is 53 [Feb 20 15:23:36] DEBUG[32580][C-00000001]: res_timing_timerfd.c:167 timerfd_timer_ack: Expected to acknowledge 1 ticks but got 3 instead [Feb 20 15:23:37] DEBUG[32580][C-00000001]: res_rtp_asterisk.c:3393 ast_rtp_raw_write: Difference is 2160, ms is 65 [Feb 20 15:23:37] DEBUG[32580][C-00000001]: res_timing_timerfd.c:167 timerfd_timer_ack: Expected to acknowledge 1 ticks but got 3 instead [Feb 20 15:23:37] DEBUG[32580][C-00000001]: res_rtp_asterisk.c:3393 ast_rtp_raw_write: Difference is 1392, ms is 49 [Feb 20 15:23:37] DEBUG[32580][C-00000001]: res_timing_timerfd.c:167 timerfd_timer_ack: Expected to acknowledge 1 ticks but got 3 instead [Feb 20 15:23:37] DEBUG[32580][C-00000001]: res_rtp_asterisk.c:3393 ast_rtp_raw_write: Difference is 1200, ms is 45 [Feb 20 15:23:37] DEBUG[32580][C-00000001]: res_timing_timerfd.c:167 timerfd_timer_ack: Expected to acknowledge 1 ticks but got 2 instead [Feb 20 15:23:37] DEBUG[32580][C-00000001]: res_rtp_asterisk.c:3393 ast_rtp_raw_write: Difference is 1200, ms is 45 [Feb 20 15:23:37] DEBUG[32580][C-00000001]: res_timing_timerfd.c:167 timerfd_timer_ack: Expected to acknowledge 1 ticks but got 2 instead [Feb 20 15:23:37] DEBUG[32580][C-00000001]: res_rtp_asterisk.c:3393 ast_rtp_raw_write: Difference is 1056, ms is 42 [Feb 20 15:23:37] DEBUG[32580][C-00000001]: res_timing_timerfd.c:167 timerfd_timer_ack: Expected to acknowledge 1 ticks but got 2 instead [Feb 20 15:23:37] DEBUG[32580][C-00000001]: res_rtp_asterisk.c:3393 ast_rtp_raw_write: Difference is 912, ms is 39 [Feb 20 15:23:37] DEBUG[32580][C-00000001]: res_timing_timerfd.c:167 timerfd_timer_ack: Expected to acknowledge 1 ticks but got 2 instead [Feb 20 15:23:37] DEBUG[32580][C-00000001]: res_rtp_asterisk.c:3393 ast_rtp_raw_write: Difference is 1104, ms is 43 [Feb 20 15:23:37] DEBUG[32580][C-00000001]: res_timing_timerfd.c:167 timerfd_timer_ack: Expected to acknowledge 1 ticks but got 3 instead [Feb 20 15:23:37] DEBUG[32580][C-00000001]: res_rtp_asterisk.c:3393 ast_rtp_raw_write: Difference is 1008, ms is 41 [Feb 20 15:23:37] DEBUG[32580][C-00000001]: res_timing_timerfd.c:167 timerfd_timer_ack: Expected to acknowledge 1 ticks but got 2 instead [Feb 20 15:23:37] DEBUG[32580][C-00000001]: res_rtp_asterisk.c:3393 ast_rtp_raw_write: Difference is 672, ms is 34 [Feb 20 15:23:37] DEBUG[32580][C-00000001]: res_timing_timerfd.c:167 timerfd_timer_ack: Expected to acknowledge 1 ticks but got 2 instead [Feb 20 15:23:37] DEBUG[32580][C-00000001]: res_rtp_asterisk.c:3393 ast_rtp_raw_write: Difference is 1776, ms is 57 [Feb 20 15:23:37] DEBUG[32580][C-00000001]: res_timing_timerfd.c:167 timerfd_timer_ack: Expected to acknowledge 1 ticks but got 3 instead [Feb 20 15:23:37] DEBUG[32580][C-00000001]: res_rtp_asterisk.c:3393 ast_rtp_raw_write: Difference is 1728, ms is 56 [Feb 20 15:23:37] DEBUG[32580][C-00000001]: res_timing_timerfd.c:167 timerfd_timer_ack: Expected to acknowledge 1 ticks but got 3 instead [Feb 20 15:23:37] DEBUG[32580][C-00000001]: channel.c:3469 ast_settimeout_full: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Feb 20 15:23:37] DEBUG[32580][C-00000001]: channel.c:3469 ast_settimeout_full: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Feb 20 15:23:37] DEBUG[32580][C-00000001]: channel.c:3469 ast_settimeout_full: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Feb 20 15:23:37] DEBUG[32580][C-00000001]: channel.c:5680 set_format: Channel SIP/50001-00000001 setting write format path: opus -> opus [Feb 20 15:23:37] DEBUG[32572][C-00000001]: res_rtp_asterisk.c:4564 ast_rtp_read: 0x94583a0 -- Probation learning mode pass with source address 192.168.1.147:14369 > 0x94583a0 -- Probation passed - setting RTP source address to 192.168.1.147:14369 [Feb 20 15:23:37] DEBUG[32572][C-00000001]: acl.c:957 ast_ouraddrfor: For destination '192.168.1.147', our source address is '192.168.41.78'. [Feb 20 15:23:37] DEBUG[32572][C-00000001]: res_rtp_asterisk.c:5036 ast_rtp_remote_address_set: Setting RTCP address on RTP instance '0x9ee14c8' [Feb 20 15:23:37] DEBUG[32580][C-00000001]: pbx_lua.c:1480 lua_find_extension: Looking up 350001@default:2 [Feb 20 15:23:37] DEBUG[32580][C-00000001]: pbx.c:2875 pbx_extension_helper: Launching 'Answer' -- Executing [350001@default:2] Answer("SIP/50001-00000001", "") in new stack [Feb 20 15:23:37] DEBUG[32572][C-00000001]: res_rtp_asterisk.c:4564 ast_rtp_read: 0x94583a0 -- Probation learning mode pass with source address 192.168.1.147:14369 > 0x94583a0 -- Probation passed - setting RTP source address to 192.168.1.147:14369 [Feb 20 15:23:37] DEBUG[32580][C-00000001]: pbx_lua.c:1480 lua_find_extension: Looking up 350001@default:3 [Feb 20 15:23:37] DEBUG[32580][C-00000001]: pbx_variables.c:708 pbx_substitute_variables_helper_full: Function CALLERID(num) result is '50001' [Feb 20 15:23:37] DEBUG[32580][C-00000001]: pbx.c:2875 pbx_extension_helper: Launching 'Set' -- Executing [350001@default:3] Set("SIP/50001-00000001", "CID=350001") in new stack [Feb 20 15:23:37] DEBUG[32580][C-00000001]: pbx_lua.c:1480 lua_find_extension: Looking up 350001@default:4 [Feb 20 15:23:37] DEBUG[32580][C-00000001]: pbx_variables.c:381 ast_str_retrieve_variable: Result of 'CID' is '350001' [Feb 20 15:23:37] DEBUG[32580][C-00000001]: pbx_variables.c:381 ast_str_retrieve_variable: Result of 'EXTEN' is '350001' [Feb 20 15:23:37] DEBUG[32580][C-00000001]: pbx_variables.c:777 pbx_substitute_variables_helper_full: Expression result is '1' [Feb 20 15:23:37] DEBUG[32580][C-00000001]: pbx.c:2875 pbx_extension_helper: Launching 'GotoIf' -- Executing [350001@default:4] GotoIf("SIP/50001-00000001", "1?cme") in new stack [Feb 20 15:23:37] DEBUG[32580][C-00000001]: pbx_lua.c:1480 lua_find_extension: Looking up 350001@default:0 -- Goto (default,350001,8) [Feb 20 15:23:37] DEBUG[32580][C-00000001]: pbx_lua.c:1480 lua_find_extension: Looking up 350001@default:8 [Feb 20 15:23:37] DEBUG[32580][C-00000001]: pbx_variables.c:381 ast_str_retrieve_variable: Result of 'EXTEN' is '350001' [Feb 20 15:23:37] DEBUG[32580][C-00000001]: pbx.c:2875 pbx_extension_helper: Launching 'ConfBridge' -- Executing [350001@default:8] ConfBridge("SIP/50001-00000001", "350001,default_bridge_32,default_useradmin,sample_user_menu") in new stack [Feb 20 15:23:37] DEBUG[32580][C-00000001]: app_confbridge.c:1422 join_conference_bridge: Trying to find conference bridge '350001' [Feb 20 15:23:37] DEBUG[32580][C-00000001]: bridge.c:496 find_best_technology: Bridge technology native_rtp does not have any capabilities we want. [Feb 20 15:23:37] DEBUG[32580][C-00000001]: bridge.c:496 find_best_technology: Bridge technology simple_bridge does not have any capabilities we want. [Feb 20 15:23:37] DEBUG[32580][C-00000001]: bridge.c:496 find_best_technology: Bridge technology holding_bridge does not have any capabilities we want. [Feb 20 15:23:37] DEBUG[32580][C-00000001]: bridge.c:515 find_best_technology: Chose bridge technology softmix [Feb 20 15:23:37] DEBUG[32580][C-00000001]: bridge.c:794 bridge_base_init: Bridge 28f1b4ee-63be-49cb-b551-d5a7209f7781: calling softmix technology constructor [Feb 20 15:23:37] DEBUG[32580][C-00000001]: bridge.c:802 bridge_base_init: Bridge 28f1b4ee-63be-49cb-b551-d5a7209f7781: calling softmix technology start [Feb 20 15:23:37] DEBUG[32581]: bridge_softmix.c:1099 softmix_mixing_thread: Bridge 28f1b4ee-63be-49cb-b551-d5a7209f7781: starting mixing thread [Feb 20 15:23:38] DEBUG[32580][C-00000001]: bridge_roles.c:272 setup_bridge_role: Set role 'announcer' [Feb 20 15:23:38] DEBUG[32580][C-00000001]: app_confbridge.c:1367 alloc_playback_chan: Created announcer channel 'CBAnn/350001-00000001;1' to conference bridge '350001' <--- SIP read from UDP:192.168.1.147:23681 ---> INVITE sip:350001@192.168.1.227 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.147:23681;branch=z9hG4bK.ridOrPDhw;rport From: ;tag=X3ughL6Dw To: sip:350001@192.168.1.227 CSeq: 20 INVITE Call-ID: AOK5~O9uyp Max-Forwards: 70 Supported: replaces, outbound Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO, UPDATE Content-Type: application/sdp Content-Length: 466 Contact: ;+sip.instance="" User-Agent: Linphone/3.10.2 (belle-sip/1.5.0) v=0 o=50002 12 2550 IN IP4 192.168.41.114 s=Talk c=IN IP4 192.168.41.114 t=0 0 a=rtcp-xr:rcvr-rtt=all:10000 stat-summary=loss,dup,jitt,TTL voip-metrics m=audio 7078 RTP/AVP 96 101 a=rtpmap:96 opus/48000/2 a=fmtp:96 useinbandfec=1 a=rtpmap:101 telephone-event/48000 a=rtcp-fb:* trr-int 5000 m=video 9078 RTP/AVP 96 a=rtpmap:96 VP8/90000 a=rtcp-fb:* trr-int 5000 a=rtcp-fb:96 nack pli a=rtcp-fb:96 nack sli a=rtcp-fb:96 ack rpsi a=rtcp-fb:96 ccm fir <-------------> [Feb 20 15:23:38] DEBUG[32544]: chan_sip.c:9893 parse_request: Header 0 [ 40]: INVITE sip:350001@192.168.1.227 SIP/2.0 [Feb 20 15:23:38] DEBUG[32544]: chan_sip.c:9893 parse_request: Header 1 [ 69]: Via: SIP/2.0/UDP 192.168.1.147:23681;branch=z9hG4bK.ridOrPDhw;rport [Feb 20 15:23:38] DEBUG[32544]: chan_sip.c:9893 parse_request: Header 2 [ 46]: From: ;tag=X3ughL6Dw [Feb 20 15:23:38] DEBUG[32544]: chan_sip.c:9893 parse_request: Header 3 [ 29]: To: sip:350001@192.168.1.227 [Feb 20 15:23:38] DEBUG[32544]: chan_sip.c:9893 parse_request: Header 4 [ 15]: CSeq: 20 INVITE [Feb 20 15:23:38] DEBUG[32544]: chan_sip.c:9893 parse_request: Header 5 [ 19]: Call-ID: AOK5~O9uyp [Feb 20 15:23:38] DEBUG[32544]: chan_sip.c:9893 parse_request: Header 6 [ 16]: Max-Forwards: 70 [Feb 20 15:23:38] DEBUG[32544]: chan_sip.c:9893 parse_request: Header 7 [ 29]: Supported: replaces, outbound [Feb 20 15:23:38] DEBUG[32544]: chan_sip.c:9893 parse_request: Header 8 [ 89]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO, UPDATE [Feb 20 15:23:38] DEBUG[32544]: chan_sip.c:9893 parse_request: Header 9 [ 29]: Content-Type: application/sdp [Feb 20 15:23:38] DEBUG[32544]: chan_sip.c:9893 parse_request: Header 10 [ 19]: Content-Length: 466 [Feb 20 15:23:38] DEBUG[32544]: chan_sip.c:9893 parse_request: Header 11 [120]: Contact: ;+sip.instance="" [Feb 20 15:23:38] DEBUG[32580][C-00000001]: app_confbridge.c:1524 join_conference_bridge: Created conference '350001' and linked to container. [Feb 20 15:23:38] DEBUG[32583]: bridge_channel.c:2654 bridge_channel_internal_join: Bridge 28f1b4ee-63be-49cb-b551-d5a7209f7781: 0x9335418(CBAnn/350001-00000001;2) is joining [Feb 20 15:23:38] DEBUG[32544]: chan_sip.c:9893 parse_request: Header 12 [ 45]: User-Agent: Linphone/3.10.2 (belle-sip/1.5.0) [Feb 20 15:23:38] DEBUG[32544]: chan_sip.c:9893 parse_request: Header 13 [ 0]: [Feb 20 15:23:38] DEBUG[32544]: chan_sip.c:9893 parse_request: Body 0 [ 3]: v=0 [Feb 20 15:23:38] DEBUG[32544]: chan_sip.c:9893 parse_request: Body 1 [ 37]: o=50002 12 2550 IN IP4 192.168.41.114 [Feb 20 15:23:38] DEBUG[32544]: chan_sip.c:9893 parse_request: Body 2 [ 6]: s=Talk [Feb 20 15:23:38] DEBUG[32544]: chan_sip.c:9893 parse_request: Body 3 [ 23]: c=IN IP4 192.168.41.114 [Feb 20 15:23:38] DEBUG[32544]: chan_sip.c:9893 parse_request: Body 4 [ 5]: t=0 0 [Feb 20 15:23:38] DEBUG[32544]: chan_sip.c:9893 parse_request: Body 5 [ 72]: a=rtcp-xr:rcvr-rtt=all:10000 stat-summary=loss,dup,jitt,TTL voip-metrics [Feb 20 15:23:38] DEBUG[32566]: app_queue.c:2482 device_state_cb: Device 'confbridge:350001' changed to state '2' (In use) but we don't care because they're not a member of any queue. [Feb 20 15:23:38] DEBUG[32544]: chan_sip.c:9893 parse_request: Body 6 [ 27]: m=audio 7078 RTP/AVP 96 101 [Feb 20 15:23:38] DEBUG[32580][C-00000001]: confbridge/conf_state.c:84 conf_change_state: Changing conference '350001' state from EMPTY to SINGLE_MARKED [Feb 20 15:23:38] DEBUG[32544]: chan_sip.c:9893 parse_request: Body 7 [ 24]: a=rtpmap:96 opus/48000/2 [Feb 20 15:23:38] DEBUG[32544]: chan_sip.c:9893 parse_request: Body 8 [ 24]: a=fmtp:96 useinbandfec=1 [Feb 20 15:23:38] DEBUG[32544]: chan_sip.c:9893 parse_request: Body 9 [ 34]: a=rtpmap:101 telephone-event/48000 [Feb 20 15:23:38] DEBUG[32544]: chan_sip.c:9893 parse_request: Body 10 [ 24]: a=rtcp-fb:* trr-int 5000 [Feb 20 15:23:38] DEBUG[32583]: bridge_channel.c:2118 bridge_channel_internal_push_full: Bridge 28f1b4ee-63be-49cb-b551-d5a7209f7781: pushing 0x9335418(CBAnn/350001-00000001;2) [Feb 20 15:23:38] DEBUG[32583]: bridge_roles.c:272 setup_bridge_role: Set role 'announcer' -- Channel CBAnn/350001-00000001;2 joined 'softmix' base-bridge <28f1b4ee-63be-49cb-b551-d5a7209f7781> [Feb 20 15:23:38] DEBUG[32544]: chan_sip.c:9893 parse_request: Body 11 [ 23]: m=video 9078 RTP/AVP 96 [Feb 20 15:23:38] DEBUG[32544]: chan_sip.c:9893 parse_request: Body 12 [ 21]: a=rtpmap:96 VP8/90000 [Feb 20 15:23:38] DEBUG[32544]: chan_sip.c:9893 parse_request: Body 13 [ 24]: a=rtcp-fb:* trr-int 5000 [Feb 20 15:23:38] DEBUG[32544]: chan_sip.c:9893 parse_request: Body 14 [ 21]: a=rtcp-fb:96 nack pli [Feb 20 15:23:38] DEBUG[32544]: chan_sip.c:9893 parse_request: Body 15 [ 21]: a=rtcp-fb:96 nack sli [Feb 20 15:23:38] DEBUG[32544]: chan_sip.c:9893 parse_request: Body 16 [ 21]: a=rtcp-fb:96 ack rpsi [Feb 20 15:23:38] DEBUG[32544]: chan_sip.c:9930 parse_request: Body 17 [ 20]: a=rtcp-fb:96 ccm fir [Feb 20 15:23:38] DEBUG[32580][C-00000001]: app_confbridge.c:1145 conf_update_user_mute: User SIP/50001-00000001 is unmuted: user:0 system:0. --- (13 headers 18 lines) --- [Feb 20 15:23:38] DEBUG[32544]: chan_sip.c:9419 __find_call: = Looking for Call ID: AOK5~O9uyp (Checking From) --From tag X3ughL6Dw --To-tag [Feb 20 15:23:38] DEBUG[32544]: acl.c:957 ast_ouraddrfor: For destination '192.168.1.147', our source address is '192.168.41.78'. [Feb 20 15:23:38] DEBUG[32544]: chan_sip.c:3877 ast_sip_ouraddrfor: Target address 192.168.1.147:23681 is not local, substituting externaddr [Feb 20 15:23:38] DEBUG[32544]: chan_sip.c:3910 ast_sip_ouraddrfor: Setting AST_TRANSPORT_UDP with address 192.168.1.227:5060 [Feb 20 15:23:38] DEBUG[32544]: netsock2.c:172 ast_sockaddr_split_hostport: Splitting '192.168.1.147:23681' into... [Feb 20 15:23:38] DEBUG[32544]: netsock2.c:226 ast_sockaddr_split_hostport: ...host '192.168.1.147' and port '23681'. [Feb 20 15:23:38] DEBUG[32583]: bridge.c:432 bridge_channel_complete_join: Bridge 28f1b4ee-63be-49cb-b551-d5a7209f7781: 0x9335418(CBAnn/350001-00000001;2) is joining softmix technology Sending to 192.168.1.147:23681 (no NAT) [Feb 20 15:23:38] DEBUG[32544]: chan_sip.c:9006 __sip_alloc: Allocating new SIP dialog for AOK5~O9uyp - INVITE (No RTP) [Feb 20 15:23:38] DEBUG[32544][C-00000002]: chan_sip.c:28652 handle_incoming: **** Received INVITE (5) - Command in SIP INVITE [Feb 20 15:23:38] DEBUG[32583]: dsp.c:499 ast_tone_detect_init: Setup tone 1100 Hz, 500 ms, block_size=160, hits_required=21 [Feb 20 15:23:38] DEBUG[32583]: dsp.c:499 ast_tone_detect_init: Setup tone 2100 Hz, 2600 ms, block_size=160, hits_required=116 [Feb 20 15:23:38] DEBUG[32544][C-00000002]: sip/reqresp_parser.c:1711 parse_sip_options: Begin: parsing SIP "Supported: replaces, outbound" [Feb 20 15:23:38] DEBUG[32544][C-00000002]: sip/reqresp_parser.c:1726 parse_sip_options: Found SIP option: -replaces- [Feb 20 15:23:38] DEBUG[32544][C-00000002]: sip/reqresp_parser.c:1734 parse_sip_options: Matched SIP option: replaces [Feb 20 15:23:38] DEBUG[32544][C-00000002]: sip/reqresp_parser.c:1726 parse_sip_options: Found SIP option: -outbound- [Feb 20 15:23:38] DEBUG[32544][C-00000002]: sip/reqresp_parser.c:1734 parse_sip_options: Matched SIP option: outbound [Feb 20 15:23:38] DEBUG[32544][C-00000002]: netsock2.c:172 ast_sockaddr_split_hostport: Splitting '192.168.1.147:23681' into... [Feb 20 15:23:38] DEBUG[32544][C-00000002]: netsock2.c:226 ast_sockaddr_split_hostport: ...host '192.168.1.147' and port '23681'. Sending to 192.168.1.147:23681 (no NAT) [Feb 20 15:23:38] DEBUG[32544][C-00000002]: chan_sip.c:26104 handle_request_invite: Initializing initreq for method INVITE - callid AOK5~O9uyp Using INVITE request as basis request - AOK5~O9uyp [Feb 20 15:23:38] DEBUG[32580][C-00000001]: bridge_channel.c:2654 bridge_channel_internal_join: Bridge 28f1b4ee-63be-49cb-b551-d5a7209f7781: 0xa01f998(SIP/50001-00000001) is joining [Feb 20 15:23:38] DEBUG[32544][C-00000002]: netsock2.c:172 ast_sockaddr_split_hostport: Splitting '192.168.1.227' into... [Feb 20 15:23:38] DEBUG[32544][C-00000002]: netsock2.c:226 ast_sockaddr_split_hostport: ...host '192.168.1.227' and port ''. Found peer '50002' for '50002' from 192.168.1.147:23681 <--- Reliably Transmitting (no NAT) to 192.168.1.147:23681 ---> SIP/2.0 401 Unauthorized Via: SIP/2.0/UDP 192.168.1.147:23681;branch=z9hG4bK.ridOrPDhw;received=192.168.1.147;rport=23681 From: ;tag=X3ughL6Dw To: sip:350001@192.168.1.227;tag=as2b490e89 Call-ID: AOK5~O9uyp CSeq: 20 INVITE Server: Asterisk PBX 13.14.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk", nonce="2d0b2d61" Content-Length: 0 <------------> [Feb 20 15:23:38] DEBUG[32544][C-00000002]: chan_sip.c:4266 __sip_reliable_xmit: *** SIP TIMER: Initializing retransmit timer on packet: Id #6 [Feb 20 15:23:38] DEBUG[32544][C-00000002]: chan_sip.c:3753 __sip_xmit: Trying to put 'SIP/2.0 401' onto UDP socket destined for 192.168.1.147:23681 Scheduling destruction of SIP dialog 'AOK5~O9uyp' in 32000 ms (Method: INVITE) [Feb 20 15:23:38] DEBUG[32580][C-00000001]: bridge_channel.c:2118 bridge_channel_internal_push_full: Bridge 28f1b4ee-63be-49cb-b551-d5a7209f7781: pushing 0xa01f998(SIP/50001-00000001) -- Channel SIP/50001-00000001 joined 'softmix' base-bridge <28f1b4ee-63be-49cb-b551-d5a7209f7781> <--- SIP read from UDP:192.168.1.147:23681 ---> ACK sip:350001@192.168.1.227 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.147:23681;branch=z9hG4bK.ridOrPDhw;rport Call-ID: AOK5~O9uyp From: ;tag=X3ughL6Dw To: ;tag=as2b490e89 Contact: ;+sip.instance="" Max-Forwards: 70 CSeq: 20 ACK <-------------> [Feb 20 15:23:38] DEBUG[32544]: chan_sip.c:9893 parse_request: Header 0 [ 37]: ACK sip:350001@192.168.1.227 SIP/2.0 [Feb 20 15:23:38] DEBUG[32544]: chan_sip.c:9893 parse_request: Header 1 [ 69]: Via: SIP/2.0/UDP 192.168.1.147:23681;branch=z9hG4bK.ridOrPDhw;rport [Feb 20 15:23:38] DEBUG[32544]: chan_sip.c:9893 parse_request: Header 2 [ 19]: Call-ID: AOK5~O9uyp [Feb 20 15:23:38] DEBUG[32496]: cdr.c:1293 cdr_object_finalize: Finalized CDR for SIP/50001-00000001 - start 1487575415.698644 answer 1487575416.073465 end 1487575418.324973 dispo ANSWERED [Feb 20 15:23:38] DEBUG[32544]: chan_sip.c:9893 parse_request: Header 3 [ 46]: From: ;tag=X3ughL6Dw [Feb 20 15:23:38] DEBUG[32544]: chan_sip.c:9893 parse_request: Header 4 [ 46]: To: ;tag=as2b490e89 [Feb 20 15:23:38] DEBUG[32544]: chan_sip.c:9893 parse_request: Header 5 [120]: Contact: ;+sip.instance="" [Feb 20 15:23:38] DEBUG[32544]: chan_sip.c:9893 parse_request: Header 6 [ 16]: Max-Forwards: 70 [Feb 20 15:23:38] DEBUG[32544]: chan_sip.c:9893 parse_request: Header 7 [ 12]: CSeq: 20 ACK --- (8 headers 0 lines) --- [Feb 20 15:23:38] DEBUG[32580][C-00000001]: bridge.c:432 bridge_channel_complete_join: Bridge 28f1b4ee-63be-49cb-b551-d5a7209f7781: 0xa01f998(SIP/50001-00000001) is joining softmix technology [Feb 20 15:23:38] DEBUG[32544]: chan_sip.c:9419 __find_call: = Looking for Call ID: AOK5~O9uyp (Checking From) --From tag X3ughL6Dw --To-tag as2b490e89 [Feb 20 15:23:38] DEBUG[32580][C-00000001]: channel.c:5444 ast_set_read_format_path: Channel SIP/50001-00000001 setting read format path: opus -> slin [Feb 20 15:23:38] DEBUG[32544][C-00000002]: chan_sip.c:28652 handle_incoming: **** Received ACK (6) - Command in SIP ACK [Feb 20 15:23:38] DEBUG[32544][C-00000002]: chan_sip.c:4526 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #6 [Feb 20 15:23:38] DEBUG[32544][C-00000002]: chan_sip.c:4537 __sip_ack: Stopping retransmission on 'AOK5~O9uyp' of Response 20: Match Found <--- SIP read from UDP:192.168.1.147:23681 ---> INVITE sip:350001@192.168.1.227 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.147:23681;branch=z9hG4bK.tvdYczhsY;rport From: ;tag=X3ughL6Dw To: sip:350001@192.168.1.227 CSeq: 21 INVITE Call-ID: AOK5~O9uyp Max-Forwards: 70 Supported: replaces, outbound Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO, UPDATE Content-Type: application/sdp Content-Length: 466 Contact: ;+sip.instance="" User-Agent: Linphone/3.10.2 (belle-sip/1.5.0) Authorization: Digest realm="asterisk", nonce="2d0b2d61", algorithm=MD5, username="50002", uri="sip:350001@192.168.1.227", response="fdf077e97670bda2fdd4119dbaef0eb6" v=0 o=50002 12 2550 IN IP4 192.168.41.114 s=Talk c=IN IP4 192.168.41.114 t=0 0 a=rtcp-xr:rcvr-rtt=all:10000 stat-summary=loss,dup,jitt,TTL voip-metrics m=audio 7078 RTP/AVP 96 101 a=rtpmap:96 opus/48000/2 a=fmtp:96 useinbandfec=1 a=rtpmap:101 telephone-event/48000 a=rtcp-fb:* trr-int 5000 m=video 9078 RTP/AVP 96 a=rtpmap:96 VP8/90000 a=rtcp-fb:* trr-int 5000 a=rtcp-fb:96 nack pli a=rtcp-fb:96 nack sli a=rtcp-fb:96 ack rpsi a=rtcp-fb:96 ccm fir <-------------> [Feb 20 15:23:38] DEBUG[32544]: chan_sip.c:9893 parse_request: Header 0 [ 40]: INVITE sip:350001@192.168.1.227 SIP/2.0 [Feb 20 15:23:38] DEBUG[32544]: chan_sip.c:9893 parse_request: Header 1 [ 69]: Via: SIP/2.0/UDP 192.168.1.147:23681;branch=z9hG4bK.tvdYczhsY;rport [Feb 20 15:23:38] DEBUG[32544]: chan_sip.c:9893 parse_request: Header 2 [ 46]: From: ;tag=X3ughL6Dw [Feb 20 15:23:38] DEBUG[32544]: chan_sip.c:9893 parse_request: Header 3 [ 29]: To: sip:350001@192.168.1.227 [Feb 20 15:23:38] DEBUG[32544]: chan_sip.c:9893 parse_request: Header 4 [ 15]: CSeq: 21 INVITE [Feb 20 15:23:38] DEBUG[32544]: chan_sip.c:9893 parse_request: Header 5 [ 19]: Call-ID: AOK5~O9uyp [Feb 20 15:23:38] DEBUG[32544]: chan_sip.c:9893 parse_request: Header 6 [ 16]: Max-Forwards: 70 [Feb 20 15:23:38] DEBUG[32544]: chan_sip.c:9893 parse_request: Header 7 [ 29]: Supported: replaces, outbound [Feb 20 15:23:38] DEBUG[32544]: chan_sip.c:9893 parse_request: Header 8 [ 89]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO, UPDATE [Feb 20 15:23:38] DEBUG[32544]: chan_sip.c:9893 parse_request: Header 9 [ 29]: Content-Type: application/sdp [Feb 20 15:23:38] DEBUG[32544]: chan_sip.c:9893 parse_request: Header 10 [ 19]: Content-Length: 466 [Feb 20 15:23:38] DEBUG[32544]: chan_sip.c:9893 parse_request: Header 11 [120]: Contact: ;+sip.instance="" [Feb 20 15:23:38] DEBUG[32544]: chan_sip.c:9893 parse_request: Header 12 [ 45]: User-Agent: Linphone/3.10.2 (belle-sip/1.5.0) [Feb 20 15:23:38] DEBUG[32544]: chan_sip.c:9893 parse_request: Header 13 [167]: Authorization: Digest realm="asterisk", nonce="2d0b2d61", algorithm=MD5, username="50002", uri="sip:350001@192.168.1.227", response="fdf077e97670bda2fdd4119dbaef0eb6" [Feb 20 15:23:38] DEBUG[32580][C-00000001]: channel.c:5680 set_format: Channel SIP/50001-00000001 setting write format path: slin -> opus [Feb 20 15:23:38] DEBUG[32580][C-00000001]: dsp.c:499 ast_tone_detect_init: Setup tone 1100 Hz, 500 ms, block_size=160, hits_required=21 [Feb 20 15:23:38] DEBUG[32580][C-00000001]: dsp.c:499 ast_tone_detect_init: Setup tone 2100 Hz, 2600 ms, block_size=160, hits_required=116 [Feb 20 15:23:38] DEBUG[32544]: chan_sip.c:9893 parse_request: Header 14 [ 0]: [Feb 20 15:23:38] DEBUG[32544]: chan_sip.c:9893 parse_request: Body 0 [ 3]: v=0 [Feb 20 15:23:38] DEBUG[32544]: chan_sip.c:9893 parse_request: Body 1 [ 37]: o=50002 12 2550 IN IP4 192.168.41.114 [Feb 20 15:23:38] DEBUG[32544]: chan_sip.c:9893 parse_request: Body 2 [ 6]: s=Talk [Feb 20 15:23:38] DEBUG[32544]: chan_sip.c:9893 parse_request: Body 3 [ 23]: c=IN IP4 192.168.41.114 [Feb 20 15:23:38] DEBUG[32544]: chan_sip.c:9893 parse_request: Body 4 [ 5]: t=0 0 [Feb 20 15:23:38] DEBUG[32544]: chan_sip.c:9893 parse_request: Body 5 [ 72]: a=rtcp-xr:rcvr-rtt=all:10000 stat-summary=loss,dup,jitt,TTL voip-metrics [Feb 20 15:23:38] DEBUG[32544]: chan_sip.c:9893 parse_request: Body 6 [ 27]: m=audio 7078 RTP/AVP 96 101 [Feb 20 15:23:38] DEBUG[32544]: chan_sip.c:9893 parse_request: Body 7 [ 24]: a=rtpmap:96 opus/48000/2 [Feb 20 15:23:38] DEBUG[32544]: chan_sip.c:9893 parse_request: Body 8 [ 24]: a=fmtp:96 useinbandfec=1 [Feb 20 15:23:38] DEBUG[32544]: chan_sip.c:9893 parse_request: Body 9 [ 34]: a=rtpmap:101 telephone-event/48000 [Feb 20 15:23:38] DEBUG[32544]: chan_sip.c:9893 parse_request: Body 10 [ 24]: a=rtcp-fb:* trr-int 5000 [Feb 20 15:23:38] DEBUG[32544]: chan_sip.c:9893 parse_request: Body 11 [ 23]: m=video 9078 RTP/AVP 96 [Feb 20 15:23:38] DEBUG[32544]: chan_sip.c:9893 parse_request: Body 12 [ 21]: a=rtpmap:96 VP8/90000 [Feb 20 15:23:38] DEBUG[32544]: chan_sip.c:9893 parse_request: Body 13 [ 24]: a=rtcp-fb:* trr-int 5000 [Feb 20 15:23:38] DEBUG[32544]: chan_sip.c:9893 parse_request: Body 14 [ 21]: a=rtcp-fb:96 nack pli [Feb 20 15:23:38] DEBUG[32544]: chan_sip.c:9893 parse_request: Body 15 [ 21]: a=rtcp-fb:96 nack sli [Feb 20 15:23:38] DEBUG[32544]: chan_sip.c:9893 parse_request: Body 16 [ 21]: a=rtcp-fb:96 ack rpsi [Feb 20 15:23:38] DEBUG[32544]: chan_sip.c:9930 parse_request: Body 17 [ 20]: a=rtcp-fb:96 ccm fir --- (14 headers 18 lines) --- [Feb 20 15:23:38] DEBUG[32544]: chan_sip.c:9419 __find_call: = Looking for Call ID: AOK5~O9uyp (Checking From) --From tag X3ughL6Dw --To-tag [Feb 20 15:23:38] DEBUG[32544]: netsock2.c:172 ast_sockaddr_split_hostport: Splitting '192.168.1.227' into... [Feb 20 15:23:38] DEBUG[32544]: netsock2.c:226 ast_sockaddr_split_hostport: ...host '192.168.1.227' and port ''. [Feb 20 15:23:38] DEBUG[32544]: netsock2.c:172 ast_sockaddr_split_hostport: Splitting '192.168.1.227' into... [Feb 20 15:23:38] DEBUG[32544]: netsock2.c:226 ast_sockaddr_split_hostport: ...host '192.168.1.227' and port ''. [Feb 20 15:23:38] DEBUG[32544][C-00000002]: chan_sip.c:28652 handle_incoming: **** Received INVITE (5) - Command in SIP INVITE [Feb 20 15:23:38] DEBUG[32544][C-00000002]: netsock2.c:172 ast_sockaddr_split_hostport: Splitting '192.168.1.147:23681' into... [Feb 20 15:23:38] DEBUG[32544][C-00000002]: netsock2.c:226 ast_sockaddr_split_hostport: ...host '192.168.1.147' and port '23681'. Sending to 192.168.1.147:23681 (no NAT) [Feb 20 15:23:38] DEBUG[32544][C-00000002]: chan_sip.c:26104 handle_request_invite: Initializing initreq for method INVITE - callid AOK5~O9uyp Using INVITE request as basis request - AOK5~O9uyp [Feb 20 15:23:38] DEBUG[32544][C-00000002]: netsock2.c:172 ast_sockaddr_split_hostport: Splitting '192.168.1.227' into... [Feb 20 15:23:38] DEBUG[32544][C-00000002]: netsock2.c:226 ast_sockaddr_split_hostport: ...host '192.168.1.227' and port ''. Found peer '50002' for '50002' from 192.168.1.147:23681 [Feb 20 15:23:38] DEBUG[32544][C-00000002]: rtp_engine.c:454 ast_rtp_instance_new: Using engine 'asterisk' for RTP instance '0x99a1948' [Feb 20 15:23:38] DEBUG[32544][C-00000002]: res_rtp_asterisk.c:2681 ast_rtp_new: Allocated port 15176 for RTP instance '0x99a1948' [Feb 20 15:23:38] DEBUG[32544][C-00000002]: res_rtp_asterisk.c:2708 ast_rtp_new: Creating ICE session 0.0.0.0:15176 (15176) for RTP instance '0x99a1948' [Feb 20 15:23:38] DEBUG[32544][C-00000002]: netsock2.c:172 ast_sockaddr_split_hostport: Splitting '192.168.41.78' into... [Feb 20 15:23:38] DEBUG[32544][C-00000002]: netsock2.c:226 ast_sockaddr_split_hostport: ...host '192.168.41.78' and port ''. [Feb 20 15:23:38] DEBUG[32544][C-00000002]: netsock2.c:172 ast_sockaddr_split_hostport: Splitting '192.168.41.78' into... [Feb 20 15:23:38] DEBUG[32544][C-00000002]: netsock2.c:226 ast_sockaddr_split_hostport: ...host '192.168.41.78' and port ''. [Feb 20 15:23:38] DEBUG[32544][C-00000002]: rtp_engine.c:463 ast_rtp_instance_new: RTP instance '0x99a1948' is setup and ready to go [Feb 20 15:23:38] DEBUG[32581]: res_timing_timerfd.c:167 timerfd_timer_ack: Expected to acknowledge 1 ticks but got 18 instead [Feb 20 15:23:38] DEBUG[32544][C-00000002]: rtp_engine.c:454 ast_rtp_instance_new: Using engine 'asterisk' for RTP instance '0xa5db3e8' [Feb 20 15:23:38] DEBUG[32544][C-00000002]: res_rtp_asterisk.c:2681 ast_rtp_new: Allocated port 19328 for RTP instance '0xa5db3e8' [Feb 20 15:23:38] DEBUG[32544][C-00000002]: res_rtp_asterisk.c:2708 ast_rtp_new: Creating ICE session 0.0.0.0:19328 (19328) for RTP instance '0xa5db3e8' [Feb 20 15:23:38] DEBUG[32580][C-00000001]: res_rtp_asterisk.c:3052 ast_rtp_update_source: Setting the marker bit due to a source update [Feb 20 15:23:38] DEBUG[32544][C-00000002]: netsock2.c:172 ast_sockaddr_split_hostport: Splitting '192.168.41.78' into... [Feb 20 15:23:38] DEBUG[32580][C-00000001]: res_rtp_asterisk.c:3052 ast_rtp_update_source: Setting the marker bit due to a source update [Feb 20 15:23:38] DEBUG[32544][C-00000002]: netsock2.c:226 ast_sockaddr_split_hostport: ...host '192.168.41.78' and port ''. [Feb 20 15:23:38] DEBUG[32544][C-00000002]: netsock2.c:172 ast_sockaddr_split_hostport: Splitting '192.168.41.78' into... [Feb 20 15:23:38] DEBUG[32544][C-00000002]: netsock2.c:226 ast_sockaddr_split_hostport: ...host '192.168.41.78' and port ''. [Feb 20 15:23:38] DEBUG[32582]: channel.c:5680 set_format: Channel CBAnn/350001-00000001;1 setting write format path: gsm -> slin [Feb 20 15:23:38] DEBUG[32544][C-00000002]: rtp_engine.c:463 ast_rtp_instance_new: RTP instance '0xa5db3e8' is setup and ready to go [Feb 20 15:23:38] DEBUG[32544][C-00000002]: netsock2.c:172 ast_sockaddr_split_hostport: Splitting 'iZ94rhgsz5zZ' into... [Feb 20 15:23:38] DEBUG[32582]: channel.c:3469 ast_settimeout_full: Scheduling timer at (50 requested / 50 actual) timer ticks per second [Feb 20 15:23:38] DEBUG[32544][C-00000002]: netsock2.c:226 ast_sockaddr_split_hostport: ...host 'iZ94rhgsz5zZ' and port ''. -- Playing 'confbridge-join.gsm' (language 'en') [Feb 20 15:23:38] DEBUG[32544][C-00000002]: res_rtp_asterisk.c:4967 ast_rtp_prop_set: Setup RTCP on RTP instance '0xa5db3e8' == Using SIP VIDEO CoS mark 6 [Feb 20 15:23:38] DEBUG[32580][C-00000001]: res_rtp_asterisk.c:3393 ast_rtp_raw_write: Difference is 56736, ms is 1202 [Feb 20 15:23:38] DEBUG[32582]: res_timing_timerfd.c:167 timerfd_timer_ack: Expected to acknowledge 1 ticks but got 2 instead [Feb 20 15:23:38] DEBUG[32544][C-00000002]: netsock2.c:172 ast_sockaddr_split_hostport: Splitting 'iZ94rhgsz5zZ' into... [Feb 20 15:23:38] DEBUG[32544][C-00000002]: netsock2.c:226 ast_sockaddr_split_hostport: ...host 'iZ94rhgsz5zZ' and port ''. [Feb 20 15:23:38] DEBUG[32544][C-00000002]: res_rtp_asterisk.c:4967 ast_rtp_prop_set: Setup RTCP on RTP instance '0x99a1948' == Using SIP RTP CoS mark 5 [Feb 20 15:23:38] DEBUG[32544][C-00000002]: chan_sip.c:5798 do_setnat: Setting NAT on RTP to On [Feb 20 15:23:38] DEBUG[32582]: res_timing_timerfd.c:167 timerfd_timer_ack: Expected to acknowledge 1 ticks but got 2 instead [Feb 20 15:23:38] DEBUG[32580][C-00000001]: res_rtp_asterisk.c:3393 ast_rtp_raw_write: Difference is 3072, ms is 84 [Feb 20 15:23:38] DEBUG[32544][C-00000002]: chan_sip.c:5802 do_setnat: Setting NAT on VRTP to On [Feb 20 15:23:38] DEBUG[32544][C-00000002]: chan_sip.c:10271 process_sdp: Processing session-level SDP v=0... UNSUPPORTED OR FAILED. [Feb 20 15:23:38] DEBUG[32544][C-00000002]: chan_sip.c:10271 process_sdp: Processing session-level SDP o=50002 12 2550 IN IP4 192.168.41.114... OK. [Feb 20 15:23:38] DEBUG[32544][C-00000002]: chan_sip.c:10271 process_sdp: Processing session-level SDP s=Talk... UNSUPPORTED OR FAILED. [Feb 20 15:23:38] DEBUG[32544][C-00000002]: netsock2.c:172 ast_sockaddr_split_hostport: Splitting '192.168.41.114' into... [Feb 20 15:23:38] DEBUG[32544][C-00000002]: netsock2.c:226 ast_sockaddr_split_hostport: ...host '192.168.41.114' and port ''. [Feb 20 15:23:38] DEBUG[32582]: res_timing_timerfd.c:167 timerfd_timer_ack: Expected to acknowledge 1 ticks but got 3 instead [Feb 20 15:23:38] DEBUG[32544][C-00000002]: chan_sip.c:10271 process_sdp: Processing session-level SDP c=IN IP4 192.168.41.114... OK. [Feb 20 15:23:38] DEBUG[32544][C-00000002]: chan_sip.c:10271 process_sdp: Processing session-level SDP t=0 0... UNSUPPORTED OR FAILED. [Feb 20 15:23:38] DEBUG[32544][C-00000002]: chan_sip.c:10271 process_sdp: Processing session-level SDP a=rtcp-xr:rcvr-rtt=all:10000 stat-summary=loss,dup,jitt,TTL voip-metrics... UNSUPPORTED OR FAILED. Found RTP audio format 96 [Feb 20 15:23:38] DEBUG[32544][C-00000002]: rtp_engine.c:684 ast_rtp_codecs_payloads_set_m_type: Don't have a default tx payload type 96 format for m type on 0x47403080 Found RTP audio format 101 [Feb 20 15:23:38] DEBUG[32544][C-00000002]: rtp_engine.c:689 ast_rtp_codecs_payloads_set_m_type: Setting tx payload type 101 based on m type on 0x47403080 [Feb 20 15:23:38] DEBUG[32580][C-00000001]: res_rtp_asterisk.c:3393 ast_rtp_raw_write: Difference is 2160, ms is 65 [Feb 20 15:23:38] DEBUG[32582]: res_timing_timerfd.c:167 timerfd_timer_ack: Expected to acknowledge 1 ticks but got 2 instead Found audio description format opus for ID 96 [Feb 20 15:23:38] DEBUG[32544][C-00000002]: chan_sip.c:10726 process_sdp: Processing media-level (audio) SDP a=rtpmap:96 opus/48000/2... OK. [Feb 20 15:23:38] DEBUG[32544][C-00000002]: chan_sip.c:10726 process_sdp: Processing media-level (audio) SDP a=fmtp:96 useinbandfec=1... OK. [Feb 20 15:23:38] DEBUG[32544][C-00000002]: rtp_engine.c:791 ast_rtp_codecs_payloads_unset: Unsetting payload 101 on 0x47403080 Found unknown media description format telephone-event for ID 101 [Feb 20 15:23:38] DEBUG[32544][C-00000002]: chan_sip.c:10726 process_sdp: Processing media-level (audio) SDP a=rtpmap:101 telephone-event/48000... UNSUPPORTED OR FAILED. [Feb 20 15:23:38] DEBUG[32544][C-00000002]: chan_sip.c:10726 process_sdp: Processing media-level (audio) SDP a=rtcp-fb:* trr-int 5000... UNSUPPORTED OR FAILED. Found RTP video format 96 [Feb 20 15:23:38] DEBUG[32544][C-00000002]: rtp_engine.c:684 ast_rtp_codecs_payloads_set_m_type: Don't have a default tx payload type 96 format for m type on 0x47403010 Found video description format VP8 for ID 96 [Feb 20 15:23:38] DEBUG[32580][C-00000001]: res_rtp_asterisk.c:3393 ast_rtp_raw_write: Difference is 3360, ms is 90 [Feb 20 15:23:38] DEBUG[32582]: res_timing_timerfd.c:167 timerfd_timer_ack: Expected to acknowledge 1 ticks but got 3 instead [Feb 20 15:23:38] DEBUG[32544][C-00000002]: chan_sip.c:10726 process_sdp: Processing media-level (video) SDP a=rtpmap:96 VP8/90000... OK. [Feb 20 15:23:38] DEBUG[32582]: res_timing_timerfd.c:167 timerfd_timer_ack: Expected to acknowledge 1 ticks but got 2 instead [Feb 20 15:23:38] DEBUG[32544][C-00000002]: chan_sip.c:10726 process_sdp: Processing media-level (video) SDP a=rtcp-fb:* trr-int 5000... UNSUPPORTED OR FAILED. [Feb 20 15:23:39] DEBUG[32544][C-00000002]: chan_sip.c:10726 process_sdp: Processing media-level (video) SDP a=rtcp-fb:96 nack pli... UNSUPPORTED OR FAILED. [Feb 20 15:23:39] DEBUG[32544][C-00000002]: chan_sip.c:10726 process_sdp: Processing media-level (video) SDP a=rtcp-fb:96 nack sli... UNSUPPORTED OR FAILED. [Feb 20 15:23:39] DEBUG[32544][C-00000002]: chan_sip.c:10726 process_sdp: Processing media-level (video) SDP a=rtcp-fb:96 ack rpsi... UNSUPPORTED OR FAILED. [Feb 20 15:23:39] DEBUG[32544][C-00000002]: chan_sip.c:10726 process_sdp: Processing media-level (video) SDP a=rtcp-fb:96 ccm fir... UNSUPPORTED OR FAILED. [Feb 20 15:23:39] DEBUG[32582]: res_timing_timerfd.c:167 timerfd_timer_ack: Expected to acknowledge 1 ticks but got 2 instead [Feb 20 15:23:39] DEBUG[32580][C-00000001]: res_rtp_asterisk.c:3393 ast_rtp_raw_write: Difference is 2928, ms is 81 Capabilities: us - (opus|vp8), peer - audio=(opus)/video=(vp8)/text=(nothing), combined - (opus|vp8) Non-codec capabilities (dtmf): us - 0x0 (nothing), peer - 0x0 (nothing), combined - 0x0 (nothing) [Feb 20 15:23:39] DEBUG[32544][C-00000002]: acl.c:957 ast_ouraddrfor: For destination '192.168.41.114', our source address is '192.168.41.78'. [Feb 20 15:23:39] DEBUG[32544][C-00000002]: res_rtp_asterisk.c:5036 ast_rtp_remote_address_set: Setting RTCP address on RTP instance '0x99a1948' Peer audio RTP is at port 192.168.41.114:7078 [Feb 20 15:23:39] DEBUG[32544][C-00000002]: rtp_engine.c:658 ast_rtp_codecs_payloads_copy: Copying payload 96 (0xbecab70) from 0x47403080 to 0x99a1b10 [Feb 20 15:23:39] DEBUG[32544][C-00000002]: res_rtp_asterisk.c:4916 ast_rtp_prop_set: Ignoring duplicate RTCP property on RTP instance '0x99a1948' [Feb 20 15:23:39] DEBUG[32544][C-00000002]: res_rtp_asterisk.c:853 ast_rtp_ice_set_role: Set role to CONTROLLED (0xa5db3e8) [Feb 20 15:23:39] DEBUG[32544][C-00000002]: res_rtp_asterisk.c:856 ast_rtp_ice_set_role: Set role failed; no ice instance (0xa5db3e8) [Feb 20 15:23:39] DEBUG[32544][C-00000002]: acl.c:957 ast_ouraddrfor: For destination '192.168.41.114', our source address is '192.168.41.78'. [Feb 20 15:23:39] DEBUG[32581]: res_timing_timerfd.c:167 timerfd_timer_ack: Expected to acknowledge 1 ticks but got 2 instead [Feb 20 15:23:39] DEBUG[32582]: res_timing_timerfd.c:167 timerfd_timer_ack: Expected to acknowledge 1 ticks but got 3 instead [Feb 20 15:23:39] DEBUG[32580][C-00000001]: res_rtp_asterisk.c:3393 ast_rtp_raw_write: Difference is 2544, ms is 73 [Feb 20 15:23:39] DEBUG[32544][C-00000002]: res_rtp_asterisk.c:5036 ast_rtp_remote_address_set: Setting RTCP address on RTP instance '0xa5db3e8' Peer video RTP is at port 192.168.41.114:9078 [Feb 20 15:23:39] DEBUG[32544][C-00000002]: rtp_engine.c:658 ast_rtp_codecs_payloads_copy: Copying payload 96 (0xbc78060) from 0x47403010 to 0xa5db5b0 [Feb 20 15:23:39] DEBUG[32544][C-00000002]: chan_sip.c:11022 process_sdp: We're settling with these formats: (opus|vp8) [Feb 20 15:23:39] DEBUG[32582]: res_timing_timerfd.c:167 timerfd_timer_ack: Expected to acknowledge 1 ticks but got 3 instead [Feb 20 15:23:39] DEBUG[32544][C-00000002]: chan_sip.c:26236 handle_request_invite: Checking SIP call limits for device 50002 [Feb 20 15:23:39] DEBUG[32544][C-00000002]: chan_sip.c:6767 update_call_counter: Updating call counter for incoming call [Feb 20 15:23:39] DEBUG[32544][C-00000002]: netsock2.c:172 ast_sockaddr_split_hostport: Splitting '192.168.1.227' into... [Feb 20 15:23:39] DEBUG[32544][C-00000002]: netsock2.c:226 ast_sockaddr_split_hostport: ...host '192.168.1.227' and port ''. [Feb 20 15:23:39] DEBUG[32544][C-00000002]: netsock2.c:172 ast_sockaddr_split_hostport: Splitting '192.168.1.227' into... [Feb 20 15:23:39] DEBUG[32580][C-00000001]: res_rtp_asterisk.c:3393 ast_rtp_raw_write: Difference is 1200, ms is 45 [Feb 20 15:23:39] DEBUG[32544][C-00000002]: netsock2.c:226 ast_sockaddr_split_hostport: ...host '192.168.1.227' and port ''. Looking for 350001 in default (domain 192.168.1.227) [Feb 20 15:23:39] DEBUG[32544][C-00000002]: pbx_lua.c:1480 lua_find_extension: Looking up 350001@default:1 [Feb 20 15:23:39] DEBUG[32582]: res_timing_timerfd.c:167 timerfd_timer_ack: Expected to acknowledge 1 ticks but got 4 instead [Feb 20 15:23:39] DEBUG[32544][C-00000002]: pbx_lua.c:1480 lua_find_extension: Looking up 350001@demo:1 [Feb 20 15:23:39] DEBUG[32580][C-00000001]: res_rtp_asterisk.c:3393 ast_rtp_raw_write: Difference is 4080, ms is 105 [Feb 20 15:23:39] DEBUG[32582]: res_timing_timerfd.c:167 timerfd_timer_ack: Expected to acknowledge 1 ticks but got 4 instead [Feb 20 15:23:39] DEBUG[32580][C-00000001]: res_rtp_asterisk.c:3393 ast_rtp_raw_write: Difference is 2544, ms is 73 [Feb 20 15:23:39] DEBUG[32582]: res_timing_timerfd.c:167 timerfd_timer_ack: Expected to acknowledge 1 ticks but got 2 instead [Feb 20 15:23:39] DEBUG[32582]: res_timing_timerfd.c:167 timerfd_timer_ack: Expected to acknowledge 1 ticks but got 2 instead [Feb 20 15:23:39] DEBUG[32580][C-00000001]: res_rtp_asterisk.c:3393 ast_rtp_raw_write: Difference is 1872, ms is 59 [Feb 20 15:23:39] DEBUG[32582]: res_timing_timerfd.c:167 timerfd_timer_ack: Expected to acknowledge 1 ticks but got 3 instead [Feb 20 15:23:39] DEBUG[32544][C-00000002]: chan_sip.c:8153 sip_new: *** Our native formats are (vp8|opus) [Feb 20 15:23:39] DEBUG[32580][C-00000001]: res_rtp_asterisk.c:3393 ast_rtp_raw_write: Difference is 2880, ms is 80 [Feb 20 15:23:39] DEBUG[32544][C-00000002]: chan_sip.c:8154 sip_new: *** Joint capabilities are (opus|vp8) [Feb 20 15:23:39] DEBUG[32544][C-00000002]: chan_sip.c:8155 sip_new: *** Our capabilities are (opus|vp8) [Feb 20 15:23:39] DEBUG[32544][C-00000002]: chan_sip.c:8156 sip_new: *** Prefcaps capabilities are (nothing) [Feb 20 15:23:39] DEBUG[32544][C-00000002]: chan_sip.c:8157 sip_new: *** AST_CODEC_CHOOSE formats are opus [Feb 20 15:23:39] DEBUG[32544][C-00000002]: chan_sip.c:8169 sip_new: *** ************* SIP_PAGE2_VIDEOSUPPORT_ALWAYS *********** [Feb 20 15:23:39] DEBUG[32582]: res_timing_timerfd.c:167 timerfd_timer_ack: Expected to acknowledge 1 ticks but got 2 instead [Feb 20 15:23:39] DEBUG[32544][C-00000002]: chan_sip.c:8197 sip_new: This channel can handle video! HOLLYWOOD next! [Feb 20 15:23:39] DEBUG[32582]: res_timing_timerfd.c:167 timerfd_timer_ack: Expected to acknowledge 1 ticks but got 3 instead [Feb 20 15:23:39] DEBUG[32544][C-00000002]: pbx_lua.c:1480 lua_find_extension: Looking up 350001@default:1 [Feb 20 15:23:39] DEBUG[32544][C-00000002]: pbx_lua.c:1480 lua_find_extension: Looking up 350001@demo:1 [Feb 20 15:23:39] DEBUG[32580][C-00000001]: res_rtp_asterisk.c:3393 ast_rtp_raw_write: Difference is 2304, ms is 68 [Feb 20 15:23:39] DEBUG[32582]: res_timing_timerfd.c:167 timerfd_timer_ack: Expected to acknowledge 1 ticks but got 3 instead [Feb 20 15:23:39] DEBUG[32580][C-00000001]: res_rtp_asterisk.c:3393 ast_rtp_raw_write: Difference is 3504, ms is 93 [Feb 20 15:23:39] DEBUG[32582]: res_timing_timerfd.c:167 timerfd_timer_ack: Expected to acknowledge 1 ticks but got 3 instead [Feb 20 15:23:39] DEBUG[32582]: channel.c:3469 ast_settimeout_full: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Feb 20 15:23:39] DEBUG[32582]: channel.c:3469 ast_settimeout_full: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Feb 20 15:23:39] DEBUG[32580][C-00000001]: res_rtp_asterisk.c:3393 ast_rtp_raw_write: Difference is 1872, ms is 59 [Feb 20 15:23:39] DEBUG[32582]: channel.c:3469 ast_settimeout_full: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Feb 20 15:23:39] DEBUG[32582]: channel.c:5680 set_format: Channel CBAnn/350001-00000001;1 setting write format path: slin -> slin sip_route_dump: route/path hop: [Feb 20 15:23:39] DEBUG[32544][C-00000002]: chan_sip.c:26440 handle_request_invite: SIP/50002-00000002: New call is still down.... Trying... [Feb 20 15:23:39] DEBUG[32580][C-00000001]: res_rtp_asterisk.c:3393 ast_rtp_raw_write: Difference is 2928, ms is 81 <--- Transmitting (no NAT) to 192.168.1.147:23681 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.1.147:23681;branch=z9hG4bK.tvdYczhsY;received=192.168.1.147;rport=23681 From: ;tag=X3ughL6Dw To: sip:350001@192.168.1.227 Call-ID: AOK5~O9uyp CSeq: 21 INVITE Server: Asterisk PBX 13.14.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Contact: Content-Length: 0 <------------> [Feb 20 15:23:39] DEBUG[32544][C-00000002]: chan_sip.c:3753 __sip_xmit: Trying to put 'SIP/2.0 100' onto UDP socket destined for 192.168.1.147:23681 [Feb 20 15:23:39] DEBUG[32580][C-00000001]: res_rtp_asterisk.c:3393 ast_rtp_raw_write: Difference is 2592, ms is 74 [Feb 20 15:23:39] DEBUG[32495]: devicestate.c:369 _ast_device_state: No provider found, checking channel drivers for SIP - 50002 [Feb 20 15:23:39] DEBUG[32495]: chan_sip.c:30178 sip_devicestate: Checking device state for peer 50002 [Feb 20 15:23:39] DEBUG[32495]: devicestate.c:474 do_state_change: Changing state for SIP/50002 - state 1 (Not in use) [Feb 20 15:23:39] DEBUG[32566]: app_queue.c:2482 device_state_cb: Device 'SIP/50002' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Feb 20 15:23:39] DEBUG[32584][C-00000002]: pbx_lua.c:1480 lua_find_extension: Looking up 350001@default:1 <--- SIP read from UDP:192.168.1.147:23681 ---> INVITE sip:350001@192.168.1.227 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.147:23681;branch=z9hG4bK.tvdYczhsY;rport From: ;tag=X3ughL6Dw To: sip:350001@192.168.1.227 CSeq: 21 INVITE Call-ID: AOK5~O9uyp Max-Forwards: 70 Supported: replaces, outbound Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO, UPDATE Content-Type: application/sdp Content-Length: 466 Contact: ;+sip.instance="" User-Agent: Linphone/3.10.2 (belle-sip/1.5.0) Authorization: Digest realm="asterisk", nonce="2d0b2d61", algorithm=MD5, username="50002", uri="sip:350001@192.168.1.227", response="fdf077e97670bda2fdd4119dbaef0eb6" v=0 o=50002 12 2550 IN IP4 192.168.41.114 s=Talk c=IN IP4 192.168.41.114 t=0 0 a=rtcp-xr:rcvr-rtt=all:10000 stat-summary=loss,dup,jitt,TTL voip-metrics m=audio 7078 RTP/AVP 96 101 a=rtpmap:96 opus/48000/2 a=fmtp:96 useinbandfec=1 a=rtpmap:101 telephone-event/48000 a=rtcp-fb:* trr-int 5000 m=video 9078 RTP/AVP 96 a=rtpmap:96 VP8/90000 a=rtcp-fb:* trr-int 5000 a=rtcp-fb:96 nack pli a=rtcp-fb:96 nack sli a=rtcp-fb:96 ack rpsi a=rtcp-fb:96 ccm fir <-------------> [Feb 20 15:23:39] DEBUG[32544]: chan_sip.c:9893 parse_request: Header 0 [ 40]: INVITE sip:350001@192.168.1.227 SIP/2.0 [Feb 20 15:23:39] DEBUG[32580][C-00000001]: res_rtp_asterisk.c:3393 ast_rtp_raw_write: Difference is 3504, ms is 93 [Feb 20 15:23:39] DEBUG[32584][C-00000002]: pbx_lua.c:1480 lua_find_extension: Looking up 350001@demo:1 [Feb 20 15:23:39] DEBUG[32584][C-00000002]: pbx.c:2875 pbx_extension_helper: Launching 'Playback' [Feb 20 15:23:39] DEBUG[32544]: chan_sip.c:9893 parse_request: Header 1 [ 69]: Via: SIP/2.0/UDP 192.168.1.147:23681;branch=z9hG4bK.tvdYczhsY;rport [Feb 20 15:23:39] DEBUG[32544]: chan_sip.c:9893 parse_request: Header 2 [ 46]: From: ;tag=X3ughL6Dw [Feb 20 15:23:39] DEBUG[32544]: chan_sip.c:9893 parse_request: Header 3 [ 29]: To: sip:350001@192.168.1.227 [Feb 20 15:23:39] DEBUG[32544]: chan_sip.c:9893 parse_request: Header 4 [ 15]: CSeq: 21 INVITE [Feb 20 15:23:39] DEBUG[32544]: chan_sip.c:9893 parse_request: Header 5 [ 19]: Call-ID: AOK5~O9uyp [Feb 20 15:23:39] DEBUG[32544]: chan_sip.c:9893 parse_request: Header 6 [ 16]: Max-Forwards: 70 [Feb 20 15:23:39] DEBUG[32544]: chan_sip.c:9893 parse_request: Header 7 [ 29]: Supported: replaces, outbound [Feb 20 15:23:39] DEBUG[32544]: chan_sip.c:9893 parse_request: Header 8 [ 89]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO, UPDATE [Feb 20 15:23:39] DEBUG[32544]: chan_sip.c:9893 parse_request: Header 9 [ 29]: Content-Type: application/sdp [Feb 20 15:23:39] DEBUG[32544]: chan_sip.c:9893 parse_request: Header 10 [ 19]: Content-Length: 466 [Feb 20 15:23:39] DEBUG[32544]: chan_sip.c:9893 parse_request: Header 11 [120]: Contact: ;+sip.instance="" -- Executing [350001@default:1] Playback("SIP/50002-00000002", "beep") in new stack [Feb 20 15:23:40] DEBUG[32581]: res_timing_timerfd.c:167 timerfd_timer_ack: Expected to acknowledge 1 ticks but got 2 instead [Feb 20 15:23:39] DEBUG[32544]: chan_sip.c:9893 parse_request: Header 12 [ 45]: User-Agent: Linphone/3.10.2 (belle-sip/1.5.0) [Feb 20 15:23:40] DEBUG[32544]: chan_sip.c:9893 parse_request: Header 13 [167]: Authorization: Digest realm="asterisk", nonce="2d0b2d61", algorithm=MD5, username="50002", uri="sip:350001@192.168.1.227", response="fdf077e97670bda2fdd4119dbaef0eb6" [Feb 20 15:23:40] DEBUG[32544]: chan_sip.c:9893 parse_request: Header 14 [ 0]: [Feb 20 15:23:40] DEBUG[32544]: chan_sip.c:9893 parse_request: Body 0 [ 3]: v=0 [Feb 20 15:23:40] DEBUG[32544]: chan_sip.c:9893 parse_request: Body 1 [ 37]: o=50002 12 2550 IN IP4 192.168.41.114 [Feb 20 15:23:40] DEBUG[32544]: chan_sip.c:9893 parse_request: Body 2 [ 6]: s=Talk [Feb 20 15:23:40] DEBUG[32544]: chan_sip.c:9893 parse_request: Body 3 [ 23]: c=IN IP4 192.168.41.114 [Feb 20 15:23:40] DEBUG[32544]: chan_sip.c:9893 parse_request: Body 4 [ 5]: t=0 0 [Feb 20 15:23:40] DEBUG[32544]: chan_sip.c:9893 parse_request: Body 5 [ 72]: a=rtcp-xr:rcvr-rtt=all:10000 stat-summary=loss,dup,jitt,TTL voip-metrics [Feb 20 15:23:40] DEBUG[32544]: chan_sip.c:9893 parse_request: Body 6 [ 27]: m=audio 7078 RTP/AVP 96 101 [Feb 20 15:23:40] DEBUG[32544]: chan_sip.c:9893 parse_request: Body 7 [ 24]: a=rtpmap:96 opus/48000/2 [Feb 20 15:23:40] DEBUG[32544]: chan_sip.c:9893 parse_request: Body 8 [ 24]: a=fmtp:96 useinbandfec=1 [Feb 20 15:23:40] DEBUG[32544]: chan_sip.c:9893 parse_request: Body 9 [ 34]: a=rtpmap:101 telephone-event/48000 [Feb 20 15:23:40] DEBUG[32544]: chan_sip.c:9893 parse_request: Body 10 [ 24]: a=rtcp-fb:* trr-int 5000 [Feb 20 15:23:40] DEBUG[32544]: chan_sip.c:9893 parse_request: Body 11 [ 23]: m=video 9078 RTP/AVP 96 [Feb 20 15:23:40] DEBUG[32544]: chan_sip.c:9893 parse_request: Body 12 [ 21]: a=rtpmap:96 VP8/90000 [Feb 20 15:23:40] DEBUG[32580][C-00000001]: res_rtp_asterisk.c:3393 ast_rtp_raw_write: Difference is 4752, ms is 119 [Feb 20 15:23:40] DEBUG[32544]: chan_sip.c:9893 parse_request: Body 13 [ 24]: a=rtcp-fb:* trr-int 5000 [Feb 20 15:23:40] DEBUG[32544]: chan_sip.c:9893 parse_request: Body 14 [ 21]: a=rtcp-fb:96 nack pli [Feb 20 15:23:40] DEBUG[32544]: chan_sip.c:9893 parse_request: Body 15 [ 21]: a=rtcp-fb:96 nack sli [Feb 20 15:23:40] DEBUG[32544]: chan_sip.c:9893 parse_request: Body 16 [ 21]: a=rtcp-fb:96 ack rpsi [Feb 20 15:23:40] DEBUG[32544]: chan_sip.c:9930 parse_request: Body 17 [ 20]: a=rtcp-fb:96 ccm fir --- (14 headers 18 lines) --- [Feb 20 15:23:40] DEBUG[32544]: chan_sip.c:9419 __find_call: = Looking for Call ID: AOK5~O9uyp (Checking From) --From tag X3ughL6Dw --To-tag [Feb 20 15:23:40] DEBUG[32580][C-00000001]: res_rtp_asterisk.c:3393 ast_rtp_raw_write: Difference is 3600, ms is 95 [Feb 20 15:23:40] DEBUG[32584][C-00000002]: chan_sip.c:7412 sip_answer: SIP answering channel: SIP/50002-00000002 [Feb 20 15:23:40] DEBUG[32495]: devicestate.c:369 _ast_device_state: No provider found, checking channel drivers for SIP - 50002 [Feb 20 15:23:40] DEBUG[32495]: chan_sip.c:30178 sip_devicestate: Checking device state for peer 50002 [Feb 20 15:23:40] DEBUG[32495]: devicestate.c:474 do_state_change: Changing state for SIP/50002 - state 1 (Not in use) [Feb 20 15:23:40] DEBUG[32584][C-00000002]: res_rtp_asterisk.c:3052 ast_rtp_update_source: Setting the marker bit due to a source update [Feb 20 15:23:40] DEBUG[32584][C-00000002]: chan_sip.c:13392 add_sdp: This call needs video offers! [Feb 20 15:23:40] DEBUG[32584][C-00000002]: chan_sip.c:13416 add_sdp: ** Our capability: (opus|vp8) Video flag: False Text flag: True [Feb 20 15:23:40] DEBUG[32584][C-00000002]: chan_sip.c:13417 add_sdp: ** Our prefcodec: (nothing) Audio is at 15176 Video is at 192.168.1.227:19328 Adding codec opus to SDP Adding video codec vp8 to SDP [Feb 20 15:23:40] DEBUG[32584][C-00000002]: chan_sip.c:13587 add_sdp: -- Done with adding codecs to SDP [Feb 20 15:23:40] DEBUG[32584][C-00000002]: chan_sip.c:13612 add_sdp: Setting framing on incoming call: 0 [Feb 20 15:23:40] DEBUG[32584][C-00000002]: chan_sip.c:13800 add_sdp: Done building SDP. Settling with this capability: (opus|vp8) [Feb 20 15:23:40] DEBUG[32580][C-00000001]: res_rtp_asterisk.c:3393 ast_rtp_raw_write: Difference is 4032, ms is 104 <--- Reliably Transmitting (no NAT) to 192.168.1.147:23681 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.147:23681;branch=z9hG4bK.tvdYczhsY;received=192.168.1.147;rport=23681 From: ;tag=X3ughL6Dw To: sip:350001@192.168.1.227;tag=as188ae036 Call-ID: AOK5~O9uyp CSeq: 21 INVITE Server: Asterisk PBX 13.14.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Contact: Content-Type: application/sdp Content-Length: 284 v=0 o=root 1207775766 1207775766 IN IP4 192.168.1.227 s=Asterisk PBX 13.14.0 c=IN IP4 192.168.1.227 b=CT:384 t=0 0 m=audio 15176 RTP/AVP 96 a=rtpmap:96 opus/48000/2 a=maxptime:20 a=sendrecv m=video 19328 RTP/AVP 96 a=rtpmap:96 VP8/90000 a=rtcp-fb:* ccm fir a=sendrecv <------------> [Feb 20 15:23:40] DEBUG[32584][C-00000002]: chan_sip.c:4266 __sip_reliable_xmit: *** SIP TIMER: Initializing retransmit timer on packet: Id #3 [Feb 20 15:23:40] DEBUG[32584][C-00000002]: chan_sip.c:3753 __sip_xmit: Trying to put 'SIP/2.0 200' onto UDP socket destined for 192.168.1.147:23681 [Feb 20 15:23:40] DEBUG[32544]: netsock2.c:172 ast_sockaddr_split_hostport: Splitting '192.168.1.227' into... [Feb 20 15:23:40] DEBUG[32544]: netsock2.c:226 ast_sockaddr_split_hostport: ...host '192.168.1.227' and port ''. [Feb 20 15:23:40] DEBUG[32544]: netsock2.c:172 ast_sockaddr_split_hostport: Splitting '192.168.1.227' into... [Feb 20 15:23:40] DEBUG[32544]: netsock2.c:226 ast_sockaddr_split_hostport: ...host '192.168.1.227' and port ''. [Feb 20 15:23:40] DEBUG[32580][C-00000001]: res_rtp_asterisk.c:3393 ast_rtp_raw_write: Difference is 3168, ms is 86 [Feb 20 15:23:40] DEBUG[32544][C-00000002]: chan_sip.c:28652 handle_incoming: **** Received INVITE (5) - Command in SIP INVITE [Feb 20 15:23:40] DEBUG[32544][C-00000002]: chan_sip.c:28677 handle_incoming: Ignoring SIP message because of retransmit (INVITE Seqno 21, ours 21) Ignoring this INVITE request [Feb 20 15:23:40] DEBUG[32544][C-00000002]: chan_sip.c:26344 handle_request_invite: Got a SIP re-transmit of INVITE for call AOK5~O9uyp [Feb 20 15:23:40] DEBUG[32544][C-00000002]: chan_sip.c:26506 handle_request_invite: SIP/50002-00000002: This call is UP.... [Feb 20 15:23:40] DEBUG[32580][C-00000001]: res_rtp_asterisk.c:3393 ast_rtp_raw_write: Difference is 1968, ms is 61 <--- Transmitting (no NAT) to 192.168.1.147:23681 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.1.147:23681;branch=z9hG4bK.tvdYczhsY;received=192.168.1.147;rport=23681 From: ;tag=X3ughL6Dw To: sip:350001@192.168.1.227 Call-ID: AOK5~O9uyp CSeq: 21 INVITE Server: Asterisk PBX 13.14.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Contact: Content-Length: 0 <------------> [Feb 20 15:23:40] DEBUG[32544][C-00000002]: chan_sip.c:3753 __sip_xmit: Trying to put 'SIP/2.0 100' onto UDP socket destined for 192.168.1.147:23681 [Feb 20 15:23:40] DEBUG[32544][C-00000002]: chan_sip.c:13392 add_sdp: This call needs video offers! [Feb 20 15:23:40] DEBUG[32544][C-00000002]: chan_sip.c:13416 add_sdp: ** Our capability: (opus|vp8) Video flag: False Text flag: True [Feb 20 15:23:40] DEBUG[32544][C-00000002]: chan_sip.c:13417 add_sdp: ** Our prefcodec: (nothing) Audio is at 15176 Video is at 192.168.1.227:19328 Adding codec opus to SDP Adding video codec vp8 to SDP [Feb 20 15:23:40] DEBUG[32544][C-00000002]: chan_sip.c:13587 add_sdp: -- Done with adding codecs to SDP [Feb 20 15:23:40] DEBUG[32544][C-00000002]: chan_sip.c:13612 add_sdp: Setting framing on incoming call: 0 [Feb 20 15:23:40] DEBUG[32544][C-00000002]: chan_sip.c:13800 add_sdp: Done building SDP. Settling with this capability: (opus|vp8) [Feb 20 15:23:40] DEBUG[32580][C-00000001]: res_rtp_asterisk.c:3393 ast_rtp_raw_write: Difference is 2496, ms is 72 <--- Transmitting (no NAT) to 192.168.1.147:23681 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.147:23681;branch=z9hG4bK.tvdYczhsY;received=192.168.1.147;rport=23681 From: ;tag=X3ughL6Dw To: sip:350001@192.168.1.227;tag=as188ae036 Call-ID: AOK5~O9uyp CSeq: 21 INVITE Server: Asterisk PBX 13.14.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Contact: Content-Type: application/sdp Content-Length: 284 v=0 o=root 1207775766 1207775767 IN IP4 192.168.1.227 s=Asterisk PBX 13.14.0 c=IN IP4 192.168.1.227 b=CT:384 t=0 0 m=audio 15176 RTP/AVP 96 a=rtpmap:96 opus/48000/2 a=maxptime:20 a=sendrecv m=video 19328 RTP/AVP 96 a=rtpmap:96 VP8/90000 a=rtcp-fb:* ccm fir a=sendrecv <------------> [Feb 20 15:23:40] DEBUG[32544][C-00000002]: chan_sip.c:3753 __sip_xmit: Trying to put 'SIP/2.0 200' onto UDP socket destined for 192.168.1.147:23681 <--- SIP read from UDP:192.168.1.147:23681 ---> INVITE sip:350001@192.168.1.227 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.147:23681;branch=z9hG4bK.tvdYczhsY;rport From: ;tag=X3ughL6Dw To: sip:350001@192.168.1.227 CSeq: 21 INVITE Call-ID: AOK5~O9uyp Max-Forwards: 70 Supported: replaces, outbound Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO, UPDATE Content-Type: application/sdp Content-Length: 466 Contact: ;+sip.instance="" User-Agent: Linphone/3.10.2 (belle-sip/1.5.0) Authorization: Digest realm="asterisk", nonce="2d0b2d61", algorithm=MD5, username="50002", uri="sip:350001@192.168.1.227", response="fdf077e97670bda2fdd4119dbaef0eb6" v=0 o=50002 12 2550 IN IP4 192.168.41.114 s=Talk c=IN IP4 192.168.41.114 t=0 0 a=rtcp-xr:rcvr-rtt=all:10000 stat-summary=loss,dup,jitt,TTL voip-metrics m=audio 7078 RTP/AVP 96 101 a=rtpmap:96 opus/48000/2 a=fmtp:96 useinbandfec=1 a=rtpmap:101 telephone-event/48000 a=rtcp-fb:* trr-int 5000 m=video 9078 RTP/AVP 96 a=rtpmap:96 VP8/90000 a=rtcp-fb:* trr-int 5000 a=rtcp-fb:96 nack pli a=rtcp-fb:96 nack sli a=rtcp-fb:96 ack rpsi a=rtcp-fb:96 ccm fir <-------------> [Feb 20 15:23:40] DEBUG[32544]: chan_sip.c:9893 parse_request: Header 0 [ 40]: INVITE sip:350001@192.168.1.227 SIP/2.0 [Feb 20 15:23:40] DEBUG[32544]: chan_sip.c:9893 parse_request: Header 1 [ 69]: Via: SIP/2.0/UDP 192.168.1.147:23681;branch=z9hG4bK.tvdYczhsY;rport [Feb 20 15:23:40] DEBUG[32544]: chan_sip.c:9893 parse_request: Header 2 [ 46]: From: ;tag=X3ughL6Dw [Feb 20 15:23:40] DEBUG[32580][C-00000001]: res_rtp_asterisk.c:3393 ast_rtp_raw_write: Difference is 2400, ms is 70 [Feb 20 15:23:40] DEBUG[32544]: chan_sip.c:9893 parse_request: Header 3 [ 29]: To: sip:350001@192.168.1.227 [Feb 20 15:23:40] DEBUG[32544]: chan_sip.c:9893 parse_request: Header 4 [ 15]: CSeq: 21 INVITE [Feb 20 15:23:40] DEBUG[32544]: chan_sip.c:9893 parse_request: Header 5 [ 19]: Call-ID: AOK5~O9uyp [Feb 20 15:23:40] DEBUG[32544]: chan_sip.c:9893 parse_request: Header 6 [ 16]: Max-Forwards: 70 [Feb 20 15:23:40] DEBUG[32544]: chan_sip.c:9893 parse_request: Header 7 [ 29]: Supported: replaces, outbound [Feb 20 15:23:40] DEBUG[32544]: chan_sip.c:9893 parse_request: Header 8 [ 89]: Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY, MESSAGE, SUBSCRIBE, INFO, UPDATE [Feb 20 15:23:40] DEBUG[32544]: chan_sip.c:9893 parse_request: Header 9 [ 29]: Content-Type: application/sdp [Feb 20 15:23:40] DEBUG[32544]: chan_sip.c:9893 parse_request: Header 10 [ 19]: Content-Length: 466 [Feb 20 15:23:40] DEBUG[32544]: chan_sip.c:9893 parse_request: Header 11 [120]: Contact: ;+sip.instance="" [Feb 20 15:23:40] DEBUG[32544]: chan_sip.c:9893 parse_request: Header 12 [ 45]: User-Agent: Linphone/3.10.2 (belle-sip/1.5.0) [Feb 20 15:23:40] DEBUG[32544]: chan_sip.c:9893 parse_request: Header 13 [167]: Authorization: Digest realm="asterisk", nonce="2d0b2d61", algorithm=MD5, username="50002", uri="sip:350001@192.168.1.227", response="fdf077e97670bda2fdd4119dbaef0eb6" [Feb 20 15:23:40] DEBUG[32544]: chan_sip.c:9893 parse_request: Header 14 [ 0]: [Feb 20 15:23:40] DEBUG[32544]: chan_sip.c:9893 parse_request: Body 0 [ 3]: v=0 [Feb 20 15:23:40] DEBUG[32544]: chan_sip.c:9893 parse_request: Body 1 [ 37]: o=50002 12 2550 IN IP4 192.168.41.114 [Feb 20 15:23:40] DEBUG[32544]: chan_sip.c:9893 parse_request: Body 2 [ 6]: s=Talk [Feb 20 15:23:40] DEBUG[32544]: chan_sip.c:9893 parse_request: Body 3 [ 23]: c=IN IP4 192.168.41.114 [Feb 20 15:23:40] DEBUG[32544]: chan_sip.c:9893 parse_request: Body 4 [ 5]: t=0 0 [Feb 20 15:23:40] DEBUG[32544]: chan_sip.c:9893 parse_request: Body 5 [ 72]: a=rtcp-xr:rcvr-rtt=all:10000 stat-summary=loss,dup,jitt,TTL voip-metrics [Feb 20 15:23:40] DEBUG[32544]: chan_sip.c:9893 parse_request: Body 6 [ 27]: m=audio 7078 RTP/AVP 96 101 [Feb 20 15:23:40] DEBUG[32544]: chan_sip.c:9893 parse_request: Body 7 [ 24]: a=rtpmap:96 opus/48000/2 [Feb 20 15:23:40] DEBUG[32580][C-00000001]: res_rtp_asterisk.c:3393 ast_rtp_raw_write: Difference is 912, ms is 39 [Feb 20 15:23:40] DEBUG[32544]: chan_sip.c:9893 parse_request: Body 8 [ 24]: a=fmtp:96 useinbandfec=1 [Feb 20 15:23:40] DEBUG[32544]: chan_sip.c:9893 parse_request: Body 9 [ 34]: a=rtpmap:101 telephone-event/48000 [Feb 20 15:23:40] DEBUG[32544]: chan_sip.c:9893 parse_request: Body 10 [ 24]: a=rtcp-fb:* trr-int 5000 [Feb 20 15:23:40] DEBUG[32544]: chan_sip.c:9893 parse_request: Body 11 [ 23]: m=video 9078 RTP/AVP 96 [Feb 20 15:23:40] DEBUG[32544]: chan_sip.c:9893 parse_request: Body 12 [ 21]: a=rtpmap:96 VP8/90000 [Feb 20 15:23:40] DEBUG[32584][C-00000002]: res_rtp_asterisk.c:4564 ast_rtp_read: 0x9341130 -- Probation learning mode pass with source address 192.168.1.147:14497 [Feb 20 15:23:40] DEBUG[32544]: chan_sip.c:9893 parse_request: Body 13 [ 24]: a=rtcp-fb:* trr-int 5000 [Feb 20 15:23:40] DEBUG[32544]: chan_sip.c:9893 parse_request: Body 14 [ 21]: a=rtcp-fb:96 nack pli [Feb 20 15:23:40] DEBUG[32544]: chan_sip.c:9893 parse_request: Body 15 [ 21]: a=rtcp-fb:96 nack sli [Feb 20 15:23:40] DEBUG[32544]: chan_sip.c:9893 parse_request: Body 16 [ 21]: a=rtcp-fb:96 ack rpsi > 0x9341130 -- Probation passed - setting RTP source address to 192.168.1.147:14497 [Feb 20 15:23:40] DEBUG[32544]: chan_sip.c:9930 parse_request: Body 17 [ 20]: a=rtcp-fb:96 ccm fir --- (14 headers 18 lines) --- [Feb 20 15:23:40] DEBUG[32544]: chan_sip.c:9419 __find_call: = Looking for Call ID: AOK5~O9uyp (Checking From) --From tag X3ughL6Dw --To-tag [Feb 20 15:23:40] DEBUG[32584][C-00000002]: acl.c:957 ast_ouraddrfor: For destination '192.168.1.147', our source address is '192.168.41.78'. [Feb 20 15:23:40] DEBUG[32584][C-00000002]: res_rtp_asterisk.c:5036 ast_rtp_remote_address_set: Setting RTCP address on RTP instance '0x99a1948' [Feb 20 15:23:40] DEBUG[32544]: netsock2.c:172 ast_sockaddr_split_hostport: Splitting '192.168.1.227' into... [Feb 20 15:23:40] DEBUG[32544]: netsock2.c:226 ast_sockaddr_split_hostport: ...host '192.168.1.227' and port ''. [Feb 20 15:23:40] DEBUG[32544]: netsock2.c:172 ast_sockaddr_split_hostport: Splitting '192.168.1.227' into... [Feb 20 15:23:40] DEBUG[32544]: netsock2.c:226 ast_sockaddr_split_hostport: ...host '192.168.1.227' and port ''. [Feb 20 15:23:40] DEBUG[32580][C-00000001]: res_rtp_asterisk.c:3393 ast_rtp_raw_write: Difference is 3648, ms is 96 [Feb 20 15:23:40] DEBUG[32584][C-00000002]: channel.c:5680 set_format: Channel SIP/50002-00000002 setting write format path: gsm -> opus [Feb 20 15:23:40] DEBUG[32544][C-00000002]: chan_sip.c:28652 handle_incoming: **** Received INVITE (5) - Command in SIP INVITE [Feb 20 15:23:40] DEBUG[32544][C-00000002]: chan_sip.c:28677 handle_incoming: Ignoring SIP message because of retransmit (INVITE Seqno 21, ours 21) Ignoring this INVITE request [Feb 20 15:23:40] DEBUG[32580][C-00000001]: res_rtp_asterisk.c:3393 ast_rtp_raw_write: Difference is 2928, ms is 81 [Feb 20 15:23:40] DEBUG[32544][C-00000002]: chan_sip.c:26344 handle_request_invite: Got a SIP re-transmit of INVITE for call AOK5~O9uyp [Feb 20 15:23:40] DEBUG[32544][C-00000002]: chan_sip.c:26506 handle_request_invite: SIP/50002-00000002: This call is UP.... <--- Transmitting (no NAT) to 192.168.1.147:23681 ---> SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.1.147:23681;branch=z9hG4bK.tvdYczhsY;received=192.168.1.147;rport=23681 From: ;tag=X3ughL6Dw To: sip:350001@192.168.1.227 Call-ID: AOK5~O9uyp CSeq: 21 INVITE Server: Asterisk PBX 13.14.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Contact: Content-Length: 0 <------------> [Feb 20 15:23:40] DEBUG[32544][C-00000002]: chan_sip.c:3753 __sip_xmit: Trying to put 'SIP/2.0 100' onto UDP socket destined for 192.168.1.147:23681 [Feb 20 15:23:40] DEBUG[32544][C-00000002]: chan_sip.c:13392 add_sdp: This call needs video offers! [Feb 20 15:23:40] DEBUG[32544][C-00000002]: chan_sip.c:13416 add_sdp: ** Our capability: (opus|vp8) Video flag: False Text flag: True [Feb 20 15:23:40] DEBUG[32544][C-00000002]: chan_sip.c:13417 add_sdp: ** Our prefcodec: (nothing) Audio is at 15176 Video is at 192.168.1.227:19328 Adding codec opus to SDP Adding video codec vp8 to SDP [Feb 20 15:23:40] DEBUG[32580][C-00000001]: res_rtp_asterisk.c:3393 ast_rtp_raw_write: Difference is 3744, ms is 98 [Feb 20 15:23:40] DEBUG[32544][C-00000002]: chan_sip.c:13587 add_sdp: -- Done with adding codecs to SDP [Feb 20 15:23:40] DEBUG[32544][C-00000002]: chan_sip.c:13612 add_sdp: Setting framing on incoming call: 0 [Feb 20 15:23:40] DEBUG[32544][C-00000002]: chan_sip.c:13800 add_sdp: Done building SDP. Settling with this capability: (opus|vp8) <--- Transmitting (no NAT) to 192.168.1.147:23681 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.147:23681;branch=z9hG4bK.tvdYczhsY;received=192.168.1.147;rport=23681 From: ;tag=X3ughL6Dw To: sip:350001@192.168.1.227;tag=as188ae036 Call-ID: AOK5~O9uyp CSeq: 21 INVITE Server: Asterisk PBX 13.14.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Contact: Content-Type: application/sdp Content-Length: 284 v=0 o=root 1207775766 1207775768 IN IP4 192.168.1.227 s=Asterisk PBX 13.14.0 c=IN IP4 192.168.1.227 b=CT:384 t=0 0 m=audio 15176 RTP/AVP 96 a=rtpmap:96 opus/48000/2 a=maxptime:20 a=sendrecv m=video 19328 RTP/AVP 96 a=rtpmap:96 VP8/90000 a=rtcp-fb:* ccm fir a=sendrecv <------------> [Feb 20 15:23:40] DEBUG[32544][C-00000002]: chan_sip.c:3753 __sip_xmit: Trying to put 'SIP/2.0 200' onto UDP socket destined for 192.168.1.147:23681 [Feb 20 15:23:40] DEBUG[32580][C-00000001]: res_rtp_asterisk.c:3393 ast_rtp_raw_write: Difference is 2880, ms is 80 [Feb 20 15:23:40] DEBUG[32544]: chan_sip.c:4007 retrans_pkt: SIP TIMER: Rescheduling retransmission #3 (1) SIP/2.0 - 1 [Feb 20 15:23:40] DEBUG[32544]: chan_sip.c:4034 retrans_pkt: ** SIP timers: Rescheduling retransmission 2 to 1000 ms (t1 500 ms (Retrans id #3)) Retransmitting #1 (no NAT) to 192.168.1.147:23681: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.147:23681;branch=z9hG4bK.tvdYczhsY;received=192.168.1.147;rport=23681 From: ;tag=X3ughL6Dw To: sip:350001@192.168.1.227;tag=as188ae036 Call-ID: AOK5~O9uyp CSeq: 21 INVITE Server: Asterisk PBX 13.14.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Contact: Content-Type: application/sdp Content-Length: 284 v=0 o=root 1207775766 1207775766 IN IP4 192.168.1.227 s=Asterisk PBX 13.14.0 c=IN IP4 192.168.1.227 b=CT:384 t=0 0 m=audio 15176 RTP/AVP 96 a=rtpmap:96 opus/48000/2 a=maxptime:20 a=sendrecv m=video 19328 RTP/AVP 96 a=rtpmap:96 VP8/90000 a=rtcp-fb:* ccm fir a=sendrecv --- [Feb 20 15:23:40] DEBUG[32544]: chan_sip.c:3753 __sip_xmit: Trying to put 'SIP/2.0 200' onto UDP socket destined for 192.168.1.147:23681 [Feb 20 15:23:41] DEBUG[32581]: res_timing_timerfd.c:167 timerfd_timer_ack: Expected to acknowledge 1 ticks but got 2 instead <--- SIP read from UDP:192.168.1.147:23681 ---> ACK sip:350001@192.168.1.227:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.147:23681;rport;branch=z9hG4bK.SGNbikXP4 From: ;tag=X3ughL6Dw To: ;tag=as188ae036 CSeq: 21 ACK Call-ID: AOK5~O9uyp Max-Forwards: 70 Authorization: Digest realm="asterisk", nonce="2d0b2d61", algorithm=MD5, username="50002", uri="sip:350001@192.168.1.227", response="fdf077e97670bda2fdd4119dbaef0eb6" User-Agent: Linphone/3.10.2 (belle-sip/1.5.0) <-------------> [Feb 20 15:23:41] DEBUG[32544]: chan_sip.c:9893 parse_request: Header 0 [ 42]: ACK sip:350001@192.168.1.227:5060 SIP/2.0 [Feb 20 15:23:41] DEBUG[32544]: chan_sip.c:9893 parse_request: Header 1 [ 69]: Via: SIP/2.0/UDP 192.168.1.147:23681;rport;branch=z9hG4bK.SGNbikXP4 [Feb 20 15:23:41] DEBUG[32544]: chan_sip.c:9893 parse_request: Header 2 [ 46]: From: ;tag=X3ughL6Dw [Feb 20 15:23:41] DEBUG[32544]: chan_sip.c:9893 parse_request: Header 3 [ 46]: To: ;tag=as188ae036 [Feb 20 15:23:41] DEBUG[32584][C-00000002]: res_rtp_asterisk.c:3625 ast_rtp_write: Ooh, format changed from none to opus [Feb 20 15:23:41] DEBUG[32544]: chan_sip.c:9893 parse_request: Header 4 [ 12]: CSeq: 21 ACK [Feb 20 15:23:41] DEBUG[32544]: chan_sip.c:9893 parse_request: Header 5 [ 19]: Call-ID: AOK5~O9uyp [Feb 20 15:23:41] DEBUG[32544]: chan_sip.c:9893 parse_request: Header 6 [ 16]: Max-Forwards: 70 [Feb 20 15:23:41] DEBUG[32544]: chan_sip.c:9893 parse_request: Header 7 [167]: Authorization: Digest realm="asterisk", nonce="2d0b2d61", algorithm=MD5, username="50002", uri="sip:350001@192.168.1.227", response="fdf077e97670bda2fdd4119dbaef0eb6" [Feb 20 15:23:41] DEBUG[32544]: chan_sip.c:9893 parse_request: Header 8 [ 45]: User-Agent: Linphone/3.10.2 (belle-sip/1.5.0) --- (9 headers 0 lines) --- [Feb 20 15:23:41] DEBUG[32544]: chan_sip.c:9419 __find_call: = Looking for Call ID: AOK5~O9uyp (Checking From) --From tag X3ughL6Dw --To-tag as188ae036 [Feb 20 15:23:41] DEBUG[32584][C-00000002]: channel.c:3469 ast_settimeout_full: Scheduling timer at (50 requested / 50 actual) timer ticks per second -- Playing 'beep.gsm' (language 'en') [Feb 20 15:23:41] DEBUG[32544][C-00000002]: chan_sip.c:28652 handle_incoming: **** Received ACK (6) - Command in SIP ACK [Feb 20 15:23:41] DEBUG[32544][C-00000002]: chan_sip.c:4526 __sip_ack: ** SIP TIMER: Cancelling retransmit of packet (reply received) Retransid #3 [Feb 20 15:23:41] DEBUG[32580][C-00000001]: res_rtp_asterisk.c:3393 ast_rtp_raw_write: Difference is 5136, ms is 127 [Feb 20 15:23:41] DEBUG[32544][C-00000002]: chan_sip.c:4537 __sip_ack: Stopping retransmission on 'AOK5~O9uyp' of Response 21: Match Found [Feb 20 15:23:41] DEBUG[32584][C-00000002]: res_timing_timerfd.c:167 timerfd_timer_ack: Expected to acknowledge 1 ticks but got 3 instead [Feb 20 15:23:41] DEBUG[32580][C-00000001]: res_rtp_asterisk.c:3393 ast_rtp_raw_write: Difference is 4416, ms is 112 [Feb 20 15:23:41] DEBUG[32584][C-00000002]: res_rtp_asterisk.c:3393 ast_rtp_raw_write: Difference is 5712, ms is 139 [Feb 20 15:23:41] DEBUG[32584][C-00000002]: res_timing_timerfd.c:167 timerfd_timer_ack: Expected to acknowledge 1 ticks but got 5 instead <--- SIP read from UDP:192.168.1.147:23681 ---> ACK sip:350001@192.168.1.227:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.147:23681;branch=z9hG4bK.SGNbikXP4;rport From: ;tag=X3ughL6Dw To: ;tag=as188ae036 CSeq: 21 ACK Call-ID: AOK5~O9uyp Max-Forwards: 70 Authorization: Digest realm="asterisk", nonce="2d0b2d61", algorithm=MD5, username="50002", uri="sip:350001@192.168.1.227", response="fdf077e97670bda2fdd4119dbaef0eb6" User-Agent: Linphone/3.10.2 (belle-sip/1.5.0) <-------------> [Feb 20 15:23:41] DEBUG[32544]: chan_sip.c:9893 parse_request: Header 0 [ 42]: ACK sip:350001@192.168.1.227:5060 SIP/2.0 [Feb 20 15:23:41] DEBUG[32544]: chan_sip.c:9893 parse_request: Header 1 [ 69]: Via: SIP/2.0/UDP 192.168.1.147:23681;branch=z9hG4bK.SGNbikXP4;rport [Feb 20 15:23:41] DEBUG[32544]: chan_sip.c:9893 parse_request: Header 2 [ 46]: From: ;tag=X3ughL6Dw [Feb 20 15:23:41] DEBUG[32544]: chan_sip.c:9893 parse_request: Header 3 [ 46]: To: ;tag=as188ae036 [Feb 20 15:23:41] DEBUG[32544]: chan_sip.c:9893 parse_request: Header 4 [ 12]: CSeq: 21 ACK [Feb 20 15:23:41] DEBUG[32544]: chan_sip.c:9893 parse_request: Header 5 [ 19]: Call-ID: AOK5~O9uyp [Feb 20 15:23:41] DEBUG[32544]: chan_sip.c:9893 parse_request: Header 6 [ 16]: Max-Forwards: 70 [Feb 20 15:23:41] DEBUG[32544]: chan_sip.c:9893 parse_request: Header 7 [167]: Authorization: Digest realm="asterisk", nonce="2d0b2d61", algorithm=MD5, username="50002", uri="sip:350001@192.168.1.227", response="fdf077e97670bda2fdd4119dbaef0eb6" [Feb 20 15:23:41] DEBUG[32544]: chan_sip.c:9893 parse_request: Header 8 [ 45]: User-Agent: Linphone/3.10.2 (belle-sip/1.5.0) --- (9 headers 0 lines) --- [Feb 20 15:23:41] DEBUG[32544]: chan_sip.c:9419 __find_call: = Looking for Call ID: AOK5~O9uyp (Checking From) --From tag X3ughL6Dw --To-tag as188ae036 [Feb 20 15:23:41] DEBUG[32580][C-00000001]: res_rtp_asterisk.c:3393 ast_rtp_raw_write: Difference is 3024, ms is 83 [Feb 20 15:23:41] DEBUG[32584][C-00000002]: res_rtp_asterisk.c:3393 ast_rtp_raw_write: Difference is 6048, ms is 146 [Feb 20 15:23:41] DEBUG[32544][C-00000002]: chan_sip.c:28652 handle_incoming: **** Received ACK (6) - Command in SIP ACK [Feb 20 15:23:41] DEBUG[32580][C-00000001]: res_rtp_asterisk.c:3393 ast_rtp_raw_write: Difference is 3936, ms is 102 [Feb 20 15:23:41] DEBUG[32584][C-00000002]: res_timing_timerfd.c:167 timerfd_timer_ack: Expected to acknowledge 1 ticks but got 7 instead [Feb 20 15:23:41] DEBUG[32542]: res_timing_timerfd.c:167 timerfd_timer_ack: Expected to acknowledge 1 ticks but got 2 instead [Feb 20 15:23:41] DEBUG[32580][C-00000001]: res_rtp_asterisk.c:3393 ast_rtp_raw_write: Difference is 3408, ms is 91 <--- SIP read from UDP:192.168.1.147:23681 ---> ACK sip:350001@192.168.1.227:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.147:23681;branch=z9hG4bK.SGNbikXP4;rport From: ;tag=X3ughL6Dw To: ;tag=as188ae036 CSeq: 21 ACK Call-ID: AOK5~O9uyp Max-Forwards: 70 Authorization: Digest realm="asterisk", nonce="2d0b2d61", algorithm=MD5, username="50002", uri="sip:350001@192.168.1.227", response="fdf077e97670bda2fdd4119dbaef0eb6" User-Agent: Linphone/3.10.2 (belle-sip/1.5.0) <-------------> [Feb 20 15:23:41] DEBUG[32544]: chan_sip.c:9893 parse_request: Header 0 [ 42]: ACK sip:350001@192.168.1.227:5060 SIP/2.0 [Feb 20 15:23:41] DEBUG[32584][C-00000002]: res_rtp_asterisk.c:3393 ast_rtp_raw_write: Difference is 5424, ms is 133 [Feb 20 15:23:41] DEBUG[32544]: chan_sip.c:9893 parse_request: Header 1 [ 69]: Via: SIP/2.0/UDP 192.168.1.147:23681;branch=z9hG4bK.SGNbikXP4;rport [Feb 20 15:23:41] DEBUG[32544]: chan_sip.c:9893 parse_request: Header 2 [ 46]: From: ;tag=X3ughL6Dw [Feb 20 15:23:41] DEBUG[32544]: chan_sip.c:9893 parse_request: Header 3 [ 46]: To: ;tag=as188ae036 [Feb 20 15:23:41] DEBUG[32544]: chan_sip.c:9893 parse_request: Header 4 [ 12]: CSeq: 21 ACK [Feb 20 15:23:41] DEBUG[32544]: chan_sip.c:9893 parse_request: Header 5 [ 19]: Call-ID: AOK5~O9uyp [Feb 20 15:23:41] DEBUG[32544]: chan_sip.c:9893 parse_request: Header 6 [ 16]: Max-Forwards: 70 [Feb 20 15:23:41] DEBUG[32544]: chan_sip.c:9893 parse_request: Header 7 [167]: Authorization: Digest realm="asterisk", nonce="2d0b2d61", algorithm=MD5, username="50002", uri="sip:350001@192.168.1.227", response="fdf077e97670bda2fdd4119dbaef0eb6" [Feb 20 15:23:41] DEBUG[32544]: chan_sip.c:9893 parse_request: Header 8 [ 45]: User-Agent: Linphone/3.10.2 (belle-sip/1.5.0) --- (9 headers 0 lines) --- [Feb 20 15:23:41] DEBUG[32544]: chan_sip.c:9419 __find_call: = Looking for Call ID: AOK5~O9uyp (Checking From) --From tag X3ughL6Dw --To-tag as188ae036 [Feb 20 15:23:41] DEBUG[32584][C-00000002]: res_timing_timerfd.c:167 timerfd_timer_ack: Expected to acknowledge 1 ticks but got 7 instead [Feb 20 15:23:41] DEBUG[32563]: netsock2.c:172 ast_sockaddr_split_hostport: Splitting '192.168.41.78:14091' into... [Feb 20 15:23:41] DEBUG[32563]: netsock2.c:226 ast_sockaddr_split_hostport: ...host '192.168.41.78' and port '14091'. [Feb 20 15:23:41] DEBUG[32563]: netsock2.c:172 ast_sockaddr_split_hostport: Splitting '192.168.1.147:14338' into... [Feb 20 15:23:41] DEBUG[32563]: netsock2.c:226 ast_sockaddr_split_hostport: ...host '192.168.1.147' and port '14338'. [Feb 20 15:23:41] DEBUG[32581]: res_timing_timerfd.c:167 timerfd_timer_ack: Expected to acknowledge 1 ticks but got 2 instead [Feb 20 15:23:41] DEBUG[32580][C-00000001]: res_rtp_asterisk.c:3393 ast_rtp_raw_write: Difference is 3504, ms is 93 [Feb 20 15:23:41] DEBUG[32584][C-00000002]: res_rtp_asterisk.c:3393 ast_rtp_raw_write: Difference is 4464, ms is 113 [Feb 20 15:23:41] DEBUG[32544][C-00000002]: chan_sip.c:28652 handle_incoming: **** Received ACK (6) - Command in SIP ACK <--- SIP read from UDP:192.168.1.147:23681 ---> ACK sip:350001@192.168.1.227:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.147:23681;branch=z9hG4bK.SGNbikXP4;rport From: ;tag=X3ughL6Dw To: ;tag=as188ae036 CSeq: 21 ACK Call-ID: AOK5~O9uyp Max-Forwards: 70 Authorization: Digest realm="asterisk", nonce="2d0b2d61", algorithm=MD5, username="50002", uri="sip:350001@192.168.1.227", response="fdf077e97670bda2fdd4119dbaef0eb6" User-Agent: Linphone/3.10.2 (belle-sip/1.5.0) <-------------> [Feb 20 15:23:41] DEBUG[32544]: chan_sip.c:9893 parse_request: Header 0 [ 42]: ACK sip:350001@192.168.1.227:5060 SIP/2.0 [Feb 20 15:23:41] DEBUG[32544]: chan_sip.c:9893 parse_request: Header 1 [ 69]: Via: SIP/2.0/UDP 192.168.1.147:23681;branch=z9hG4bK.SGNbikXP4;rport [Feb 20 15:23:41] DEBUG[32544]: chan_sip.c:9893 parse_request: Header 2 [ 46]: From: ;tag=X3ughL6Dw [Feb 20 15:23:41] DEBUG[32544]: chan_sip.c:9893 parse_request: Header 3 [ 46]: To: ;tag=as188ae036 [Feb 20 15:23:41] DEBUG[32544]: chan_sip.c:9893 parse_request: Header 4 [ 12]: CSeq: 21 ACK [Feb 20 15:23:41] DEBUG[32544]: chan_sip.c:9893 parse_request: Header 5 [ 19]: Call-ID: AOK5~O9uyp [Feb 20 15:23:41] DEBUG[32584][C-00000002]: res_timing_timerfd.c:167 timerfd_timer_ack: Expected to acknowledge 1 ticks but got 5 instead [Feb 20 15:23:41] DEBUG[32544]: chan_sip.c:9893 parse_request: Header 6 [ 16]: Max-Forwards: 70 [Feb 20 15:23:41] DEBUG[32544]: chan_sip.c:9893 parse_request: Header 7 [167]: Authorization: Digest realm="asterisk", nonce="2d0b2d61", algorithm=MD5, username="50002", uri="sip:350001@192.168.1.227", response="fdf077e97670bda2fdd4119dbaef0eb6" [Feb 20 15:23:41] DEBUG[32544]: chan_sip.c:9893 parse_request: Header 8 [ 45]: User-Agent: Linphone/3.10.2 (belle-sip/1.5.0) --- (9 headers 0 lines) --- [Feb 20 15:23:41] DEBUG[32544]: chan_sip.c:9419 __find_call: = Looking for Call ID: AOK5~O9uyp (Checking From) --From tag X3ughL6Dw --To-tag as188ae036 [Feb 20 15:23:41] DEBUG[32580][C-00000001]: res_rtp_asterisk.c:3393 ast_rtp_raw_write: Difference is 3936, ms is 102 [Feb 20 15:23:41] DEBUG[32584][C-00000002]: res_rtp_asterisk.c:3393 ast_rtp_raw_write: Difference is 5088, ms is 126 [Feb 20 15:23:41] DEBUG[32544][C-00000002]: chan_sip.c:28652 handle_incoming: **** Received ACK (6) - Command in SIP ACK <--- SIP read from UDP:192.168.1.147:23681 ---> <-------------> [Feb 20 15:23:41] DEBUG[32544]: chan_sip.c:9893 parse_request: Header 0 [ 0]: [Feb 20 15:23:41] DEBUG[32584][C-00000002]: res_timing_timerfd.c:167 timerfd_timer_ack: Expected to acknowledge 1 ticks but got 6 instead [Feb 20 15:23:41] DEBUG[32580][C-00000001]: res_rtp_asterisk.c:3393 ast_rtp_raw_write: Difference is 4368, ms is 111 [Feb 20 15:23:41] DEBUG[32584][C-00000002]: res_rtp_asterisk.c:3393 ast_rtp_raw_write: Difference is 3984, ms is 103 [Feb 20 15:23:41] DEBUG[32584][C-00000002]: res_timing_timerfd.c:167 timerfd_timer_ack: Expected to acknowledge 1 ticks but got 6 instead [Feb 20 15:23:41] DEBUG[32580][C-00000001]: res_rtp_asterisk.c:3393 ast_rtp_raw_write: Difference is 4080, ms is 105 [Feb 20 15:23:41] DEBUG[32542]: res_timing_timerfd.c:167 timerfd_timer_ack: Expected to acknowledge 1 ticks but got 4 instead [Feb 20 15:23:41] DEBUG[32581]: res_timing_timerfd.c:167 timerfd_timer_ack: Expected to acknowledge 1 ticks but got 5 instead [Feb 20 15:23:42] DEBUG[32580][C-00000001]: res_rtp_asterisk.c:3393 ast_rtp_raw_write: Difference is 6048, ms is 146 [Feb 20 15:23:42] DEBUG[32584][C-00000002]: res_rtp_asterisk.c:3393 ast_rtp_raw_write: Difference is 8160, ms is 190 [Feb 20 15:23:42] DEBUG[32584][C-00000002]: res_timing_timerfd.c:167 timerfd_timer_ack: Expected to acknowledge 1 ticks but got 10 instead [Feb 20 15:23:42] DEBUG[32580][C-00000001]: res_rtp_asterisk.c:3393 ast_rtp_raw_write: Difference is 2352, ms is 69 [Feb 20 15:23:42] DEBUG[32584][C-00000002]: res_rtp_asterisk.c:3393 ast_rtp_raw_write: Difference is 3456, ms is 92 [Feb 20 15:23:42] DEBUG[32584][C-00000002]: res_timing_timerfd.c:167 timerfd_timer_ack: Expected to acknowledge 1 ticks but got 4 instead [Feb 20 15:23:42] DEBUG[32580][C-00000001]: res_rtp_asterisk.c:3393 ast_rtp_raw_write: Difference is 3984, ms is 103 [Feb 20 15:23:42] DEBUG[32584][C-00000002]: res_rtp_asterisk.c:3393 ast_rtp_raw_write: Difference is 3792, ms is 99 [Feb 20 15:23:42] DEBUG[32584][C-00000002]: res_timing_timerfd.c:167 timerfd_timer_ack: Expected to acknowledge 1 ticks but got 5 instead [Feb 20 15:23:42] DEBUG[32580][C-00000001]: res_rtp_asterisk.c:3393 ast_rtp_raw_write: Difference is 3216, ms is 87 [Feb 20 15:23:42] DEBUG[32542]: res_timing_timerfd.c:167 timerfd_timer_ack: Expected to acknowledge 1 ticks but got 4 instead [Feb 20 15:23:42] DEBUG[32584][C-00000002]: res_rtp_asterisk.c:3393 ast_rtp_raw_write: Difference is 3072, ms is 84 [Feb 20 15:23:42] DEBUG[32581]: res_timing_timerfd.c:167 timerfd_timer_ack: Expected to acknowledge 1 ticks but got 4 instead [Feb 20 15:23:42] DEBUG[32542]: res_timing_timerfd.c:167 timerfd_timer_ack: Expected to acknowledge 1 ticks but got 2 instead [Feb 20 15:23:42] DEBUG[32581]: res_timing_timerfd.c:167 timerfd_timer_ack: Expected to acknowledge 1 ticks but got 2 instead [Feb 20 15:23:42] DEBUG[32584][C-00000002]: res_timing_timerfd.c:167 timerfd_timer_ack: Expected to acknowledge 1 ticks but got 10 instead [Feb 20 15:23:42] DEBUG[32584][C-00000002]: res_rtp_asterisk.c:3393 ast_rtp_raw_write: Difference is 13104, ms is 293 [Feb 20 15:23:42] DEBUG[32580][C-00000001]: res_rtp_asterisk.c:3393 ast_rtp_raw_write: Difference is 6720, ms is 160 [Feb 20 15:23:42] DEBUG[32581]: res_timing_timerfd.c:167 timerfd_timer_ack: Expected to acknowledge 1 ticks but got 8 instead [Feb 20 15:23:42] DEBUG[32542]: res_timing_timerfd.c:167 timerfd_timer_ack: Expected to acknowledge 1 ticks but got 6 instead [Feb 20 15:23:42] DEBUG[32584][C-00000002]: res_timing_timerfd.c:167 timerfd_timer_ack: Expected to acknowledge 1 ticks but got 9 instead [Feb 20 15:23:42] DEBUG[32542]: res_timing_timerfd.c:167 timerfd_timer_ack: Expected to acknowledge 1 ticks but got 3 instead [Feb 20 15:23:42] DEBUG[32581]: res_timing_timerfd.c:167 timerfd_timer_ack: Expected to acknowledge 1 ticks but got 2 instead [Feb 20 15:23:42] DEBUG[32581]: res_timing_timerfd.c:167 timerfd_timer_ack: Expected to acknowledge 1 ticks but got 5 instead [Feb 20 15:23:42] DEBUG[32584][C-00000002]: res_rtp_asterisk.c:3393 ast_rtp_raw_write: Difference is 7296, ms is 172 [Feb 20 15:23:42] DEBUG[32563]: netsock2.c:172 ast_sockaddr_split_hostport: Splitting '192.168.41.78:16635' into... [Feb 20 15:23:42] DEBUG[32563]: netsock2.c:226 ast_sockaddr_split_hostport: ...host '192.168.41.78' and port '16635'. [Feb 20 15:23:42] DEBUG[32580][C-00000001]: res_rtp_asterisk.c:3393 ast_rtp_raw_write: Difference is 13488, ms is 301 [Feb 20 15:23:42] DEBUG[32542]: res_timing_timerfd.c:167 timerfd_timer_ack: Expected to acknowledge 1 ticks but got 4 instead [Feb 20 15:23:42] DEBUG[32584][C-00000002]: res_timing_timerfd.c:167 timerfd_timer_ack: Expected to acknowledge 1 ticks but got 9 instead [Feb 20 15:23:42] DEBUG[32563]: netsock2.c:172 ast_sockaddr_split_hostport: Splitting '192.168.1.147:14370' into... [Feb 20 15:23:42] DEBUG[32563]: netsock2.c:226 ast_sockaddr_split_hostport: ...host '192.168.1.147' and port '14370'. [Feb 20 15:23:42] DEBUG[32581]: res_timing_timerfd.c:167 timerfd_timer_ack: Expected to acknowledge 1 ticks but got 6 instead [Feb 20 15:23:42] DEBUG[32581]: res_timing_timerfd.c:167 timerfd_timer_ack: Expected to acknowledge 1 ticks but got 2 instead [Feb 20 15:23:42] DEBUG[32542]: res_timing_timerfd.c:167 timerfd_timer_ack: Expected to acknowledge 1 ticks but got 2 instead [Feb 20 15:23:43] DEBUG[32581]: res_timing_timerfd.c:167 timerfd_timer_ack: Expected to acknowledge 1 ticks but got 4 instead [Feb 20 15:23:43] DEBUG[32580][C-00000001]: res_rtp_asterisk.c:3393 ast_rtp_raw_write: Difference is 12768, ms is 286 [Feb 20 15:23:43] DEBUG[32581]: res_timing_timerfd.c:167 timerfd_timer_ack: Expected to acknowledge 1 ticks but got 2 instead [Feb 20 15:23:43] DEBUG[32584][C-00000002]: res_rtp_asterisk.c:3393 ast_rtp_raw_write: Difference is 14592, ms is 324 [Feb 20 15:23:43] DEBUG[32542]: res_timing_timerfd.c:167 timerfd_timer_ack: Expected to acknowledge 1 ticks but got 2 instead [Feb 20 15:23:43] DEBUG[32581]: res_timing_timerfd.c:167 timerfd_timer_ack: Expected to acknowledge 1 ticks but got 5 instead [Feb 20 15:23:43] DEBUG[32584][C-00000002]: res_timing_timerfd.c:167 timerfd_timer_ack: Expected to acknowledge 1 ticks but got 17 instead [Feb 20 15:23:43] DEBUG[32542]: res_timing_timerfd.c:167 timerfd_timer_ack: Expected to acknowledge 1 ticks but got 2 instead [Feb 20 15:23:43] DEBUG[32542]: res_timing_timerfd.c:167 timerfd_timer_ack: Expected to acknowledge 1 ticks but got 3 instead [Feb 20 15:23:43] DEBUG[32542]: res_timing_timerfd.c:167 timerfd_timer_ack: Expected to acknowledge 1 ticks but got 2 instead [Feb 20 15:23:43] DEBUG[32584][C-00000002]: res_rtp_asterisk.c:3393 ast_rtp_raw_write: Difference is 7728, ms is 181 [Feb 20 15:23:43] DEBUG[32581]: res_timing_timerfd.c:167 timerfd_timer_ack: Expected to acknowledge 1 ticks but got 8 instead [Feb 20 15:23:43] DEBUG[32580][C-00000001]: res_rtp_asterisk.c:3393 ast_rtp_raw_write: Difference is 12864, ms is 288 [Feb 20 15:23:43] DEBUG[32581]: res_timing_timerfd.c:167 timerfd_timer_ack: Expected to acknowledge 1 ticks but got 2 instead [Feb 20 15:23:43] DEBUG[32584][C-00000002]: res_timing_timerfd.c:167 timerfd_timer_ack: Expected to acknowledge 1 ticks but got 12 instead [Feb 20 15:23:43] DEBUG[32581]: res_timing_timerfd.c:167 timerfd_timer_ack: Expected to acknowledge 1 ticks but got 3 instead [Feb 20 15:23:43] DEBUG[32542]: res_timing_timerfd.c:167 timerfd_timer_ack: Expected to acknowledge 1 ticks but got 2 instead [Feb 20 15:23:43] DEBUG[32542]: res_timing_timerfd.c:167 timerfd_timer_ack: Expected to acknowledge 1 ticks but got 6 instead [Feb 20 15:23:43] DEBUG[32581]: res_timing_timerfd.c:167 timerfd_timer_ack: Expected to acknowledge 1 ticks but got 10 instead [Feb 20 15:23:43] DEBUG[32584][C-00000002]: res_rtp_asterisk.c:3393 ast_rtp_raw_write: Difference is 9936, ms is 227 [Feb 20 15:23:43] DEBUG[32580][C-00000001]: res_rtp_asterisk.c:3393 ast_rtp_raw_write: Difference is 13104, ms is 293 [Feb 20 15:23:43] DEBUG[32584][C-00000002]: res_timing_timerfd.c:167 timerfd_timer_ack: Expected to acknowledge 1 ticks but got 14 instead [Feb 20 15:23:43] DEBUG[32580][C-00000001]: res_rtp_asterisk.c:3393 ast_rtp_raw_write: Difference is 2544, ms is 73 [Feb 20 15:23:43] DEBUG[32584][C-00000002]: res_rtp_asterisk.c:3393 ast_rtp_raw_write: Difference is 9360, ms is 215 [Feb 20 15:23:43] DEBUG[32584][C-00000002]: res_timing_timerfd.c:167 timerfd_timer_ack: Expected to acknowledge 1 ticks but got 4 instead [Feb 20 15:23:43] DEBUG[32580][C-00000001]: res_rtp_asterisk.c:3393 ast_rtp_raw_write: Difference is 3696, ms is 97 [Feb 20 15:23:43] DEBUG[32584][C-00000002]: res_rtp_asterisk.c:3393 ast_rtp_raw_write: Difference is 4080, ms is 105 [Feb 20 15:23:43] DEBUG[32584][C-00000002]: res_timing_timerfd.c:167 timerfd_timer_ack: Expected to acknowledge 1 ticks but got 6 instead [Feb 20 15:23:43] DEBUG[32580][C-00000001]: res_rtp_asterisk.c:3393 ast_rtp_raw_write: Difference is 2496, ms is 72 [Feb 20 15:23:43] DEBUG[32580][C-00000001]: res_rtp_asterisk.c:3393 ast_rtp_raw_write: Difference is 2736, ms is 77 [Feb 20 15:23:43] DEBUG[32584][C-00000002]: res_rtp_asterisk.c:3393 ast_rtp_raw_write: Difference is 4992, ms is 124 [Feb 20 15:23:43] DEBUG[32584][C-00000002]: res_timing_timerfd.c:167 timerfd_timer_ack: Expected to acknowledge 1 ticks but got 5 instead [Feb 20 15:23:44] DEBUG[32542]: res_timing_timerfd.c:167 timerfd_timer_ack: Expected to acknowledge 1 ticks but got 5 instead [Feb 20 15:23:44] DEBUG[32581]: res_timing_timerfd.c:167 timerfd_timer_ack: Expected to acknowledge 1 ticks but got 6 instead [Feb 20 15:23:44] DEBUG[32584][C-00000002]: res_rtp_asterisk.c:3393 ast_rtp_raw_write: Difference is 7872, ms is 184 [Feb 20 15:23:44] DEBUG[32580][C-00000001]: res_rtp_asterisk.c:3393 ast_rtp_raw_write: Difference is 8496, ms is 197 [Feb 20 15:23:44] DEBUG[32584][C-00000002]: res_timing_timerfd.c:167 timerfd_timer_ack: Expected to acknowledge 1 ticks but got 10 instead [Feb 20 15:23:44] DEBUG[32584][C-00000002]: res_rtp_asterisk.c:3393 ast_rtp_raw_write: Difference is 4416, ms is 112 [Feb 20 15:23:44] DEBUG[32580][C-00000001]: res_rtp_asterisk.c:3393 ast_rtp_raw_write: Difference is 3360, ms is 90 [Feb 20 15:23:44] DEBUG[32584][C-00000002]: res_timing_timerfd.c:167 timerfd_timer_ack: Expected to acknowledge 1 ticks but got 5 instead [Feb 20 15:23:44] DEBUG[32580][C-00000001]: res_rtp_asterisk.c:3393 ast_rtp_raw_write: Difference is 2112, ms is 64 [Feb 20 15:23:44] DEBUG[32584][C-00000002]: res_rtp_asterisk.c:3393 ast_rtp_raw_write: Difference is 2400, ms is 70 [Feb 20 15:23:44] DEBUG[32584][C-00000002]: res_timing_timerfd.c:167 timerfd_timer_ack: Expected to acknowledge 1 ticks but got 4 instead [Feb 20 15:23:44] DEBUG[32584][C-00000002]: channel.c:3469 ast_settimeout_full: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Feb 20 15:23:44] DEBUG[32584][C-00000002]: channel.c:3469 ast_settimeout_full: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Feb 20 15:23:44] DEBUG[32584][C-00000002]: channel.c:3469 ast_settimeout_full: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Feb 20 15:23:44] DEBUG[32580][C-00000001]: res_rtp_asterisk.c:3393 ast_rtp_raw_write: Difference is 2784, ms is 78 [Feb 20 15:23:44] DEBUG[32584][C-00000002]: channel.c:5680 set_format: Channel SIP/50002-00000002 setting write format path: opus -> opus [Feb 20 15:23:44] DEBUG[32584][C-00000002]: pbx_lua.c:1480 lua_find_extension: Looking up 350001@default:2 [Feb 20 15:23:44] DEBUG[32580][C-00000001]: res_rtp_asterisk.c:3393 ast_rtp_raw_write: Difference is 2832, ms is 79 [Feb 20 15:23:44] DEBUG[32584][C-00000002]: pbx.c:2875 pbx_extension_helper: Launching 'Answer' -- Executing [350001@default:2] Answer("SIP/50002-00000002", "") in new stack [Feb 20 15:23:44] DEBUG[32580][C-00000001]: res_rtp_asterisk.c:3393 ast_rtp_raw_write: Difference is 2496, ms is 72 [Feb 20 15:23:44] DEBUG[32581]: res_timing_timerfd.c:167 timerfd_timer_ack: Expected to acknowledge 1 ticks but got 2 instead [Feb 20 15:23:44] DEBUG[32542]: res_timing_timerfd.c:167 timerfd_timer_ack: Expected to acknowledge 1 ticks but got 3 instead [Feb 20 15:23:44] DEBUG[32581]: res_timing_timerfd.c:167 timerfd_timer_ack: Expected to acknowledge 1 ticks but got 2 instead [Feb 20 15:23:44] DEBUG[32584][C-00000002]: pbx_lua.c:1480 lua_find_extension: Looking up 350001@default:3 [Feb 20 15:23:44] DEBUG[32584][C-00000002]: pbx_variables.c:708 pbx_substitute_variables_helper_full: Function CALLERID(num) result is '50002' [Feb 20 15:23:44] DEBUG[32580][C-00000001]: res_rtp_asterisk.c:3393 ast_rtp_raw_write: Difference is 6576, ms is 157 [Feb 20 15:23:44] DEBUG[32584][C-00000002]: pbx.c:2875 pbx_extension_helper: Launching 'Set' -- Executing [350001@default:3] Set("SIP/50002-00000002", "CID=350002") in new stack [Feb 20 15:23:44] DEBUG[32580][C-00000001]: res_rtp_asterisk.c:3393 ast_rtp_raw_write: Difference is 3168, ms is 86 [Feb 20 15:23:44] DEBUG[32584][C-00000002]: pbx_lua.c:1480 lua_find_extension: Looking up 350001@default:4 [Feb 20 15:23:44] DEBUG[32584][C-00000002]: pbx_variables.c:381 ast_str_retrieve_variable: Result of 'CID' is '350002' [Feb 20 15:23:44] DEBUG[32584][C-00000002]: pbx_variables.c:381 ast_str_retrieve_variable: Result of 'EXTEN' is '350001' [Feb 20 15:23:44] DEBUG[32584][C-00000002]: pbx_variables.c:777 pbx_substitute_variables_helper_full: Expression result is '0' [Feb 20 15:23:44] DEBUG[32584][C-00000002]: pbx.c:2875 pbx_extension_helper: Launching 'GotoIf' -- Executing [350001@default:4] GotoIf("SIP/50002-00000002", "0?cme") in new stack [Feb 20 15:23:44] DEBUG[32580][C-00000001]: res_rtp_asterisk.c:3393 ast_rtp_raw_write: Difference is 3264, ms is 88 [Feb 20 15:23:44] DEBUG[32584][C-00000002]: pbx_builtins.c:1174 pbx_builtin_gotoif: Not taking any branch [Feb 20 15:23:44] DEBUG[32584][C-00000002]: pbx_lua.c:1480 lua_find_extension: Looking up 350001@default:5 [Feb 20 15:23:44] DEBUG[32580][C-00000001]: res_rtp_asterisk.c:3393 ast_rtp_raw_write: Difference is 4032, ms is 104 [Feb 20 15:23:44] DEBUG[32584][C-00000002]: pbx_variables.c:381 ast_str_retrieve_variable: Result of 'CID' is '350002' [Feb 20 15:23:44] DEBUG[32584][C-00000002]: pbx_variables.c:378 ast_str_retrieve_variable: Result of 'exten' is NULL [Feb 20 15:23:44] DEBUG[32584][C-00000002]: pbx.c:2875 pbx_extension_helper: Launching 'GotoIf' -- Executing [350001@default:5] GotoIf("SIP/50002-00000002", "350002 = ") in new stack [Feb 20 15:23:45] DEBUG[32584][C-00000002]: pbx_builtins.c:1174 pbx_builtin_gotoif: Not taking any branch [Feb 20 15:23:45] DEBUG[32584][C-00000002]: pbx_lua.c:1480 lua_find_extension: Looking up 350001@default:6 [Feb 20 15:23:45] DEBUG[32580][C-00000001]: res_rtp_asterisk.c:3393 ast_rtp_raw_write: Difference is 3504, ms is 93 [Feb 20 15:23:45] DEBUG[32572][C-00000002]: res_rtp_asterisk.c:4212 ast_rtcp_read: Got RTCP report of 100 bytes [Feb 20 15:23:45] DEBUG[32584][C-00000002]: pbx_variables.c:381 ast_str_retrieve_variable: Result of 'EXTEN' is '350001' [Feb 20 15:23:45] DEBUG[32584][C-00000002]: pbx.c:2875 pbx_extension_helper: Launching 'ConfBridge' [Feb 20 15:23:45] DEBUG[32563]: netsock2.c:172 ast_sockaddr_split_hostport: Splitting '192.168.1.147:14593' into... [Feb 20 15:23:45] DEBUG[32563]: netsock2.c:226 ast_sockaddr_split_hostport: ...host '192.168.1.147' and port '14593'. [Feb 20 15:23:45] DEBUG[32563]: netsock2.c:172 ast_sockaddr_split_hostport: Splitting '192.168.41.78:19329' into... -- Executing [350001@default:6] ConfBridge("SIP/50002-00000002", "350001,default_bridge_32,default_user,sample_user_menu") in new stack [Feb 20 15:23:45] DEBUG[32563]: netsock2.c:226 ast_sockaddr_split_hostport: ...host '192.168.41.78' and port '19329'. [Feb 20 15:23:45] DEBUG[32580][C-00000001]: res_rtp_asterisk.c:3393 ast_rtp_raw_write: Difference is 3744, ms is 98 [Feb 20 15:23:45] DEBUG[32580][C-00000001]: res_rtp_asterisk.c:3393 ast_rtp_raw_write: Difference is 3984, ms is 103 [Feb 20 15:23:45] DEBUG[32584][C-00000002]: app_confbridge.c:1422 join_conference_bridge: Trying to find conference bridge '350001' [Feb 20 15:23:45] DEBUG[32584][C-00000002]: app_confbridge.c:1145 conf_update_user_mute: User SIP/50001-00000001 is unmuted: user:0 system:0. [Feb 20 15:23:45] DEBUG[32584][C-00000002]: app_confbridge.c:1145 conf_update_user_mute: User SIP/50002-00000002 is unmuted: user:0 system:0. [Feb 20 15:23:45] DEBUG[32584][C-00000002]: confbridge/conf_state.c:84 conf_change_state: Changing conference '350001' state from SINGLE_MARKED to MULTI_MARKED [Feb 20 15:23:45] DEBUG[32580][C-00000001]: res_rtp_asterisk.c:3393 ast_rtp_raw_write: Difference is 2976, ms is 82 [Feb 20 15:23:45] DEBUG[32584][C-00000002]: bridge_channel.c:2654 bridge_channel_internal_join: Bridge 28f1b4ee-63be-49cb-b551-d5a7209f7781: 0xabebb28(SIP/50002-00000002) is joining [Feb 20 15:23:45] DEBUG[32580][C-00000001]: res_rtp_asterisk.c:3393 ast_rtp_raw_write: Difference is 2880, ms is 80 [Feb 20 15:23:45] DEBUG[32584][C-00000002]: bridge_channel.c:2118 bridge_channel_internal_push_full: Bridge 28f1b4ee-63be-49cb-b551-d5a7209f7781: pushing 0xabebb28(SIP/50002-00000002) -- Channel SIP/50002-00000002 joined 'softmix' base-bridge <28f1b4ee-63be-49cb-b551-d5a7209f7781> [Feb 20 15:23:45] DEBUG[32584][C-00000002]: bridge.c:432 bridge_channel_complete_join: Bridge 28f1b4ee-63be-49cb-b551-d5a7209f7781: 0xabebb28(SIP/50002-00000002) is joining softmix technology [Feb 20 15:23:45] DEBUG[32580][C-00000001]: res_rtp_asterisk.c:3393 ast_rtp_raw_write: Difference is 2064, ms is 63 [Feb 20 15:23:45] DEBUG[32496]: cdr.c:1293 cdr_object_finalize: Finalized CDR for SIP/50002-00000002 - start 1487575419.396883 answer 1487575420.117179 end 1487575425.493797 dispo ANSWERED [Feb 20 15:23:45] DEBUG[32584][C-00000002]: channel.c:5444 ast_set_read_format_path: Channel SIP/50002-00000002 setting read format path: opus -> slin [Feb 20 15:23:45] DEBUG[32580][C-00000001]: res_rtp_asterisk.c:3393 ast_rtp_raw_write: Difference is 2256, ms is 67 [Feb 20 15:23:45] DEBUG[32584][C-00000002]: channel.c:5680 set_format: Channel SIP/50002-00000002 setting write format path: slin -> opus [Feb 20 15:23:45] DEBUG[32584][C-00000002]: dsp.c:499 ast_tone_detect_init: Setup tone 1100 Hz, 500 ms, block_size=160, hits_required=21 [Feb 20 15:23:45] DEBUG[32584][C-00000002]: dsp.c:499 ast_tone_detect_init: Setup tone 2100 Hz, 2600 ms, block_size=160, hits_required=116 [Feb 20 15:23:45] DEBUG[32580][C-00000001]: res_rtp_asterisk.c:3393 ast_rtp_raw_write: Difference is 1296, ms is 47 [Feb 20 15:23:45] DEBUG[32580][C-00000001]: res_rtp_asterisk.c:3393 ast_rtp_raw_write: Difference is 1584, ms is 53 [Feb 20 15:23:45] DEBUG[32563]: netsock2.c:172 ast_sockaddr_split_hostport: Splitting '192.168.41.78:15177' into... [Feb 20 15:23:45] DEBUG[32563]: netsock2.c:226 ast_sockaddr_split_hostport: ...host '192.168.41.78' and port '15177'. [Feb 20 15:23:45] DEBUG[32563]: netsock2.c:172 ast_sockaddr_split_hostport: Splitting '192.168.1.147:14498' into... [Feb 20 15:23:45] DEBUG[32563]: netsock2.c:226 ast_sockaddr_split_hostport: ...host '192.168.1.147' and port '14498'. [Feb 20 15:23:45] DEBUG[32580][C-00000001]: res_rtp_asterisk.c:3393 ast_rtp_raw_write: Difference is 2352, ms is 69 [Feb 20 15:23:45] DEBUG[32580][C-00000001]: res_rtp_asterisk.c:3393 ast_rtp_raw_write: Difference is 4608, ms is 116 [Feb 20 15:23:45] DEBUG[32580][C-00000001]: res_rtp_asterisk.c:3393 ast_rtp_raw_write: Difference is 2592, ms is 74 [Feb 20 15:23:45] DEBUG[32584][C-00000002]: res_rtp_asterisk.c:3052 ast_rtp_update_source: Setting the marker bit due to a source update [Feb 20 15:23:45] DEBUG[32584][C-00000002]: res_rtp_asterisk.c:3052 ast_rtp_update_source: Setting the marker bit due to a source update [Feb 20 15:23:45] DEBUG[32581]: res_timing_timerfd.c:167 timerfd_timer_ack: Expected to acknowledge 1 ticks but got 30 instead [Feb 20 15:23:45] DEBUG[32580][C-00000001]: res_rtp_asterisk.c:3393 ast_rtp_raw_write: Difference is 1872, ms is 59 [Feb 20 15:23:46] DEBUG[32582]: channel.c:5680 set_format: Channel CBAnn/350001-00000001;1 setting write format path: gsm -> slin [Feb 20 15:23:46] DEBUG[32584][C-00000002]: res_rtp_asterisk.c:3393 ast_rtp_raw_write: Difference is 80208, ms is 1691 [Feb 20 15:23:46] DEBUG[32581]: res_timing_timerfd.c:167 timerfd_timer_ack: Expected to acknowledge 1 ticks but got 3 instead [Feb 20 15:23:46] DEBUG[32582]: channel.c:3469 ast_settimeout_full: Scheduling timer at (50 requested / 50 actual) timer ticks per second <--- SIP read from UDP:192.168.1.147:43521 ---> <-------------> [Feb 20 15:23:46] DEBUG[32544]: chan_sip.c:9893 parse_request: Header 0 [ 0]: -- Playing 'confbridge-join.gsm' (language 'en') [Feb 20 15:23:46] DEBUG[32584][C-00000002]: res_rtp_asterisk.c:3393 ast_rtp_raw_write: Difference is 2160, ms is 65 [Feb 20 15:23:46] DEBUG[32580][C-00000001]: res_rtp_asterisk.c:3393 ast_rtp_raw_write: Difference is 4320, ms is 110 [Feb 20 15:23:46] DEBUG[32582]: res_timing_timerfd.c:167 timerfd_timer_ack: Expected to acknowledge 1 ticks but got 2 instead [Feb 20 15:23:46] DEBUG[32581]: res_timing_timerfd.c:167 timerfd_timer_ack: Expected to acknowledge 1 ticks but got 5 instead [Feb 20 15:23:46] DEBUG[32584][C-00000002]: res_rtp_asterisk.c:3393 ast_rtp_raw_write: Difference is 2016, ms is 62 [Feb 20 15:23:46] DEBUG[32580][C-00000001]: res_rtp_asterisk.c:3393 ast_rtp_raw_write: Difference is 3168, ms is 86 [Feb 20 15:23:46] DEBUG[32582]: res_timing_timerfd.c:167 timerfd_timer_ack: Expected to acknowledge 1 ticks but got 2 instead [Feb 20 15:23:46] DEBUG[32584][C-00000002]: res_rtp_asterisk.c:3393 ast_rtp_raw_write: Difference is 3264, ms is 88 [Feb 20 15:23:46] DEBUG[32581]: res_timing_timerfd.c:167 timerfd_timer_ack: Expected to acknowledge 1 ticks but got 4 instead [Feb 20 15:23:46] DEBUG[32580][C-00000001]: res_rtp_asterisk.c:3393 ast_rtp_raw_write: Difference is 3120, ms is 85 [Feb 20 15:23:46] DEBUG[32582]: res_timing_timerfd.c:167 timerfd_timer_ack: Expected to acknowledge 1 ticks but got 2 instead [Feb 20 15:23:46] DEBUG[32581]: res_timing_timerfd.c:167 timerfd_timer_ack: Expected to acknowledge 1 ticks but got 3 instead [Feb 20 15:23:46] DEBUG[32584][C-00000002]: res_rtp_asterisk.c:3393 ast_rtp_raw_write: Difference is 2688, ms is 76 [Feb 20 15:23:46] DEBUG[32582]: res_timing_timerfd.c:167 timerfd_timer_ack: Expected to acknowledge 1 ticks but got 2 instead [Feb 20 15:23:46] DEBUG[32580][C-00000001]: res_rtp_asterisk.c:3393 ast_rtp_raw_write: Difference is 2784, ms is 78 [Feb 20 15:23:46] DEBUG[32581]: res_timing_timerfd.c:167 timerfd_timer_ack: Expected to acknowledge 1 ticks but got 5 instead [Feb 20 15:23:46] DEBUG[32584][C-00000002]: res_rtp_asterisk.c:3393 ast_rtp_raw_write: Difference is 4416, ms is 112 [Feb 20 15:23:46] DEBUG[32563]: netsock2.c:172 ast_sockaddr_split_hostport: Splitting '192.168.41.78:14091' into... [Feb 20 15:23:46] DEBUG[32563]: netsock2.c:226 ast_sockaddr_split_hostport: ...host '192.168.41.78' and port '14091'. [Feb 20 15:23:46] DEBUG[32563]: netsock2.c:172 ast_sockaddr_split_hostport: Splitting '192.168.1.147:14338' into... [Feb 20 15:23:46] DEBUG[32563]: netsock2.c:226 ast_sockaddr_split_hostport: ...host '192.168.1.147' and port '14338'. [Feb 20 15:23:46] DEBUG[32582]: res_timing_timerfd.c:167 timerfd_timer_ack: Expected to acknowledge 1 ticks but got 2 instead [Feb 20 15:23:46] DEBUG[32580][C-00000001]: res_rtp_asterisk.c:3393 ast_rtp_raw_write: Difference is 4272, ms is 109 [Feb 20 15:23:46] DEBUG[32582]: res_timing_timerfd.c:167 timerfd_timer_ack: Expected to acknowledge 1 ticks but got 3 instead [Feb 20 15:23:46] DEBUG[32581]: res_timing_timerfd.c:167 timerfd_timer_ack: Expected to acknowledge 1 ticks but got 6 instead [Feb 20 15:23:46] DEBUG[32582]: res_timing_timerfd.c:167 timerfd_timer_ack: Expected to acknowledge 1 ticks but got 2 instead [Feb 20 15:23:46] DEBUG[32580][C-00000001]: res_rtp_asterisk.c:3393 ast_rtp_raw_write: Difference is 4176, ms is 107 [Feb 20 15:23:46] DEBUG[32584][C-00000002]: res_rtp_asterisk.c:3393 ast_rtp_raw_write: Difference is 5424, ms is 133 [Feb 20 15:23:46] DEBUG[32582]: res_timing_timerfd.c:167 timerfd_timer_ack: Expected to acknowledge 1 ticks but got 3 instead [Feb 20 15:23:46] DEBUG[32581]: res_timing_timerfd.c:167 timerfd_timer_ack: Expected to acknowledge 1 ticks but got 4 instead [Feb 20 15:23:46] DEBUG[32584][C-00000002]: res_rtp_asterisk.c:3393 ast_rtp_raw_write: Difference is 2592, ms is 74 [Feb 20 15:23:46] DEBUG[32580][C-00000001]: res_rtp_asterisk.c:3393 ast_rtp_raw_write: Difference is 2736, ms is 77 [Feb 20 15:23:46] DEBUG[32582]: res_timing_timerfd.c:167 timerfd_timer_ack: Expected to acknowledge 1 ticks but got 2 instead [Feb 20 15:23:46] DEBUG[32584][C-00000002]: res_rtp_asterisk.c:3393 ast_rtp_raw_write: Difference is 2928, ms is 81 [Feb 20 15:23:46] DEBUG[32581]: res_timing_timerfd.c:167 timerfd_timer_ack: Expected to acknowledge 1 ticks but got 5 instead [Feb 20 15:23:46] DEBUG[32582]: channel.c:3469 ast_settimeout_full: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Feb 20 15:23:46] DEBUG[32582]: channel.c:3469 ast_settimeout_full: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Feb 20 15:23:46] DEBUG[32582]: channel.c:3469 ast_settimeout_full: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Feb 20 15:23:46] DEBUG[32584][C-00000002]: res_rtp_asterisk.c:4212 ast_rtcp_read: Got RTCP report of 100 bytes [Feb 20 15:23:46] DEBUG[32580][C-00000001]: res_rtp_asterisk.c:3393 ast_rtp_raw_write: Difference is 2400, ms is 70 [Feb 20 15:23:46] DEBUG[32582]: channel.c:5680 set_format: Channel CBAnn/350001-00000001;1 setting write format path: slin -> slin [Feb 20 15:23:46] DEBUG[32581]: res_timing_timerfd.c:167 timerfd_timer_ack: Expected to acknowledge 1 ticks but got 5 instead [Feb 20 15:23:46] DEBUG[32580][C-00000001]: res_rtp_asterisk.c:3393 ast_rtp_raw_write: Difference is 3984, ms is 103 [Feb 20 15:23:46] DEBUG[32563]: netsock2.c:172 ast_sockaddr_split_hostport: Splitting '192.168.1.147:14593' into... [Feb 20 15:23:46] DEBUG[32563]: netsock2.c:226 ast_sockaddr_split_hostport: ...host '192.168.1.147' and port '14593'. [Feb 20 15:23:46] DEBUG[32563]: netsock2.c:172 ast_sockaddr_split_hostport: Splitting '192.168.41.78:19329' into... [Feb 20 15:23:46] DEBUG[32563]: netsock2.c:226 ast_sockaddr_split_hostport: ...host '192.168.41.78' and port '19329'. [Feb 20 15:23:46] DEBUG[32584][C-00000002]: res_rtp_asterisk.c:3393 ast_rtp_raw_write: Difference is 6624, ms is 158 [Feb 20 15:23:46] DEBUG[32581]: res_timing_timerfd.c:167 timerfd_timer_ack: Expected to acknowledge 1 ticks but got 4 instead [Feb 20 15:23:46] DEBUG[32580][C-00000001]: res_rtp_asterisk.c:3393 ast_rtp_raw_write: Difference is 3072, ms is 84 [Feb 20 15:23:46] DEBUG[32581]: res_timing_timerfd.c:167 timerfd_timer_ack: Expected to acknowledge 1 ticks but got 4 instead [Feb 20 15:23:46] DEBUG[32584][C-00000002]: res_rtp_asterisk.c:3393 ast_rtp_raw_write: Difference is 1536, ms is 52 [Feb 20 15:23:46] DEBUG[32580][C-00000001]: res_rtp_asterisk.c:3393 ast_rtp_raw_write: Difference is 3312, ms is 89 [Feb 20 15:23:47] DEBUG[32584][C-00000002]: res_rtp_asterisk.c:3393 ast_rtp_raw_write: Difference is 2688, ms is 76 [Feb 20 15:23:47] DEBUG[32581]: res_timing_timerfd.c:167 timerfd_timer_ack: Expected to acknowledge 1 ticks but got 4 instead [Feb 20 15:23:47] DEBUG[32580][C-00000001]: res_rtp_asterisk.c:3393 ast_rtp_raw_write: Difference is 1680, ms is 55 [Feb 20 15:23:47] DEBUG[32584][C-00000002]: res_rtp_asterisk.c:4564 ast_rtp_read: 0x9442c50 -- Probation learning mode pass with source address 192.168.1.147:14561 > 0x9442c50 -- Probation passed - setting RTP source address to 192.168.1.147:14561 [Feb 20 15:23:47] DEBUG[32584][C-00000002]: acl.c:957 ast_ouraddrfor: For destination '192.168.1.147', our source address is '192.168.41.78'. [Feb 20 15:23:47] DEBUG[32584][C-00000002]: res_rtp_asterisk.c:5036 ast_rtp_remote_address_set: Setting RTCP address on RTP instance '0xa5db3e8' [Feb 20 15:23:47] DEBUG[32581]: res_timing_timerfd.c:167 timerfd_timer_ack: Expected to acknowledge 1 ticks but got 4 instead [Feb 20 15:23:47] DEBUG[32580][C-00000001]: res_rtp_asterisk.c:3393 ast_rtp_raw_write: Difference is 2112, ms is 64 [Feb 20 15:23:47] DEBUG[32584][C-00000002]: res_rtp_asterisk.c:3393 ast_rtp_raw_write: Difference is 5040, ms is 125 [Feb 20 15:23:47] DEBUG[32584][C-00000002]: res_rtp_asterisk.c:4564 ast_rtp_read: 0x9442c50 -- Probation learning mode pass with source address 192.168.1.147:14561 [Feb 20 15:23:47] DEBUG[32581]: res_timing_timerfd.c:167 timerfd_timer_ack: Expected to acknowledge 1 ticks but got 4 instead > 0x9442c50 -- Probation passed - setting RTP source address to 192.168.1.147:14561 [Feb 20 15:23:47] DEBUG[32580][C-00000001]: res_rtp_asterisk.c:3393 ast_rtp_raw_write: Difference is 2976, ms is 82 [Feb 20 15:23:47] DEBUG[32581]: res_timing_timerfd.c:167 timerfd_timer_ack: Expected to acknowledge 1 ticks but got 3 instead [Feb 20 15:23:47] DEBUG[32584][C-00000002]: res_rtp_asterisk.c:3393 ast_rtp_raw_write: Difference is 4512, ms is 114 [Feb 20 15:23:47] DEBUG[32580][C-00000001]: res_rtp_asterisk.c:3393 ast_rtp_raw_write: Difference is 3792, ms is 99 [Feb 20 15:23:47] DEBUG[32584][C-00000002]: res_rtp_asterisk.c:3393 ast_rtp_raw_write: Difference is 672, ms is 34 [Feb 20 15:23:47] DEBUG[32580][C-00000001]: res_rtp_asterisk.c:3393 ast_rtp_raw_write: Difference is 1440, ms is 50 [Feb 20 15:23:47] DEBUG[32581]: res_timing_timerfd.c:167 timerfd_timer_ack: Expected to acknowledge 1 ticks but got 4 instead [Feb 20 15:23:47] DEBUG[32584][C-00000002]: res_rtp_asterisk.c:3393 ast_rtp_raw_write: Difference is 2544, ms is 73 [Feb 20 15:23:47] DEBUG[32580][C-00000001]: res_rtp_asterisk.c:3393 ast_rtp_raw_write: Difference is 2496, ms is 72 [Feb 20 15:23:47] DEBUG[32584][C-00000002]: res_rtp_asterisk.c:3393 ast_rtp_raw_write: Difference is 1872, ms is 59 [Feb 20 15:23:47] DEBUG[32581]: res_timing_timerfd.c:167 timerfd_timer_ack: Expected to acknowledge 1 ticks but got 5 instead [Feb 20 15:23:47] DEBUG[32542]: res_timing_timerfd.c:167 timerfd_timer_ack: Expected to acknowledge 1 ticks but got 2 instead [Feb 20 15:23:47] DEBUG[32581]: res_timing_timerfd.c:167 timerfd_timer_ack: Expected to acknowledge 1 ticks but got 3 instead [Feb 20 15:23:47] DEBUG[32584][C-00000002]: res_rtp_asterisk.c:3393 ast_rtp_raw_write: Difference is 2688, ms is 76 [Feb 20 15:23:47] DEBUG[32580][C-00000001]: res_rtp_asterisk.c:3393 ast_rtp_raw_write: Difference is 2832, ms is 79 [Feb 20 15:23:47] DEBUG[32581]: res_timing_timerfd.c:167 timerfd_timer_ack: Expected to acknowledge 1 ticks but got 4 instead [Feb 20 15:23:47] DEBUG[32580][C-00000001]: res_rtp_asterisk.c:3393 ast_rtp_raw_write: Difference is 2496, ms is 72 [Feb 20 15:23:47] DEBUG[32584][C-00000002]: res_rtp_asterisk.c:3393 ast_rtp_raw_write: Difference is 2448, ms is 71 > Video source in bridge '350001' (28f1b4ee-63be-49cb-b551-d5a7209f7781) is now 'SIP/50002-00000002' (1487575419.7) [Feb 20 15:23:47] DEBUG[32580][C-00000001]: res_rtp_asterisk.c:3393 ast_rtp_raw_write: Difference is 2928, ms is 81 [Feb 20 15:23:47] DEBUG[32581]: res_timing_timerfd.c:167 timerfd_timer_ack: Expected to acknowledge 1 ticks but got 4 instead [Feb 20 15:23:47] DEBUG[32563]: netsock2.c:172 ast_sockaddr_split_hostport: Splitting '192.168.41.78:16635' into... [Feb 20 15:23:47] DEBUG[32563]: netsock2.c:226 ast_sockaddr_split_hostport: ...host '192.168.41.78' and port '16635'. [Feb 20 15:23:47] DEBUG[32563]: netsock2.c:172 ast_sockaddr_split_hostport: Splitting '192.168.1.147:14370' into... [Feb 20 15:23:47] DEBUG[32563]: netsock2.c:226 ast_sockaddr_split_hostport: ...host '192.168.1.147' and port '14370'. [Feb 20 15:23:47] DEBUG[32580][C-00000001]: res_rtp_asterisk.c:3393 ast_rtp_raw_write: Difference is 2112, ms is 64 [Feb 20 15:23:47] DEBUG[32584][C-00000002]: res_rtp_asterisk.c:3393 ast_rtp_raw_write: Difference is 7920, ms is 185 [Feb 20 15:23:47] DEBUG[32580][C-00000001]: res_rtp_asterisk.c:3393 ast_rtp_raw_write: Difference is 3216, ms is 87 [Feb 20 15:23:47] DEBUG[32581]: res_timing_timerfd.c:167 timerfd_timer_ack: Expected to acknowledge 1 ticks but got 8 instead [Feb 20 15:23:47] DEBUG[32580][C-00000001]: res_rtp_asterisk.c:3393 ast_rtp_raw_write: Difference is 1440, ms is 50 [Feb 20 15:23:47] DEBUG[32584][C-00000002]: res_rtp_asterisk.c:3625 ast_rtp_write: Ooh, format changed from none to vp8 [Feb 20 15:23:47] DEBUG[32584][C-00000002]: res_rtp_asterisk.c:3393 ast_rtp_raw_write: Difference is 4752, ms is 119 [Feb 20 15:23:47] DEBUG[32581]: res_timing_timerfd.c:167 timerfd_timer_ack: Expected to acknowledge 1 ticks but got 3 instead [Feb 20 15:23:47] DEBUG[32580][C-00000001]: res_rtp_asterisk.c:3393 ast_rtp_raw_write: Difference is 2160, ms is 65 [Feb 20 15:23:47] DEBUG[32584][C-00000002]: res_rtp_asterisk.c:3393 ast_rtp_raw_write: Difference is 1104, ms is 43 [Feb 20 15:23:47] DEBUG[32584][C-00000002]: res_rtp_asterisk.c:3408 ast_rtp_raw_write: Difference is 49230, ms is 107 (9630), pred/ts/samples 51930/2700/2700 [Feb 20 15:23:47] DEBUG[32580][C-00000001]: res_rtp_asterisk.c:3393 ast_rtp_raw_write: Difference is 2400, ms is 70 [Feb 20 15:23:47] DEBUG[32581]: bridge_softmix.c:860 analyse_softmix_stats: Multiple above internal rate. Bridge changed from 8000 to 48000. [Feb 20 15:23:47] DEBUG[32581]: res_timing_timerfd.c:167 timerfd_timer_ack: Expected to acknowledge 1 ticks but got 6 instead [Feb 20 15:23:47] DEBUG[32584][C-00000002]: res_rtp_asterisk.c:3393 ast_rtp_raw_write: Difference is 2976, ms is 82 [Feb 20 15:23:48] DEBUG[32581]: channel.c:5444 ast_set_read_format_path: Channel CBAnn/350001-00000001;2 setting read format path: slin -> slin48 [Feb 20 15:23:48] DEBUG[32580][C-00000001]: res_rtp_asterisk.c:3393 ast_rtp_raw_write: Difference is 2544, ms is 73 [Feb 20 15:23:48] DEBUG[32581]: channel.c:5680 set_format: Channel CBAnn/350001-00000001;2 setting write format path: slin48 -> slin [Feb 20 15:23:48] DEBUG[32581]: dsp.c:499 ast_tone_detect_init: Setup tone 1100 Hz, 500 ms, block_size=960, hits_required=21 [Feb 20 15:23:48] DEBUG[32581]: dsp.c:499 ast_tone_detect_init: Setup tone 2100 Hz, 2600 ms, block_size=960, hits_required=116 [Feb 20 15:23:48] DEBUG[32580][C-00000001]: res_rtp_asterisk.c:3393 ast_rtp_raw_write: Difference is 1632, ms is 54 [Feb 20 15:23:48] DEBUG[32581]: channel.c:5444 ast_set_read_format_path: Channel SIP/50001-00000001 setting read format path: opus -> slin48 [Feb 20 15:23:48] DEBUG[32581]: channel.c:5680 set_format: Channel SIP/50001-00000001 setting write format path: slin48 -> opus [Feb 20 15:23:48] DEBUG[32581]: dsp.c:499 ast_tone_detect_init: Setup tone 1100 Hz, 500 ms, block_size=960, hits_required=21 [Feb 20 15:23:48] DEBUG[32581]: dsp.c:499 ast_tone_detect_init: Setup tone 2100 Hz, 2600 ms, block_size=960, hits_required=116 [Feb 20 15:23:48] DEBUG[32581]: channel.c:5444 ast_set_read_format_path: Channel SIP/50002-00000002 setting read format path: opus -> slin48 [Feb 20 15:23:48] DEBUG[32580][C-00000001]: channel.c:5264 ast_write: Channel SIP/50001-00000001 changing write format from slin48 to slin, native formats (vp8|opus) [Feb 20 15:23:48] DEBUG[32581]: channel.c:5680 set_format: Channel SIP/50002-00000002 setting write format path: slin48 -> opus [Feb 20 15:23:48] DEBUG[32581]: dsp.c:499 ast_tone_detect_init: Setup tone 1100 Hz, 500 ms, block_size=960, hits_required=21 [Feb 20 15:23:48] DEBUG[32581]: dsp.c:499 ast_tone_detect_init: Setup tone 2100 Hz, 2600 ms, block_size=960, hits_required=116 [Feb 20 15:23:48] DEBUG[32580][C-00000001]: channel.c:5680 set_format: Channel SIP/50001-00000001 setting write format path: slin -> opus [Feb 20 15:23:48] DEBUG[32581]: res_timing_timerfd.c:167 timerfd_timer_ack: Expected to acknowledge 1 ticks but got 11 instead <--- SIP read from UDP:192.168.1.147:23681 ---> BYE sip:350001@192.168.1.227:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.147:23681;branch=z9hG4bK.rUwOPluXX;rport From: ;tag=X3ughL6Dw To: ;tag=as188ae036 CSeq: 22 BYE Call-ID: AOK5~O9uyp Max-Forwards: 70 User-Agent: Linphone/3.10.2 (belle-sip/1.5.0) Authorization: Digest realm="asterisk", nonce="2d0b2d61", algorithm=MD5, username="50002", uri="sip:350001@192.168.1.227:5060", response="fb15e343bca87c65212124ec31f60475" <-------------> [Feb 20 15:23:48] DEBUG[32544]: chan_sip.c:9893 parse_request: Header 0 [ 42]: BYE sip:350001@192.168.1.227:5060 SIP/2.0 [Feb 20 15:23:48] DEBUG[32544]: chan_sip.c:9893 parse_request: Header 1 [ 69]: Via: SIP/2.0/UDP 192.168.1.147:23681;branch=z9hG4bK.rUwOPluXX;rport [Feb 20 15:23:48] DEBUG[32544]: chan_sip.c:9893 parse_request: Header 2 [ 46]: From: ;tag=X3ughL6Dw [Feb 20 15:23:48] DEBUG[32544]: chan_sip.c:9893 parse_request: Header 3 [ 46]: To: ;tag=as188ae036 [Feb 20 15:23:48] DEBUG[32580][C-00000001]: res_rtp_asterisk.c:3393 ast_rtp_raw_write: Difference is 4320, ms is 110 [Feb 20 15:23:48] DEBUG[32584][C-00000002]: res_rtp_asterisk.c:3393 ast_rtp_raw_write: Difference is 9792, ms is 224 [Feb 20 15:23:48] DEBUG[32544]: chan_sip.c:9893 parse_request: Header 4 [ 12]: CSeq: 22 BYE [Feb 20 15:23:48] DEBUG[32544]: chan_sip.c:9893 parse_request: Header 5 [ 19]: Call-ID: AOK5~O9uyp [Feb 20 15:23:48] DEBUG[32544]: chan_sip.c:9893 parse_request: Header 6 [ 16]: Max-Forwards: 70 [Feb 20 15:23:48] DEBUG[32544]: chan_sip.c:9893 parse_request: Header 7 [ 45]: User-Agent: Linphone/3.10.2 (belle-sip/1.5.0) [Feb 20 15:23:48] DEBUG[32544]: chan_sip.c:9893 parse_request: Header 8 [172]: Authorization: Digest realm="asterisk", nonce="2d0b2d61", algorithm=MD5, username="50002", uri="sip:350001@192.168.1.227:5060", response="fb15e343bca87c65212124ec31f60475" --- (9 headers 0 lines) --- [Feb 20 15:23:48] DEBUG[32584][C-00000002]: res_rtp_asterisk.c:3408 ast_rtp_raw_write: Difference is 18540, ms is 283 (25470), pred/ts/samples 70470/51930/0 [Feb 20 15:23:48] DEBUG[32581]: res_timing_timerfd.c:167 timerfd_timer_ack: Expected to acknowledge 1 ticks but got 3 instead [Feb 20 15:23:48] DEBUG[32544]: chan_sip.c:9419 __find_call: = Looking for Call ID: AOK5~O9uyp (Checking From) --From tag X3ughL6Dw --To-tag as188ae036 [Feb 20 15:23:48] DEBUG[32584][C-00000002]: res_rtp_asterisk.c:3393 ast_rtp_raw_write: Difference is 1680, ms is 55 [Feb 20 15:23:48] DEBUG[32581]: res_timing_timerfd.c:167 timerfd_timer_ack: Expected to acknowledge 1 ticks but got 2 instead [Feb 20 15:23:48] DEBUG[32544][C-00000002]: chan_sip.c:28652 handle_incoming: **** Received BYE (8) - Command in SIP BYE [Feb 20 15:23:48] DEBUG[32544][C-00000002]: chan_sip.c:27096 handle_request_bye: Initializing initreq for method BYE - callid AOK5~O9uyp [Feb 20 15:23:48] DEBUG[32544][C-00000002]: netsock2.c:172 ast_sockaddr_split_hostport: Splitting '192.168.1.147:23681' into... [Feb 20 15:23:48] DEBUG[32580][C-00000001]: res_rtp_asterisk.c:3393 ast_rtp_raw_write: Difference is 2976, ms is 82 [Feb 20 15:23:48] DEBUG[32544][C-00000002]: netsock2.c:226 ast_sockaddr_split_hostport: ...host '192.168.1.147' and port '23681'. Sending to 192.168.1.147:23681 (no NAT) [Feb 20 15:23:48] DEBUG[32544][C-00000002]: chan_sip.c:3409 sip_alreadygone: Setting SIP_ALREADYGONE on dialog AOK5~O9uyp [Feb 20 15:23:48] DEBUG[32581]: res_timing_timerfd.c:167 timerfd_timer_ack: Expected to acknowledge 1 ticks but got 2 instead [Feb 20 15:23:48] DEBUG[32584][C-00000002]: res_rtp_asterisk.c:3393 ast_rtp_raw_write: Difference is 3360, ms is 90 [Feb 20 15:23:48] DEBUG[32580][C-00000001]: res_rtp_asterisk.c:3393 ast_rtp_raw_write: Difference is 2304, ms is 68 [Feb 20 15:23:48] DEBUG[32584][C-00000002]: res_rtp_asterisk.c:3408 ast_rtp_raw_write: Difference is 13680, ms is 131 (11790), pred/ts/samples 56790/70470/0 [Feb 20 15:23:48] DEBUG[32581]: res_timing_timerfd.c:167 timerfd_timer_ack: Expected to acknowledge 1 ticks but got 3 instead [Feb 20 15:23:48] DEBUG[32581]: res_timing_timerfd.c:167 timerfd_timer_ack: Expected to acknowledge 1 ticks but got 2 instead [Feb 20 15:23:48] DEBUG[32584][C-00000002]: res_rtp_asterisk.c:3393 ast_rtp_raw_write: Difference is 3312, ms is 89 [Feb 20 15:23:48] DEBUG[32580][C-00000001]: res_rtp_asterisk.c:3393 ast_rtp_raw_write: Difference is 4272, ms is 109 [Feb 20 15:23:48] DEBUG[32581]: res_timing_timerfd.c:167 timerfd_timer_ack: Expected to acknowledge 1 ticks but got 2 instead [Feb 20 15:23:48] DEBUG[32581]: res_timing_timerfd.c:167 timerfd_timer_ack: Expected to acknowledge 1 ticks but got 2 instead [Feb 20 15:23:48] DEBUG[32580][C-00000001]: res_rtp_asterisk.c:3393 ast_rtp_raw_write: Difference is 1584, ms is 53 [Feb 20 15:23:48] DEBUG[32584][C-00000002]: res_rtp_asterisk.c:3393 ast_rtp_raw_write: Difference is 3792, ms is 99 [Feb 20 15:23:48] DEBUG[32581]: res_timing_timerfd.c:167 timerfd_timer_ack: Expected to acknowledge 1 ticks but got 3 instead [Feb 20 15:23:48] DEBUG[32584][C-00000002]: channel.c:5113 ast_write: Deadlock avoided for write to channel 'SIP/50002-00000002' [Feb 20 15:23:48] DEBUG[32581]: res_timing_timerfd.c:167 timerfd_timer_ack: Expected to acknowledge 1 ticks but got 3 instead [Feb 20 15:23:48] DEBUG[32580][C-00000001]: res_rtp_asterisk.c:3393 ast_rtp_raw_write: Difference is 3840, ms is 100 [Feb 20 15:23:48] DEBUG[32584][C-00000002]: res_rtp_asterisk.c:3393 ast_rtp_raw_write: Difference is 5424, ms is 133 [Feb 20 15:23:48] DEBUG[32581]: res_timing_timerfd.c:167 timerfd_timer_ack: Expected to acknowledge 1 ticks but got 2 instead [Feb 20 15:23:48] DEBUG[32580][C-00000001]: res_rtp_asterisk.c:3393 ast_rtp_raw_write: Difference is 3072, ms is 84 [Feb 20 15:23:48] DEBUG[32581]: res_timing_timerfd.c:167 timerfd_timer_ack: Expected to acknowledge 1 ticks but got 3 instead [Feb 20 15:23:48] DEBUG[32544][C-00000002]: res_rtp_asterisk.c:5036 ast_rtp_remote_address_set: Setting RTCP address on RTP instance '0x99a1948' [Feb 20 15:23:48] DEBUG[32544][C-00000002]: res_rtp_asterisk.c:5036 ast_rtp_remote_address_set: Setting RTCP address on RTP instance '0xa5db3e8' [Feb 20 15:23:48] DEBUG[32581]: res_timing_timerfd.c:167 timerfd_timer_ack: Expected to acknowledge 1 ticks but got 2 instead [Feb 20 15:23:48] DEBUG[32584][C-00000002]: res_rtp_asterisk.c:3543 ast_rtp_write: No remote address on RTP instance '0x99a1948' so dropping frame [Feb 20 15:23:48] DEBUG[32580][C-00000001]: res_rtp_asterisk.c:3393 ast_rtp_raw_write: Difference is 3312, ms is 89 [Feb 20 15:23:48] DEBUG[32584][C-00000002]: res_rtp_asterisk.c:3543 ast_rtp_write: No remote address on RTP instance '0x99a1948' so dropping frame [Feb 20 15:23:48] DEBUG[32584][C-00000002]: res_rtp_asterisk.c:3543 ast_rtp_write: No remote address on RTP instance '0x99a1948' so dropping frame [Feb 20 15:23:48] DEBUG[32584][C-00000002]: res_rtp_asterisk.c:3543 ast_rtp_write: No remote address on RTP instance '0x99a1948' so dropping frame [Feb 20 15:23:48] DEBUG[32581]: res_timing_timerfd.c:167 timerfd_timer_ack: Expected to acknowledge 1 ticks but got 3 instead [Feb 20 15:23:48] DEBUG[32584][C-00000002]: res_rtp_asterisk.c:3543 ast_rtp_write: No remote address on RTP instance '0x99a1948' so dropping frame [Feb 20 15:23:48] DEBUG[32580][C-00000001]: res_rtp_asterisk.c:3393 ast_rtp_raw_write: Difference is 2688, ms is 76 [Feb 20 15:23:48] DEBUG[32581]: res_timing_timerfd.c:167 timerfd_timer_ack: Expected to acknowledge 1 ticks but got 4 instead Scheduling destruction of SIP dialog 'AOK5~O9uyp' in 32000 ms (Method: BYE) [Feb 20 15:23:48] DEBUG[32544][C-00000002]: chan_sip.c:27259 handle_request_bye: Received bye, issuing owner hangup [Feb 20 15:23:48] DEBUG[32584][C-00000002]: channel.c:5113 ast_write: Deadlock avoided for write to channel 'SIP/50002-00000002' <--- Transmitting (no NAT) to 192.168.1.147:23681 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.147:23681;branch=z9hG4bK.rUwOPluXX;received=192.168.1.147;rport=23681 From: ;tag=X3ughL6Dw To: ;tag=as188ae036 Call-ID: AOK5~O9uyp CSeq: 22 BYE Server: Asterisk PBX 13.14.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 <------------> [Feb 20 15:23:48] DEBUG[32544][C-00000002]: chan_sip.c:3753 __sip_xmit: Trying to put 'SIP/2.0 200' onto UDP socket destined for 192.168.1.147:23681 [Feb 20 15:23:48] DEBUG[32581]: res_timing_timerfd.c:167 timerfd_timer_ack: Expected to acknowledge 1 ticks but got 2 instead [Feb 20 15:23:48] DEBUG[32580][C-00000001]: res_rtp_asterisk.c:3393 ast_rtp_raw_write: Difference is 3936, ms is 102 [Feb 20 15:23:48] DEBUG[32584][C-00000002]: bridge_channel.c:288 ast_bridge_channel_leave_bridge_nolock: Setting 0xabebb28(SIP/50002-00000002) state from:0 to:1 <--- SIP read from UDP:192.168.1.147:23681 ---> BYE sip:350001@192.168.1.227:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.147:23681;branch=z9hG4bK.rUwOPluXX;rport From: ;tag=X3ughL6Dw To: ;tag=as188ae036 CSeq: 22 BYE Call-ID: AOK5~O9uyp Max-Forwards: 70 User-Agent: Linphone/3.10.2 (belle-sip/1.5.0) Authorization: Digest realm="asterisk", nonce="2d0b2d61", algorithm=MD5, username="50002", uri="sip:350001@192.168.1.227:5060", response="fb15e343bca87c65212124ec31f60475" <-------------> [Feb 20 15:23:48] DEBUG[32544]: chan_sip.c:9893 parse_request: Header 0 [ 42]: BYE sip:350001@192.168.1.227:5060 SIP/2.0 [Feb 20 15:23:48] DEBUG[32544]: chan_sip.c:9893 parse_request: Header 1 [ 69]: Via: SIP/2.0/UDP 192.168.1.147:23681;branch=z9hG4bK.rUwOPluXX;rport [Feb 20 15:23:48] DEBUG[32544]: chan_sip.c:9893 parse_request: Header 2 [ 46]: From: ;tag=X3ughL6Dw [Feb 20 15:23:48] DEBUG[32544]: chan_sip.c:9893 parse_request: Header 3 [ 46]: To: ;tag=as188ae036 [Feb 20 15:23:49] DEBUG[32544]: chan_sip.c:9893 parse_request: Header 4 [ 12]: CSeq: 22 BYE [Feb 20 15:23:49] DEBUG[32544]: chan_sip.c:9893 parse_request: Header 5 [ 19]: Call-ID: AOK5~O9uyp [Feb 20 15:23:49] DEBUG[32544]: chan_sip.c:9893 parse_request: Header 6 [ 16]: Max-Forwards: 70 [Feb 20 15:23:49] DEBUG[32544]: chan_sip.c:9893 parse_request: Header 7 [ 45]: User-Agent: Linphone/3.10.2 (belle-sip/1.5.0) [Feb 20 15:23:49] DEBUG[32544]: chan_sip.c:9893 parse_request: Header 8 [172]: Authorization: Digest realm="asterisk", nonce="2d0b2d61", algorithm=MD5, username="50002", uri="sip:350001@192.168.1.227:5060", response="fb15e343bca87c65212124ec31f60475" --- (9 headers 0 lines) --- [Feb 20 15:23:49] DEBUG[32544]: chan_sip.c:9419 __find_call: = Looking for Call ID: AOK5~O9uyp (Checking From) --From tag X3ughL6Dw --To-tag as188ae036 [Feb 20 15:23:49] DEBUG[32580][C-00000001]: res_rtp_asterisk.c:3393 ast_rtp_raw_write: Difference is 3312, ms is 89 [Feb 20 15:23:49] DEBUG[32581]: res_timing_timerfd.c:167 timerfd_timer_ack: Expected to acknowledge 1 ticks but got 8 instead [Feb 20 15:23:49] DEBUG[32544][C-00000002]: chan_sip.c:28652 handle_incoming: **** Received BYE (8) - Command in SIP BYE [Feb 20 15:23:49] DEBUG[32584][C-00000002]: bridge_channel.c:2055 bridge_channel_internal_pull: Bridge 28f1b4ee-63be-49cb-b551-d5a7209f7781: pulling 0xabebb28(SIP/50002-00000002) [Feb 20 15:23:49] DEBUG[32544][C-00000002]: chan_sip.c:28677 handle_incoming: Ignoring SIP message because of retransmit (BYE Seqno 22, ours 22) [Feb 20 15:23:49] DEBUG[32544][C-00000002]: chan_sip.c:27096 handle_request_bye: Initializing initreq for method BYE - callid AOK5~O9uyp [Feb 20 15:23:49] DEBUG[32544][C-00000002]: netsock2.c:172 ast_sockaddr_split_hostport: Splitting '192.168.1.147:23681' into... -- Channel SIP/50002-00000002 left 'softmix' base-bridge <28f1b4ee-63be-49cb-b551-d5a7209f7781> [Feb 20 15:23:49] DEBUG[32584][C-00000002]: bridge_channel.c:2067 bridge_channel_internal_pull: Bridge 28f1b4ee-63be-49cb-b551-d5a7209f7781: 0xabebb28(SIP/50002-00000002) is leaving softmix technology [Feb 20 15:23:49] DEBUG[32544][C-00000002]: netsock2.c:226 ast_sockaddr_split_hostport: ...host '192.168.1.147' and port '23681'. Sending to 192.168.1.147:23681 (no NAT) [Feb 20 15:23:49] DEBUG[32580][C-00000001]: res_rtp_asterisk.c:3393 ast_rtp_raw_write: Difference is 1920, ms is 60 [Feb 20 15:23:49] DEBUG[32544][C-00000002]: chan_sip.c:3409 sip_alreadygone: Setting SIP_ALREADYGONE on dialog AOK5~O9uyp [Feb 20 15:23:49] DEBUG[32496]: cdr.c:1293 cdr_object_finalize: Finalized CDR for SIP/50001-00000001 - start 1487575415.698644 answer 1487575416.073465 end 1487575429.176213 dispo ANSWERED [Feb 20 15:23:49] DEBUG[32580][C-00000001]: res_rtp_asterisk.c:3393 ast_rtp_raw_write: Difference is 1872, ms is 59 [Feb 20 15:23:49] DEBUG[32580][C-00000001]: res_rtp_asterisk.c:3393 ast_rtp_raw_write: Difference is 1104, ms is 43 [Feb 20 15:23:49] DEBUG[32580][C-00000001]: res_rtp_asterisk.c:3393 ast_rtp_raw_write: Difference is 1392, ms is 49 [Feb 20 15:23:49] DEBUG[32580][C-00000001]: res_rtp_asterisk.c:3393 ast_rtp_raw_write: Difference is 5136, ms is 127 [Feb 20 15:23:49] DEBUG[32581]: res_timing_timerfd.c:167 timerfd_timer_ack: Expected to acknowledge 1 ticks but got 15 instead [Feb 20 15:23:49] DEBUG[32584][C-00000002]: bridge_channel.c:336 ast_bridge_channel_restore_formats: Bridge is returning 0xabebb28(SIP/50002-00000002) to read format opus [Feb 20 15:23:49] DEBUG[32584][C-00000002]: channel.c:5680 set_format: Channel SIP/50002-00000002 setting read format path: opus -> opus [Feb 20 15:23:49] DEBUG[32584][C-00000002]: bridge_channel.c:346 ast_bridge_channel_restore_formats: Bridge is returning 0xabebb28(SIP/50002-00000002) to write format opus [Feb 20 15:23:49] DEBUG[32580][C-00000001]: res_rtp_asterisk.c:3393 ast_rtp_raw_write: Difference is 3984, ms is 103 [Feb 20 15:23:49] DEBUG[32584][C-00000002]: channel.c:5680 set_format: Channel SIP/50002-00000002 setting write format path: opus -> opus [Feb 20 15:23:49] DEBUG[32580][C-00000001]: res_rtp_asterisk.c:3393 ast_rtp_raw_write: Difference is 3840, ms is 100 [Feb 20 15:23:49] DEBUG[32580][C-00000001]: res_rtp_asterisk.c:3393 ast_rtp_raw_write: Difference is 3792, ms is 99 [Feb 20 15:23:49] DEBUG[32580][C-00000001]: res_rtp_asterisk.c:3393 ast_rtp_raw_write: Difference is 3936, ms is 102 [Feb 20 15:23:49] DEBUG[32584][C-00000002]: confbridge/conf_state.c:84 conf_change_state: Changing conference '350001' state from MULTI_MARKED to SINGLE_MARKED [Feb 20 15:23:49] DEBUG[32584][C-00000002]: app_confbridge.c:1145 conf_update_user_mute: User SIP/50001-00000001 is unmuted: user:0 system:0. [Feb 20 15:23:49] DEBUG[32582]: channel.c:5680 set_format: Channel CBAnn/350001-00000001;1 setting write format path: gsm -> slin [Feb 20 15:23:49] DEBUG[32584][C-00000002]: pbx.c:4321 __ast_pbx_run: Extension 350001, priority 6 returned normally even though call was hung up [Feb 20 15:23:49] DEBUG[32582]: channel.c:3469 ast_settimeout_full: Scheduling timer at (50 requested / 50 actual) timer ticks per second [Feb 20 15:23:49] DEBUG[32544][C-00000002]: res_rtp_asterisk.c:5036 ast_rtp_remote_address_set: Setting RTCP address on RTP instance '0x99a1948' [Feb 20 15:23:49] DEBUG[32580][C-00000001]: res_rtp_asterisk.c:3393 ast_rtp_raw_write: Difference is 6096, ms is 147 [Feb 20 15:23:49] DEBUG[32544][C-00000002]: res_rtp_asterisk.c:5036 ast_rtp_remote_address_set: Setting RTCP address on RTP instance '0xa5db3e8' -- Playing 'confbridge-leave.gsm' (language 'en') [Feb 20 15:23:50] DEBUG[32544][C-00000002]: chan_sip.c:27259 handle_request_bye: Received bye, issuing owner hangup <--- Transmitting (no NAT) to 192.168.1.147:23681 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.147:23681;branch=z9hG4bK.rUwOPluXX;received=192.168.1.147;rport=23681 From: ;tag=X3ughL6Dw To: ;tag=as188ae036 Call-ID: AOK5~O9uyp CSeq: 22 BYE Server: Asterisk PBX 13.14.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 <------------> [Feb 20 15:23:50] DEBUG[32544][C-00000002]: chan_sip.c:3753 __sip_xmit: Trying to put 'SIP/2.0 200' onto UDP socket destined for 192.168.1.147:23681 [Feb 20 15:23:50] DEBUG[32584][C-00000002]: channel.c:2579 ast_softhangup_nolock: Soft-Hanging (0x10) up channel 'SIP/50002-00000002' [Feb 20 15:23:50] DEBUG[32582]: res_timing_timerfd.c:167 timerfd_timer_ack: Expected to acknowledge 1 ticks but got 3 instead [Feb 20 15:23:50] DEBUG[32584][C-00000002]: pbx_lua.c:1480 lua_find_extension: Looking up h@default:1 [Feb 20 15:23:50] DEBUG[32584][C-00000002]: pbx_lua.c:1480 lua_find_extension: Looking up h@demo:1 [Feb 20 15:23:50] DEBUG[32582]: res_timing_timerfd.c:167 timerfd_timer_ack: Expected to acknowledge 1 ticks but got 3 instead [Feb 20 15:23:50] DEBUG[32584][C-00000002]: channel.c:2728 ast_hangup: Hanging up channel 'SIP/50002-00000002' [Feb 20 15:23:50] DEBUG[32584][C-00000002]: chan_sip.c:7154 sip_hangup: Hangup call SIP/50002-00000002, SIP callid AOK5~O9uyp [Feb 20 15:23:50] DEBUG[32580][C-00000001]: res_rtp_asterisk.c:3393 ast_rtp_raw_write: Difference is 6144, ms is 148 [Feb 20 15:23:50] DEBUG[32582]: res_timing_timerfd.c:167 timerfd_timer_ack: Expected to acknowledge 1 ticks but got 2 instead [Feb 20 15:23:50] DEBUG[32584][C-00000002]: res_rtp_asterisk.c:5036 ast_rtp_remote_address_set: Setting RTCP address on RTP instance '0x99a1948' [Feb 20 15:23:50] DEBUG[32584][C-00000002]: res_rtp_asterisk.c:5036 ast_rtp_remote_address_set: Setting RTCP address on RTP instance '0xa5db3e8' [Feb 20 15:23:50] DEBUG[32582]: res_timing_timerfd.c:167 timerfd_timer_ack: Expected to acknowledge 1 ticks but got 2 instead [Feb 20 15:23:50] DEBUG[32580][C-00000001]: res_rtp_asterisk.c:3393 ast_rtp_raw_write: Difference is 5472, ms is 134 [Feb 20 15:23:50] DEBUG[32582]: res_timing_timerfd.c:167 timerfd_timer_ack: Expected to acknowledge 1 ticks but got 3 instead [Feb 20 15:23:50] DEBUG[32582]: res_timing_timerfd.c:167 timerfd_timer_ack: Expected to acknowledge 1 ticks but got 2 instead [Feb 20 15:23:50] DEBUG[32581]: res_timing_timerfd.c:167 timerfd_timer_ack: Expected to acknowledge 1 ticks but got 2 instead [Feb 20 15:23:50] DEBUG[32580][C-00000001]: res_rtp_asterisk.c:3393 ast_rtp_raw_write: Difference is 4032, ms is 104 [Feb 20 15:23:50] DEBUG[32582]: res_timing_timerfd.c:167 timerfd_timer_ack: Expected to acknowledge 1 ticks but got 2 instead [Feb 20 15:23:50] DEBUG[32581]: res_timing_timerfd.c:167 timerfd_timer_ack: Expected to acknowledge 1 ticks but got 2 instead <--- SIP read from UDP:192.168.1.147:43521 ---> BYE sip:350001@192.168.1.227:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.147:43521;branch=z9hG4bK.fUFPtiw3U;rport From: ;tag=cdYXYwnFg To: ;tag=as413e790c CSeq: 22 BYE Call-ID: dV~eC~AMO2 Max-Forwards: 70 User-Agent: Linphone/3.9.1 (belle-sip/1.4.2) Authorization: Digest realm="asterisk", nonce="2bc3b02a", algorithm=MD5, username="50001", uri="sip:350001@192.168.1.227:5060", response="e051529e67800e40ff80885613f9469a" <-------------> [Feb 20 15:23:50] DEBUG[32544]: chan_sip.c:9893 parse_request: Header 0 [ 42]: BYE sip:350001@192.168.1.227:5060 SIP/2.0 [Feb 20 15:23:50] DEBUG[32544]: chan_sip.c:9893 parse_request: Header 1 [ 69]: Via: SIP/2.0/UDP 192.168.1.147:43521;branch=z9hG4bK.fUFPtiw3U;rport [Feb 20 15:23:50] DEBUG[32544]: chan_sip.c:9893 parse_request: Header 2 [ 46]: From: ;tag=cdYXYwnFg [Feb 20 15:23:50] DEBUG[32495]: devicestate.c:369 _ast_device_state: No provider found, checking channel drivers for SIP - 50002 [Feb 20 15:23:50] DEBUG[32495]: chan_sip.c:30178 sip_devicestate: Checking device state for peer 50002 [Feb 20 15:23:50] DEBUG[32544]: chan_sip.c:9893 parse_request: Header 3 [ 46]: To: ;tag=as413e790c [Feb 20 15:23:50] DEBUG[32544]: chan_sip.c:9893 parse_request: Header 4 [ 12]: CSeq: 22 BYE [Feb 20 15:23:50] DEBUG[32582]: res_timing_timerfd.c:167 timerfd_timer_ack: Expected to acknowledge 1 ticks but got 4 instead [Feb 20 15:23:50] DEBUG[32581]: res_timing_timerfd.c:167 timerfd_timer_ack: Expected to acknowledge 1 ticks but got 2 instead [Feb 20 15:23:50] DEBUG[32495]: devicestate.c:474 do_state_change: Changing state for SIP/50002 - state 1 (Not in use) [Feb 20 15:23:50] DEBUG[32496]: cdr_radius.c:222 radius_log: Unable to create RADIUS record. CDR not recorded! [Feb 20 15:23:50] DEBUG[32544]: chan_sip.c:9893 parse_request: Header 5 [ 19]: Call-ID: dV~eC~AMO2 [Feb 20 15:23:50] DEBUG[32544]: chan_sip.c:9893 parse_request: Header 6 [ 16]: Max-Forwards: 70 [Feb 20 15:23:50] DEBUG[32544]: chan_sip.c:9893 parse_request: Header 7 [ 44]: User-Agent: Linphone/3.9.1 (belle-sip/1.4.2) [Feb 20 15:23:50] DEBUG[32580][C-00000001]: res_rtp_asterisk.c:3393 ast_rtp_raw_write: Difference is 5856, ms is 142 [Feb 20 15:23:50] DEBUG[32544]: chan_sip.c:9893 parse_request: Header 8 [172]: Authorization: Digest realm="asterisk", nonce="2bc3b02a", algorithm=MD5, username="50001", uri="sip:350001@192.168.1.227:5060", response="e051529e67800e40ff80885613f9469a" --- (9 headers 0 lines) --- [Feb 20 15:23:50] DEBUG[32544]: chan_sip.c:9419 __find_call: = Looking for Call ID: dV~eC~AMO2 (Checking From) --From tag cdYXYwnFg --To-tag as413e790c [Feb 20 15:23:50] DEBUG[32582]: res_timing_timerfd.c:167 timerfd_timer_ack: Expected to acknowledge 1 ticks but got 3 instead [Feb 20 15:23:50] DEBUG[32496]: res_config_sqlite.c:827 cdr_handler: SQL query: INSERT INTO ast_cdr (clid,src,dst,dcontext,channel,lastapp,lastdata,start,answer,end,duration,billsec,disposition,amaflags,uniqueid) VALUES ('"" <50002>','50002','350001','default','SIP/50002-00000002','ConfBridge','350001,default_bridge_32,default_user,sample_user_menu','2017-02-20 15:23:39','2017-02-20 15:23:40','2017-02-20 15:23:45','6','5','ANSWERED','DOCUMENTATION','1487575419.7') [Feb 20 15:23:50] DEBUG[32582]: res_timing_timerfd.c:167 timerfd_timer_ack: Expected to acknowledge 1 ticks but got 2 instead [Feb 20 15:23:50] DEBUG[32582]: res_timing_timerfd.c:167 timerfd_timer_ack: Expected to acknowledge 1 ticks but got 2 instead [Feb 20 15:23:50] DEBUG[32580][C-00000001]: res_rtp_asterisk.c:3393 ast_rtp_raw_write: Difference is 4416, ms is 112 [Feb 20 15:23:50] DEBUG[32544][C-00000001]: chan_sip.c:28652 handle_incoming: **** Received BYE (8) - Command in SIP BYE [Feb 20 15:23:50] DEBUG[32544][C-00000001]: chan_sip.c:27096 handle_request_bye: Initializing initreq for method BYE - callid dV~eC~AMO2 [Feb 20 15:23:50] DEBUG[32544][C-00000001]: netsock2.c:172 ast_sockaddr_split_hostport: Splitting '192.168.1.147:43521' into... [Feb 20 15:23:50] DEBUG[32580][C-00000001]: channel.c:5113 ast_write: Deadlock avoided for write to channel 'SIP/50001-00000001' [Feb 20 15:23:50] DEBUG[32544][C-00000001]: netsock2.c:226 ast_sockaddr_split_hostport: ...host '192.168.1.147' and port '43521'. Sending to 192.168.1.147:43521 (no NAT) [Feb 20 15:23:50] DEBUG[32544][C-00000001]: chan_sip.c:3409 sip_alreadygone: Setting SIP_ALREADYGONE on dialog dV~eC~AMO2 [Feb 20 15:23:50] DEBUG[32582]: res_timing_timerfd.c:167 timerfd_timer_ack: Expected to acknowledge 1 ticks but got 3 instead [Feb 20 15:23:50] DEBUG[32580][C-00000001]: res_rtp_asterisk.c:3393 ast_rtp_raw_write: Difference is 5088, ms is 126 [Feb 20 15:23:50] DEBUG[32582]: res_timing_timerfd.c:167 timerfd_timer_ack: Expected to acknowledge 1 ticks but got 2 instead [Feb 20 15:23:50] DEBUG[32582]: channel.c:3469 ast_settimeout_full: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Feb 20 15:23:50] DEBUG[32582]: channel.c:3469 ast_settimeout_full: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Feb 20 15:23:50] DEBUG[32582]: channel.c:3469 ast_settimeout_full: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Feb 20 15:23:50] DEBUG[32582]: channel.c:5680 set_format: Channel CBAnn/350001-00000001;1 setting write format path: slin -> slin [Feb 20 15:23:50] DEBUG[32580][C-00000001]: res_rtp_asterisk.c:3393 ast_rtp_raw_write: Difference is 3072, ms is 84 [Feb 20 15:23:50] DEBUG[32580][C-00000001]: res_rtp_asterisk.c:3393 ast_rtp_raw_write: Difference is 2304, ms is 68 [Feb 20 15:23:51] DEBUG[32580][C-00000001]: res_rtp_asterisk.c:3393 ast_rtp_raw_write: Difference is 4608, ms is 116 [Feb 20 15:23:51] DEBUG[32580][C-00000001]: channel.c:5113 ast_write: Deadlock avoided for write to channel 'SIP/50001-00000001' [Feb 20 15:23:51] DEBUG[32544][C-00000001]: res_rtp_asterisk.c:5036 ast_rtp_remote_address_set: Setting RTCP address on RTP instance '0xa02e108' [Feb 20 15:23:51] DEBUG[32544][C-00000001]: res_rtp_asterisk.c:5036 ast_rtp_remote_address_set: Setting RTCP address on RTP instance '0x9ee14c8' [Feb 20 15:23:51] DEBUG[32581]: res_timing_timerfd.c:167 timerfd_timer_ack: Expected to acknowledge 1 ticks but got 3 instead Scheduling destruction of SIP dialog 'dV~eC~AMO2' in 32000 ms (Method: BYE) [Feb 20 15:23:51] DEBUG[32544][C-00000001]: chan_sip.c:27259 handle_request_bye: Received bye, issuing owner hangup <--- Transmitting (no NAT) to 192.168.1.147:43521 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.147:43521;branch=z9hG4bK.fUFPtiw3U;received=192.168.1.147;rport=43521 From: ;tag=cdYXYwnFg To: ;tag=as413e790c Call-ID: dV~eC~AMO2 CSeq: 22 BYE Server: Asterisk PBX 13.14.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 <------------> [Feb 20 15:23:51] DEBUG[32544][C-00000001]: chan_sip.c:3753 __sip_xmit: Trying to put 'SIP/2.0 200' onto UDP socket destined for 192.168.1.147:43521 <--- SIP read from UDP:192.168.1.147:43521 ---> BYE sip:350001@192.168.1.227:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.1.147:43521;branch=z9hG4bK.fUFPtiw3U;rport From: ;tag=cdYXYwnFg To: ;tag=as413e790c CSeq: 22 BYE Call-ID: dV~eC~AMO2 Max-Forwards: 70 User-Agent: Linphone/3.9.1 (belle-sip/1.4.2) Authorization: Digest realm="asterisk", nonce="2bc3b02a", algorithm=MD5, username="50001", uri="sip:350001@192.168.1.227:5060", response="e051529e67800e40ff80885613f9469a" <-------------> [Feb 20 15:23:51] DEBUG[32544]: chan_sip.c:9893 parse_request: Header 0 [ 42]: BYE sip:350001@192.168.1.227:5060 SIP/2.0 [Feb 20 15:23:51] DEBUG[32544]: chan_sip.c:9893 parse_request: Header 1 [ 69]: Via: SIP/2.0/UDP 192.168.1.147:43521;branch=z9hG4bK.fUFPtiw3U;rport [Feb 20 15:23:51] DEBUG[32544]: chan_sip.c:9893 parse_request: Header 2 [ 46]: From: ;tag=cdYXYwnFg [Feb 20 15:23:51] DEBUG[32544]: chan_sip.c:9893 parse_request: Header 3 [ 46]: To: ;tag=as413e790c [Feb 20 15:23:51] DEBUG[32544]: chan_sip.c:9893 parse_request: Header 4 [ 12]: CSeq: 22 BYE [Feb 20 15:23:51] DEBUG[32544]: chan_sip.c:9893 parse_request: Header 5 [ 19]: Call-ID: dV~eC~AMO2 [Feb 20 15:23:51] DEBUG[32544]: chan_sip.c:9893 parse_request: Header 6 [ 16]: Max-Forwards: 70 [Feb 20 15:23:51] DEBUG[32544]: chan_sip.c:9893 parse_request: Header 7 [ 44]: User-Agent: Linphone/3.9.1 (belle-sip/1.4.2) [Feb 20 15:23:51] DEBUG[32544]: chan_sip.c:9893 parse_request: Header 8 [172]: Authorization: Digest realm="asterisk", nonce="2bc3b02a", algorithm=MD5, username="50001", uri="sip:350001@192.168.1.227:5060", response="e051529e67800e40ff80885613f9469a" --- (9 headers 0 lines) --- [Feb 20 15:23:51] DEBUG[32544]: chan_sip.c:9419 __find_call: = Looking for Call ID: dV~eC~AMO2 (Checking From) --From tag cdYXYwnFg --To-tag as413e790c [Feb 20 15:23:51] DEBUG[32544][C-00000001]: chan_sip.c:28652 handle_incoming: **** Received BYE (8) - Command in SIP BYE [Feb 20 15:23:51] DEBUG[32544][C-00000001]: chan_sip.c:28677 handle_incoming: Ignoring SIP message because of retransmit (BYE Seqno 22, ours 22) [Feb 20 15:23:51] DEBUG[32544][C-00000001]: chan_sip.c:27096 handle_request_bye: Initializing initreq for method BYE - callid dV~eC~AMO2 [Feb 20 15:23:51] DEBUG[32544][C-00000001]: netsock2.c:172 ast_sockaddr_split_hostport: Splitting '192.168.1.147:43521' into... [Feb 20 15:23:51] DEBUG[32544][C-00000001]: netsock2.c:226 ast_sockaddr_split_hostport: ...host '192.168.1.147' and port '43521'. Sending to 192.168.1.147:43521 (no NAT) [Feb 20 15:23:51] DEBUG[32544][C-00000001]: chan_sip.c:3409 sip_alreadygone: Setting SIP_ALREADYGONE on dialog dV~eC~AMO2 [Feb 20 15:23:51] DEBUG[32580][C-00000001]: channel.c:5113 ast_write: Deadlock avoided for write to channel 'SIP/50001-00000001' [Feb 20 15:23:51] DEBUG[32544][C-00000001]: res_rtp_asterisk.c:5036 ast_rtp_remote_address_set: Setting RTCP address on RTP instance '0xa02e108' [Feb 20 15:23:51] DEBUG[32544][C-00000001]: res_rtp_asterisk.c:5036 ast_rtp_remote_address_set: Setting RTCP address on RTP instance '0x9ee14c8' [Feb 20 15:23:51] DEBUG[32544][C-00000001]: chan_sip.c:27259 handle_request_bye: Received bye, issuing owner hangup <--- Transmitting (no NAT) to 192.168.1.147:43521 ---> SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.1.147:43521;branch=z9hG4bK.fUFPtiw3U;received=192.168.1.147;rport=43521 From: ;tag=cdYXYwnFg To: ;tag=as413e790c Call-ID: dV~eC~AMO2 CSeq: 22 BYE Server: Asterisk PBX 13.14.0 Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces, timer Content-Length: 0 <------------> [Feb 20 15:23:51] DEBUG[32544][C-00000001]: chan_sip.c:3753 __sip_xmit: Trying to put 'SIP/2.0 200' onto UDP socket destined for 192.168.1.147:43521 <--- SIP read from UDP:192.168.1.147:23681 ---> <-------------> [Feb 20 15:23:51] DEBUG[32544]: chan_sip.c:9893 parse_request: Header 0 [ 0]: [Feb 20 15:23:51] DEBUG[32544]: chan_sip.c:4361 __sip_autodestruct: Auto destroying SIP dialog 'RFZ-H-NZHO' [Feb 20 15:23:51] DEBUG[32544]: chan_sip.c:6589 sip_pvt_dtor: Destroying SIP dialog RFZ-H-NZHO Really destroying SIP dialog 'RFZ-H-NZHO' Method: BYE [Feb 20 15:23:51] DEBUG[32544]: rtp_engine.c:397 instance_destructor: Destroyed RTP instance '0x9405778' [Feb 20 15:23:51] DEBUG[32544]: rtp_engine.c:397 instance_destructor: Destroyed RTP instance '0x94c4028' [Feb 20 15:23:54] DEBUG[32580][C-00000001]: bridge_channel.c:288 ast_bridge_channel_leave_bridge_nolock: Setting 0xa01f998(SIP/50001-00000001) state from:0 to:1 [Feb 20 15:23:54] DEBUG[32581]: res_timing_timerfd.c:167 timerfd_timer_ack: Expected to acknowledge 1 ticks but got 4 instead [Feb 20 15:23:54] DEBUG[32580][C-00000001]: bridge_channel.c:2055 bridge_channel_internal_pull: Bridge 28f1b4ee-63be-49cb-b551-d5a7209f7781: pulling 0xa01f998(SIP/50001-00000001) -- Channel SIP/50001-00000001 left 'softmix' base-bridge <28f1b4ee-63be-49cb-b551-d5a7209f7781> [Feb 20 15:23:54] DEBUG[32580][C-00000001]: bridge_channel.c:2067 bridge_channel_internal_pull: Bridge 28f1b4ee-63be-49cb-b551-d5a7209f7781: 0xa01f998(SIP/50001-00000001) is leaving softmix technology [Feb 20 15:23:54] DEBUG[32581]: res_timing_timerfd.c:167 timerfd_timer_ack: Expected to acknowledge 1 ticks but got 2 instead [Feb 20 15:23:54] DEBUG[32580][C-00000001]: bridge_channel.c:336 ast_bridge_channel_restore_formats: Bridge is returning 0xa01f998(SIP/50001-00000001) to read format opus [Feb 20 15:23:54] DEBUG[32580][C-00000001]: channel.c:5680 set_format: Channel SIP/50001-00000001 setting read format path: opus -> opus [Feb 20 15:23:54] DEBUG[32580][C-00000001]: bridge_channel.c:346 ast_bridge_channel_restore_formats: Bridge is returning 0xa01f998(SIP/50001-00000001) to write format opus [Feb 20 15:23:55] DEBUG[32580][C-00000001]: channel.c:5680 set_format: Channel SIP/50001-00000001 setting write format path: opus -> opus [Feb 20 15:23:55] DEBUG[32580][C-00000001]: confbridge/conf_state.c:84 conf_change_state: Changing conference '350001' state from SINGLE_MARKED to EMPTY [Feb 20 15:23:55] DEBUG[32582]: channel.c:5680 set_format: Channel CBAnn/350001-00000001;1 setting write format path: gsm -> slin [Feb 20 15:23:55] DEBUG[32566]: app_queue.c:2482 device_state_cb: Device 'confbridge:350001' changed to state '1' (Not in use) but we don't care because they're not a member of any queue. [Feb 20 15:23:55] DEBUG[32582]: channel.c:3469 ast_settimeout_full: Scheduling timer at (50 requested / 50 actual) timer ticks per second -- Playing 'confbridge-leave.gsm' (language 'en') [Feb 20 15:23:55] DEBUG[32580][C-00000001]: app_confbridge.c:1027 destroy_conference_bridge: Destroying conference bridge '350001' [Feb 20 15:23:55] DEBUG[32581]: bridge_softmix.c:866 analyse_softmix_stats: All below internal rate. Bridge changed from 48000 to 8000. [Feb 20 15:23:55] DEBUG[32581]: channel.c:5444 ast_set_read_format_path: Channel CBAnn/350001-00000001;2 setting read format path: slin -> slin [Feb 20 15:23:55] DEBUG[32581]: channel.c:5680 set_format: Channel CBAnn/350001-00000001;2 setting write format path: slin -> slin [Feb 20 15:23:55] DEBUG[32583]: channel.c:5264 ast_write: Channel CBAnn/350001-00000001;2 changing write format from slin to slin48, native formats (slin) [Feb 20 15:23:55] DEBUG[32581]: dsp.c:499 ast_tone_detect_init: Setup tone 1100 Hz, 500 ms, block_size=160, hits_required=21 [Feb 20 15:23:55] DEBUG[32581]: dsp.c:499 ast_tone_detect_init: Setup tone 2100 Hz, 2600 ms, block_size=160, hits_required=116 [Feb 20 15:23:55] DEBUG[32581]: res_timing_timerfd.c:167 timerfd_timer_ack: Expected to acknowledge 1 ticks but got 2 instead [Feb 20 15:23:55] DEBUG[32583]: channel.c:5680 set_format: Channel CBAnn/350001-00000001;2 setting write format path: slin48 -> slin [Feb 20 15:23:55] DEBUG[32582]: res_timing_timerfd.c:167 timerfd_timer_ack: Expected to acknowledge 1 ticks but got 2 instead [Feb 20 15:23:55] DEBUG[32582]: channel.c:3469 ast_settimeout_full: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Feb 20 15:23:55] DEBUG[32582]: channel.c:3469 ast_settimeout_full: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Feb 20 15:23:55] DEBUG[32582]: channel.c:3469 ast_settimeout_full: Scheduling timer at (0 requested / 0 actual) timer ticks per second [Feb 20 15:23:55] DEBUG[32582]: channel.c:5680 set_format: Channel CBAnn/350001-00000001;1 setting write format path: slin -> slin [Feb 20 15:23:55] DEBUG[32582]: channel.c:2728 ast_hangup: Hanging up channel 'CBAnn/350001-00000001;1' [Feb 20 15:23:55] DEBUG[32495]: devicestate.c:369 _ast_device_state: No provider found, checking channel drivers for CBAnn - 350001 [Feb 20 15:23:55] DEBUG[32580][C-00000001]: bridge.c:960 ast_bridge_destroy: Bridge 28f1b4ee-63be-49cb-b551-d5a7209f7781: telling all channels to leave the party [Feb 20 15:23:55] DEBUG[32495]: devicestate.c:474 do_state_change: Changing state for CBAnn/350001 - state 2 (In use) [Feb 20 15:23:55] DEBUG[32580][C-00000001]: bridge.c:322 bridge_dissolve: Bridge 28f1b4ee-63be-49cb-b551-d5a7209f7781: dissolving bridge with cause 16(Normal Clearing) [Feb 20 15:23:55] DEBUG[32566]: app_queue.c:2482 device_state_cb: Device 'CBAnn/350001' changed to state '2' (In use) but we don't care because they're not a member of any queue. [Feb 20 15:23:55] DEBUG[32580][C-00000001]: bridge_channel.c:288 ast_bridge_channel_leave_bridge_nolock: Setting 0x9335418(CBAnn/350001-00000001;2) state from:0 to:2 [Feb 20 15:23:55] DEBUG[32580][C-00000001]: bridge.c:283 bridge_queue_action_nodup: Bridge 28f1b4ee-63be-49cb-b551-d5a7209f7781: queueing action type:13 sub:1001 [Feb 20 15:23:55] DEBUG[32583]: bridge_channel.c:2055 bridge_channel_internal_pull: Bridge 28f1b4ee-63be-49cb-b551-d5a7209f7781: pulling 0x9335418(CBAnn/350001-00000001;2) -- Channel CBAnn/350001-00000001;2 left 'softmix' base-bridge <28f1b4ee-63be-49cb-b551-d5a7209f7781> [Feb 20 15:23:55] DEBUG[32583]: bridge_channel.c:2067 bridge_channel_internal_pull: Bridge 28f1b4ee-63be-49cb-b551-d5a7209f7781: 0x9335418(CBAnn/350001-00000001;2) is leaving softmix technology [Feb 20 15:23:55] DEBUG[32580][C-00000001]: pbx.c:4321 __ast_pbx_run: Extension 350001, priority 8 returned normally even though call was hung up [Feb 20 15:23:55] DEBUG[32580][C-00000001]: channel.c:2579 ast_softhangup_nolock: Soft-Hanging (0x10) up channel 'SIP/50001-00000001' [Feb 20 15:23:55] DEBUG[32580][C-00000001]: pbx_lua.c:1480 lua_find_extension: Looking up h@default:1 [Feb 20 15:23:55] DEBUG[32580][C-00000001]: pbx_lua.c:1480 lua_find_extension: Looking up h@demo:1 [Feb 20 15:23:55] DEBUG[32583]: bridge_channel.c:346 ast_bridge_channel_restore_formats: Bridge is returning 0x9335418(CBAnn/350001-00000001;2) to write format slin [Feb 20 15:23:55] DEBUG[32583]: channel.c:5680 set_format: Channel CBAnn/350001-00000001;2 setting write format path: slin -> slin [Feb 20 15:23:55] DEBUG[32580][C-00000001]: channel.c:2728 ast_hangup: Hanging up channel 'SIP/50001-00000001' [Feb 20 15:23:55] DEBUG[32580][C-00000001]: chan_sip.c:7154 sip_hangup: Hangup call SIP/50001-00000001, SIP callid dV~eC~AMO2 [Feb 20 15:23:55] DEBUG[32580][C-00000001]: res_rtp_asterisk.c:5036 ast_rtp_remote_address_set: Setting RTCP address on RTP instance '0xa02e108' [Feb 20 15:23:55] DEBUG[32580][C-00000001]: res_rtp_asterisk.c:5036 ast_rtp_remote_address_set: Setting RTCP address on RTP instance '0x9ee14c8' [Feb 20 15:23:55] DEBUG[32583]: channel.c:2728 ast_hangup: Hanging up channel 'CBAnn/350001-00000001;2' [Feb 20 15:23:55] DEBUG[32583]: bridge.c:648 destroy_bridge: Bridge 28f1b4ee-63be-49cb-b551-d5a7209f7781: actually destroying base bridge, nobody wants it anymore [Feb 20 15:23:55] DEBUG[32583]: bridge.c:673 destroy_bridge: Bridge 28f1b4ee-63be-49cb-b551-d5a7209f7781: calling base bridge destructor [Feb 20 15:23:55] DEBUG[32583]: bridge.c:679 destroy_bridge: Bridge 28f1b4ee-63be-49cb-b551-d5a7209f7781: calling softmix technology stop [Feb 20 15:23:55] DEBUG[32583]: bridge.c:686 destroy_bridge: Bridge 28f1b4ee-63be-49cb-b551-d5a7209f7781: calling softmix technology destructor [Feb 20 15:23:55] DEBUG[32581]: bridge_softmix.c:1127 softmix_mixing_thread: Bridge 28f1b4ee-63be-49cb-b551-d5a7209f7781: stopping mixing thread [Feb 20 15:23:55] DEBUG[32583]: bridge_softmix.c:1221 softmix_bridge_destroy: Bridge 28f1b4ee-63be-49cb-b551-d5a7209f7781: Waiting for mixing thread to die. [Feb 20 15:23:56] DEBUG[32496]: cdr_radius.c:222 radius_log: Unable to create RADIUS record. CDR not recorded! [Feb 20 15:23:56] DEBUG[32495]: devicestate.c:369 _ast_device_state: No provider found, checking channel drivers for SIP - 50001 [Feb 20 15:23:56] DEBUG[32495]: chan_sip.c:30178 sip_devicestate: Checking device state for peer 50001 [Feb 20 15:23:56] DEBUG[32495]: devicestate.c:474 do_state_change: Changing state for SIP/50001 - state 1 (Not in use) [Feb 20 15:23:56] DEBUG[32496]: res_config_sqlite.c:827 cdr_handler: SQL query: INSERT INTO ast_cdr (clid,src,dst,dcontext,channel,dstchannel,lastapp,lastdata,start,answer,end,duration,billsec,disposition,amaflags,uniqueid) VALUES ('"" <50001>','50001','350001','default','SIP/50001-00000001','SIP/50002-00000002','ConfBridge','350001,default_bridge_32,default_useradmin,sample_user_menu','2017-02-20 15:23:35','2017-02-20 15:23:36','2017-02-20 15:23:49','13','13','ANSWERED','DOCUMENTATION','1487575415.4') <--- SIP read from UDP:192.168.1.147:43521 ---> <-------------> [Feb 20 15:23:56] DEBUG[32544]: chan_sip.c:9893 parse_request: Header 0 [ 0]: [Feb 20 15:23:56] DEBUG[32495]: devicestate.c:369 _ast_device_state: No provider found, checking channel drivers for CBAnn - 350001 [Feb 20 15:23:56] DEBUG[32495]: devicestate.c:474 do_state_change: Changing state for CBAnn/350001 - state 0 (Unknown) [Feb 20 15:23:56] DEBUG[32566]: app_queue.c:2482 device_state_cb: Device 'CBAnn/350001' changed to state '0' (Unknown) but we don't care because they're not a member of any queue. <--- SIP read from UDP:192.168.1.147:23681 ---> <-------------> [Feb 20 15:24:01] DEBUG[32544]: chan_sip.c:9893 parse_request: Header 0 [ 0]: [Feb 20 15:24:04] DEBUG[32509]: threadpool.c:1137 worker_idle: Worker thread idle timeout reached. Dying. [Feb 20 15:24:04] DEBUG[32503]: threadpool.c:996 worker_thread_destroy: Destroying worker thread 6 [Feb 20 15:24:04] DEBUG[32508]: threadpool.c:1137 worker_idle: Worker thread idle timeout reached. Dying. [Feb 20 15:24:04] DEBUG[32503]: threadpool.c:996 worker_thread_destroy: Destroying worker thread 5 [Feb 20 15:24:05] DEBUG[32550]: res_pjsip_registrar_expire.c:78 check_expiration_thread: Woke up at 1487575445 Interval: 30 [Feb 20 15:24:05] DEBUG[32550]: res_pjsip_registrar_expire.c:85 check_expiration_thread: Expiring 0 contacts <--- SIP read from UDP:192.168.1.147:43521 ---> <-------------> [Feb 20 15:24:06] DEBUG[32544]: chan_sip.c:9893 parse_request: Header 0 [ 0]: <--- SIP read from UDP:192.168.1.147:23681 ---> <-------------> [Feb 20 15:24:11] DEBUG[32544]: chan_sip.c:9893 parse_request: Header 0 [ 0]: [Feb 20 15:24:16] DEBUG[32568]: threadpool.c:1137 worker_idle: Worker thread idle timeout reached. Dying. [Feb 20 15:24:16] DEBUG[32484]: threadpool.c:996 worker_thread_destroy: Destroying worker thread 13 <--- SIP read from UDP:192.168.1.147:43521 ---> <-------------> [Feb 20 15:24:16] DEBUG[32544]: chan_sip.c:9893 parse_request: Header 0 [ 0]: [Feb 20 15:24:20] DEBUG[32544]: chan_sip.c:4361 __sip_autodestruct: Auto destroying SIP dialog 'AOK5~O9uyp' [Feb 20 15:24:20] DEBUG[32544]: chan_sip.c:6589 sip_pvt_dtor: Destroying SIP dialog AOK5~O9uyp Really destroying SIP dialog 'AOK5~O9uyp' Method: BYE [Feb 20 15:24:20] DEBUG[32544]: rtp_engine.c:397 instance_destructor: Destroyed RTP instance '0x99a1948' [Feb 20 15:24:20] DEBUG[32544]: rtp_engine.c:397 instance_destructor: Destroyed RTP instance '0xa5db3e8' <--- SIP read from UDP:192.168.1.147:23681 ---> <-------------> [Feb 20 15:24:21] DEBUG[32544]: chan_sip.c:9893 parse_request: Header 0 [ 0]: [Feb 20 15:24:23] DEBUG[32544]: chan_sip.c:4361 __sip_autodestruct: Auto destroying SIP dialog 'dV~eC~AMO2' [Feb 20 15:24:23] DEBUG[32544]: chan_sip.c:6589 sip_pvt_dtor: Destroying SIP dialog dV~eC~AMO2 Really destroying SIP dialog 'dV~eC~AMO2' Method: BYE [Feb 20 15:24:23] DEBUG[32544]: rtp_engine.c:397 instance_destructor: Destroyed RTP instance '0xa02e108' [Feb 20 15:24:23] DEBUG[32544]: rtp_engine.c:397 instance_destructor: Destroyed RTP instance '0x9ee14c8' <--- SIP read from UDP:192.168.1.147:43521 ---> <-------------> [Feb 20 15:24:26] DEBUG[32544]: chan_sip.c:9893 parse_request: Header 0 [ 0]: <--- SIP read from UDP:192.168.1.147:23681 ---> <-------------> [Feb 20 15:24:31] DEBUG[32544]: chan_sip.c:9893 parse_request: Header 0 [ 0]: [Feb 20 15:24:35] DEBUG[32550]: res_pjsip_registrar_expire.c:78 check_expiration_thread: Woke up at 1487575475 Interval: 30 [Feb 20 15:24:35] DEBUG[32550]: res_pjsip_registrar_expire.c:85 check_expiration_thread: Expiring 0 contacts <--- SIP read from UDP:192.168.1.147:43521 ---> <-------------> [Feb 20 15:24:36] DEBUG[32544]: chan_sip.c:9893 parse_request: Header 0 [ 0]: <--- SIP read from UDP:192.168.1.147:23681 ---> <-------------> [Feb 20 15:24:41] DEBUG[32544]: chan_sip.c:9893 parse_request: Header 0 [ 0]: <--- SIP read from UDP:192.168.1.147:23681 ---> <-------------> [Feb 20 15:24:51] DEBUG[32544]: chan_sip.c:9893 parse_request: Header 0 [ 0]: <--- SIP read from UDP:192.168.1.147:43521 ---> <-------------> [Feb 20 15:24:55] DEBUG[32544]: chan_sip.c:9893 parse_request: Header 0 [ 0]: <--- SIP read from UDP:192.168.1.147:23681 ---> <-------------> [Feb 20 15:25:01] DEBUG[32544]: chan_sip.c:9893 parse_request: Header 0 [ 0]: [Feb 20 15:25:05] DEBUG[32550]: res_pjsip_registrar_expire.c:78 check_expiration_thread: Woke up at 1487575505 Interval: 30 [Feb 20 15:25:05] DEBUG[32550]: res_pjsip_registrar_expire.c:85 check_expiration_thread: Expiring 0 contacts <--- SIP read from UDP:192.168.1.147:43521 ---> <-------------> [Feb 20 15:25:06] DEBUG[32544]: chan_sip.c:9893 parse_request: Header 0 [ 0]: <--- SIP read from UDP:192.168.1.147:23681 ---> <-------------> [Feb 20 15:25:11] DEBUG[32544]: chan_sip.c:9893 parse_request: Header 0 [ 0]: <--- SIP read from UDP:192.168.1.147:43521 ---> <-------------> [Feb 20 15:25:17] DEBUG[32544]: chan_sip.c:9893 parse_request: Header 0 [ 0]: <--- SIP read from UDP:192.168.1.147:23681 ---> <-------------> [Feb 20 15:25:21] DEBUG[32544]: chan_sip.c:9893 parse_request: Header 0 [ 0]: <--- SIP read from UDP:192.168.1.147:43521 ---> <-------------> [Feb 20 15:25:28] DEBUG[32544]: chan_sip.c:9893 parse_request: Header 0 [ 0]: <--- SIP read from UDP:192.168.1.147:23681 ---> <-------------> [Feb 20 15:25:31] DEBUG[32544]: chan_sip.c:9893 parse_request: Header 0 [ 0]: [Feb 20 15:25:35] DEBUG[32550]: res_pjsip_registrar_expire.c:78 check_expiration_thread: Woke up at 1487575535 Interval: 30 [Feb 20 15:25:35] DEBUG[32550]: res_pjsip_registrar_expire.c:85 check_expiration_thread: Expiring 0 contacts <--- SIP read from UDP:192.168.1.147:23681 ---> <-------------> [Feb 20 15:25:41] DEBUG[32544]: chan_sip.c:9893 parse_request: Header 0 [ 0]: <--- SIP read from UDP:192.168.1.147:43521 ---> <-------------> [Feb 20 15:25:42] DEBUG[32544]: chan_sip.c:9893 parse_request: Header 0 [ 0]: <--- SIP read from UDP:192.168.1.147:23681 ---> <-------------> [Feb 20 15:25:51] DEBUG[32544]: chan_sip.c:9893 parse_request: Header 0 [ 0]: <--- SIP read from UDP:192.168.1.147:43521 ---> <-------------> [Feb 20 15:25:52] DEBUG[32544]: chan_sip.c:9893 parse_request: Header 0 [ 0]: <--- SIP read from UDP:192.168.1.147:23681 ---> <-------------> [Feb 20 15:26:01] DEBUG[32544]: chan_sip.c:9893 parse_request: Header 0 [ 0]: <--- SIP read from UDP:192.168.1.147:43521 ---> <-------------> [Feb 20 15:26:03] DEBUG[32544]: chan_sip.c:9893 parse_request: Header 0 [ 0]: [Feb 20 15:26:05] DEBUG[32550]: res_pjsip_registrar_expire.c:78 check_expiration_thread: Woke up at 1487575565 Interval: 30 [Feb 20 15:26:05] DEBUG[32550]: res_pjsip_registrar_expire.c:85 check_expiration_thread: Expiring 0 contacts <--- SIP read from UDP:192.168.1.147:23681 ---> <-------------> [Feb 20 15:26:11] DEBUG[32544]: chan_sip.c:9893 parse_request: Header 0 [ 0]: <--- SIP read from UDP:192.168.1.147:43521 ---> <-------------> [Feb 20 15:26:14] DEBUG[32544]: chan_sip.c:9893 parse_request: Header 0 [ 0]: <--- SIP read from UDP:192.168.1.147:23681 ---> <-------------> [Feb 20 15:26:21] DEBUG[32544]: chan_sip.c:9893 parse_request: Header 0 [ 0]: <--- SIP read from UDP:192.168.1.147:43521 ---> <-------------> [Feb 20 15:26:25] DEBUG[32544]: chan_sip.c:9893 parse_request: Header 0 [ 0]: