[root@kha ~]# /usr/sbin/asterisk -vvvvc /usr/sbin/asterisk -vvvvc == Parsing '/etc/asterisk/asterisk.conf': Found == Parsing '/etc/asterisk/extconfig.conf': Found Asterisk CVS-HEAD, Copyright (C) 1999 - 2005 Digium. Written by Mark Spencer ========================================================================= == Parsing '/etc/asterisk/logger.conf': Found Asterisk Event Logger Started /var/log/asterisk/event_log == Parsing '/etc/asterisk/dnsmgr.conf': Found Asterisk Dynamic Loader loading preload modules: == Parsing '/etc/asterisk/modules.conf': Found == Manager registered action Ping == Manager registered action Events == Manager registered action Logoff == Manager registered action Hangup == Manager registered action Status == Manager registered action Setvar == Manager registered action Getvar == Manager registered action Redirect == Manager registered action Originate == Manager registered action Command == Manager registered action ExtensionState == Manager registered action AbsoluteTimeout == Manager registered action MailboxStatus == Manager registered action MailboxCount == Manager registered action ListCommands == Parsing '/etc/asterisk/manager.conf': Found Asterisk Management interface listening on port 5038 == Parsing '/etc/asterisk/cdr.conf': Found Aug 24 12:03:20 NOTICE[12153]: cdr.c:1160 do_reload: CDR simple logging enabled. == Parsing '/etc/asterisk/rtp.conf': Found == RTP Allocating from port range 10000 -> 20000 Asterisk PBX Core Initializing Registering builtin applications: [AbsoluteTimeout] == Registered application 'AbsoluteTimeout' [Answer] == Registered application 'Answer' [BackGround] == Registered application 'BackGround' [Busy] == Registered application 'Busy' [Congestion] == Registered application 'Congestion' [DigitTimeout] == Registered application 'DigitTimeout' [Goto] == Registered application 'Goto' [GotoIf] == Registered application 'GotoIf' [GotoIfTime] == Registered application 'GotoIfTime' [ExecIfTime] == Registered application 'ExecIfTime' [Hangup] == Registered application 'Hangup' [NoOp] == Registered application 'NoOp' [Prefix] == Registered application 'Prefix' [Progress] == Registered application 'Progress' [ResetCDR] == Registered application 'ResetCDR' [ResponseTimeout] == Registered application 'ResponseTimeout' [Ringing] == Registered application 'Ringing' [SayNumber] == Registered application 'SayNumber' [SayDigits] == Registered application 'SayDigits' [SayAlpha] == Registered application 'SayAlpha' [SayPhonetic] == Registered application 'SayPhonetic' [SetAccount] == Registered application 'SetAccount' [SetAMAFlags] == Registered application 'SetAMAFlags' [SetGlobalVar] == Registered application 'SetGlobalVar' [SetLanguage] == Registered application 'SetLanguage' [Set] == Registered application 'Set' [SetVar] == Registered application 'SetVar' [ImportVar] == Registered application 'ImportVar' [StripMSD] == Registered application 'StripMSD' [Suffix] == Registered application 'Suffix' [Wait] == Registered application 'Wait' [WaitExten] == Registered application 'WaitExten' Asterisk Dynamic Loader Starting: == Parsing '/etc/asterisk/modules.conf': Found [chan_modem.so] => (Generic Voice Modem Driver) == Parsing '/etc/asterisk/modem.conf': Found == Loading modem driver chan_modem_aopen.so => (A/Open (Rockwell Chipset) ITU-2 VoiceModem Driver) == Registered channel type 'Modem' (Generic Voice Modem Channel Driver) [res_musiconhold.so] => (Music On Hold Resource) == Registered application 'MusicOnHold' == Registered application 'WaitMusicOnHold' == Registered application 'SetMusicOnHold' == Registered application 'StartMusicOnHold' == Registered application 'StopMusicOnHold' == Parsing '/etc/asterisk/musiconhold.conf': Found Aug 24 12:03:21 WARNING[12153]: res_musiconhold.c:976 load_moh_classes: The old musiconhold.conf syntax has been deprecated! Please refer to the sample configuration for information on the new syntax. Aug 24 12:03:21 WARNING[12153]: res_musiconhold.c:813 moh_register: Unable to open pseudo channel for timing... Sound may be choppy. [res_odbc.so] => (ODBC Resource) == Parsing '/etc/asterisk/res_odbc.conf': Found Aug 24 12:03:21 NOTICE[12153]: res_odbc.c:215 load_odbc_config: registered database handle 'asterisk' dsn->[asterisk] Aug 24 12:03:21 NOTICE[12153]: res_odbc.c:473 odbc_obj_connect: Connecting asterisk Aug 24 12:03:21 WARNING[12153]: res_odbc.c:484 odbc_obj_connect: res_odbc: Error SQLConnect=-1 errno=0 [unixODBC][Driver Manager]Data source name not found, and no default driver specified Aug 24 12:03:21 NOTICE[12153]: res_odbc.c:518 load_module: res_odbc loaded. [res_adsi.so] => (ADSI Resource) == Parsing '/etc/asterisk/adsi.conf': Found [res_features.so] => (Call Features Resource) == Parsing '/etc/asterisk/features.conf': Found -- Registered extension context 'parkedcalls' -- Added extension '700' priority 1 to parkedcalls == Registered application 'ParkedCall' == Registered application 'Park' == Manager registered action ParkedCalls [res_config_odbc.so] => (ODBC Configuration) Aug 24 12:03:21 NOTICE[12153]: config.c:836 ast_config_engine_register: Registered Config Engine odbc res_config_odbc loaded. [res_indications.so] => (Indications Configuration) == Parsing '/etc/asterisk/indications.conf': Found -- Registered indication country 'at' -- Registered indication country 'au' -- 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 'fi' -- Registered indication country 'fr' -- Registered indication country 'gr' -- Registered indication country 'hu' -- Registered indication country 'it' -- Registered indication country 'lt' -- Registered indication country 'mx' -- Registered indication country 'nl' -- Registered indication country 'no' -- Registered indication country 'nz' -- Registered indication country 'pl' -- Registered indication country 'pt' -- Registered indication country 'ru' -- Registered indication country 'se' -- Registered indication country 'sg' -- Registered indication country 'uk' -- Registered indication country 'us' -- Registered indication country 'us-o' -- Registered indication country 'tw' -- Registered indication country 'za' -- Setting default indication country to 'us' == Registered application 'PlayTones' == Registered application 'StopPlayTones' [res_monitor.so] => (Call Monitoring Resource) == Registered application 'Monitor' == Registered application 'StopMonitor' == Registered application 'ChangeMonitor' == Manager registered action Monitor == Manager registered action StopMonitor == Manager registered action ChangeMonitor [res_crypto.so] => (Cryptographic Digital Signatures) -- Loaded PUBLIC key 'iaxtel' -- Loaded PUBLIC key 'freeworlddialup' [res_agi.so] => (Asterisk Gateway Interface (AGI)) == Registered application 'DeadAGI' == Registered application 'EAGI' == Registered application 'AGI' [chan_sip.so] => (Session Initiation Protocol (SIP)) == Registered channel type 'SIP' (Session Initiation Protocol (SIP)) == Parsing '/etc/asterisk/sip.conf': Found -- SIP Seeding peer from astdb: '1236' at 1236@192.168.125.45:5068 for 900 -- SIP Seeding peer from astdb: '1238' at 1238@192.168.125.46:5060 for 200 -- SIP Seeding peer from astdb: '1239' at 1239@192.168.125.48:2051 for 3600 -- SIP Seeding peer from astdb: '1240' at 1240@192.168.125.49:2051 for 3600 == SIP Listening on 0.0.0.0:5060 == Using TOS bits 0 == Parsing '/etc/asterisk/sip_notify.conf': Found == Registered application 'SIPDtmfMode' == Registered application 'SIPAddHeader' == Registered application 'SIPGetHeader' == Registered custom function SIP_HEADER == Registered custom function SIPPEER == Registered custom function SIPCHANINFO == Manager registered action SIPpeers == Manager registered action SIPshowpeer [chan_phone.so] => (Linux Telephony API Support) == Parsing '/etc/asterisk/phone.conf': Found == Registered channel type 'Phone' (Standard Linux Telephony API Driver) [chan_modem_bestdata.so] => (BestData (Conexant V.90 Chipset) VoiceModem Driver) [chan_mgcp.so] => (Media Gateway Control Protocol (MGCP)) == Parsing '/etc/asterisk/mgcp.conf': Found == MGCP Listening on 0.0.0.0:2727 == Using TOS bits 0 == Registered channel type 'MGCP' (Media Gateway Control Protocol (MGCP)) [chan_features.so] => (Feature Proxy Channel) == Registered channel type 'Feature' (Feature Proxy Channel Driver) Warning, flexibel rate not heavily tested! -- Message count requested for mailbox 40@12 but voicemail not loaded. [chan_zap.so] => (Zapata Telephony) == Parsing '/etc/asterisk/zapata.conf': Found -- Automatically generated pseudo channel == Registered channel type 'Zap' (Zapata Telephony Driver) == Manager registered action ZapTransfer == Manager registered action ZapHangup == Manager registered action ZapDialOffhook == Manager registered action ZapDNDon == Manager registered action ZapDNDoff == Manager registered action ZapShowChannels [chan_skinny.so] => (Skinny Client Control Protocol (Skinny)) == Parsing '/etc/asterisk/skinny.conf': Found == Skinny listening on 0.0.0.0:2000 == Registered channel type 'Skinny' (Skinny Client Control Protocol (Skinny)) [skipping chan_oss.so] [chan_modem_i4l.so] => (ISDN4Linux Emulated Modem Driver) [chan_agent.so] => (Agent Proxy Channel) == Registered channel type 'Agent' (Call Agent Proxy Channel) == Registered application 'AgentLogin' == Registered application 'AgentCallbackLogin' == Registered application 'AgentMonitorOutgoing' == Manager registered action Agents == Parsing '/etc/asterisk/agents.conf': Found [chan_local.so] => (Local Proxy Channel) == Registered channel type 'Local' (Local Proxy Channel Driver) [chan_iax2.so] => (Inter Asterisk eXchange (Ver 2)) == Registered custom function IAXPEER Aug 24 12:03:22 WARNING[12153]: chan_iax2.c:9355 load_module: Unable to open IAX timing interface: No such file or directory == Registered application 'IAX2Provision' == Manager registered action IAXpeers == Manager registered action IAXnetstats == Parsing '/etc/asterisk/iax.conf': Found -- doing lookup for '216.207.245.47' == Registered channel type 'IAX2' (Inter Asterisk eXchange Driver (Ver 2)) == Using TOS bits 16 == Binding IAX2 to default address 0.0.0.0:4569 == IAX Ready and Listening == Loaded firmware 'iaxy.bin' == Parsing '/etc/asterisk/iaxprov.conf': Found -- Loaded provisioning template 'default' [skipping chan_alsa.so] [pbx_dundi.so] => (Distributed Universal Number Discovery (DUNDi)) == Parsing '/etc/asterisk/dundi.conf': Found == Using TOS bits 0 == DUNDi Ready and Listening on 0.0.0.0 port 4520 == Registered application 'DUNDiLookup' [pbx_realtime.so] => (Realtime Switch) [pbx_loopback.so] => (Loopback Switch) [pbx_spool.so] => (Outgoing Spool Support) [pbx_ael.so] => (Asterisk Extension Language Compiler) -- Registered extension context 'macro-std-exten-ael' -- Added extension 's' priority 1 to macro-std-exten-ael -- Added extension 's' priority 2 to macro-std-exten-ael -- Added extension 's' priority 3 to macro-std-exten-ael -- Added extension 's' priority 4 to macro-std-exten-ael -- Added extension 's' priority 5 to macro-std-exten-ael -- Added extension 'sw-std-exten-ael-4-BUSY' priority 1 to macro-std-exten-ael -- Added extension 'sw-std-exten-ael-4-BUSY' priority 2 to macro-std-exten-ael -- Added extension '_sw-std-exten-ael-4-.' priority 1 to macro-std-exten-ael -- Added extension 'a' priority 1 to macro-std-exten-ael -- Added extension 'a' priority 2 to macro-std-exten-ael -- Registered extension context 'ael-demo' -- 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 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 's' priority 7 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 '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 '#' 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 [pbx_functions.so] => (Builtin dialplan functions) == Registered custom function MD5 == Registered custom function CHECK_MD5 == Registered custom function MATH == Registered custom function GROUP_COUNT == Registered custom function GROUP_MATCH_COUNT == Registered custom function GROUP -- Saved useragent "Linphone-1.0.1/eXosip" for peer 1236 == Registered custom function GROUP_LIST == Registered custom function FIELDQTY == Registered custom function REGEX == Registered custom function LEN == Registered custom function STRFTIME == Registered custom function EVAL == Registered custom function CDR == Registered custom function ISNULL == Registered custom function SET == Registered custom function EXISTS == Registered custom function IF == Registered custom function IFTIME == Registered custom function ENV == Registered custom function DB == Registered custom function DB_EXISTS == Registered custom function TIMEOUT == Registered custom function LANGUAGE == Registered custom function MUSICCLASS [pbx_config.so] => (Text Extension Configuration) == Parsing '/etc/asterisk/extensions.conf': Found -- Registered extension context 'default' -- Added extension '1' priority 1 to default -- Added extension '1' priority 2 to default -- Added extension '2' priority 1 to default -- Added extension '2' priority 2 to default -- Added extension '3' priority 1 to default -- Added extension '3' priority 2 to default -- Added extension '4' priority 1 to default Aug 24 12:03:22 WARNING[12153]: pbx.c:4555 ast_add_extension2: Unable to register extension '3', priority 2 in 'default', already in use Aug 24 12:03:22 WARNING[12153]: pbx_config.c:1758 pbx_load_module: Unable to register extension at line 25 -- Registered extension context 'dial-BUSY' -- Added extension '1' priority 1 to dial-BUSY -- Added extension '2' priority 1 to dial-BUSY -- Added extension '3' priority 1 to dial-BUSY -- Added extension '4' priority 1 to dial-BUSY -- Registered extension context 'dial-ANSWER' -- Added extension '_[1234]' priority 1 to dial-ANSWER -- Registered extension context 'dial-NOANSWER' -- Added extension '_[1234]' priority 1 to dial-NOANSWER -- Registered extension context 'dial-CONGESTION' -- Added extension '_[1234]' priority 1 to dial-CONGESTION -- Registered extension context 'dial-CHANUNAVAIL' -- Added extension '_[1234]' priority 1 to dial-CHANUNAVAIL -- Registered extension context 'dial-CANCEL' -- Added extension '_[1234]' priority 1 to dial-CANCEL [app_dictate.so] => (Virtual Dictation Machine) == Registered application 'Dictate' [app_cut.so] => (Cuts up variables) == Registered application 'Cut' [app_dial.so] => (Dialing Application) == Registered application 'Dial' == Registered application 'RetryDial' [app_milliwatt.so] => (Digital Milliwatt (mu-law) Test Application) == Registered application 'Milliwatt' [format_g723.so] => (G.723.1 Simple Timestamp File Format) == Registered file format g723sf, extension(s) g723|g723sf [app_settransfercapability.so] => (Set ISDN Transfer Capability) == Registered application 'SetTransferCapability' [app_system.so] => (Generic System() application) == Registered application 'TrySystem' == Registered application 'System' [codec_adpcm.so] => (Adaptive Differential PCM Coder/Decoder) == Parsing '/etc/asterisk/codecs.conf': Found -- codec_adpcm: using generic PLC == Registered translator 'adpcmtolin' from format adpcm to slin, cost 1 == Registered translator 'lintoadpcm' from format slin to adpcm, cost 1 [format_sln.so] => (Raw Signed Linear Audio support (SLN)) == Registered file format sln, extension(s) sln|raw [app_hasnewvoicemail.so] => (Indicator for whether a voice mailbox has messages in a given folder.) == Registered application 'HasVoicemail' == Registered application 'HasNewVoicemail' [format_jpeg.so] => (JPEG (Joint Picture Experts Group) Image Format) == Registered format 'jpg' (JPEG (Joint Picture Experts Group)) [format_h263.so] => (Raw h263 data) == Registered file format h263, extension(s) h263 [app_math.so] => (Basic Math Functions) == Registered application 'Math' [app_read.so] => (Read Variable Application) == Registered application 'Read' [app_setcallerid.so] => (Set CallerID Application) == Registered application 'SetCallerPres' == Registered application 'SetCallerID' [app_talkdetect.so] => (Playback with Talk Detection) == Registered application 'BackgroundDetect' [app_db.so] => (Database access functions for Asterisk extension logic) == Registered application 'DBget' == Registered application 'DBput' == Registered application 'DBdel' == Registered application 'DBdeltree' [app_enumlookup.so] => (ENUM Lookup) == Registered application 'EnumLookup' == Parsing '/etc/asterisk/enum.conf': Found [app_ices.so] => (Encode and Stream via icecast and ices) == Registered application 'ICES' [codec_gsm.so] => (GSM/PCM16 (signed linear) Codec Translator) == Parsing '/etc/asterisk/codecs.conf': Found -- codec_gsm: using generic PLC == Registered translator 'gsmtolin' from format gsm to slin, cost 1 == Registered translator 'lintogsm' from format slin to gsm, cost 4 [app_forkcdr.so] => (Fork The CDR into 2 separate entities.) == Registered application 'ForkCDR' [app_chanisavail.so] => (Check if channel is available) == Registered application 'ChanIsAvail' [format_vox.so] => (Dialogic VOX (ADPCM) File Format) == Registered file format vox, extension(s) vox [app_cdr.so] => (Make sure asterisk doesn't save CDR for a certain call) == Registered application 'NoCDR' [app_dumpchan.so] => (Dump Info About The Calling Channel) == Registered application 'DumpChan' [app_lookupblacklist.so] => (Look up Caller*ID name/number from blacklist database) == Registered application 'LookupBlacklist' [app_zapateller.so] => (Block Telemarketers with Special Information Tone) == Registered application 'Zapateller' [app_sendtext.so] => (Send Text Applications) == Registered application 'SendText' [app_groupcount.so] => (Group Management Routines) == Registered application 'GetGroupCount' == Registered application 'SetGroup' == Registered application 'CheckGroup' == Registered application 'GetGroupMatchCount' [func_callerid.so] => (Caller ID related dialplan function) == Registered custom function CALLERID [app_substring.so] => ((Deprecated) Save substring digits in a given variable) == Registered application 'SubString' [codec_speex.so] => (Speex/PCM16 (signed linear) Codec Translator) == Parsing '/etc/asterisk/codecs.conf': Found -- CODEC SPEEX: Setting Quality to 3 -- CODEC SPEEX: Setting Complexity to 4 -- CODEC SPEEX: Perceptual Enhancement Mode. [on] -- CODEC SPEEX: VAD Mode. [off] -- CODEC SPEEX: VBR Mode. [off] -- CODEC SPEEX: Disabling ABR -- CODEC SPEEX: Setting VBR Quality to 5.000000 -- CODEC SPEEX: DTX Mode. [off] == Registered translator 'speextolin' from format speex to slin, cost 3 == Registered translator 'lintospeex' from format slin to speex, cost 45 [app_record.so] => (Trivial Record Application) == Registered application 'Record' [format_wav.so] => (Microsoft WAV format (8000hz Signed Linear)) == Registered file format wav, extension(s) wav [codec_alaw.so] => (A-law Coder/Decoder) == Parsing '/etc/asterisk/codecs.conf': Found -- codec_alaw: using generic PLC == Registered translator 'alawtolin' from format alaw to slin, cost 1 == Registered translator 'lintoalaw' from format slin to alaw, cost 1 [app_test.so] => (Interface Test Application) == Registered application 'TestClient' == Registered application 'TestServer' [format_pcm.so] => (Raw uLaw 8khz Audio support (PCM)) == Registered file format pcm, extension(s) pcm|ulaw|ul|mu [app_authenticate.so] => (Authentication Application) == Registered application 'Authenticate' [format_wav_gsm.so] => (Microsoft WAV format (Proprietary GSM)) == Registered file format wav49, extension(s) WAV|wav49 [cdr_pgsql.so] => (PostgreSQL CDR Backend) == Parsing '/etc/asterisk/cdr_pgsql.conf': Found [app_chanspy.so] => (Tap into any type of asterisk channel and listen to audio) == Registered application 'ChanSpy' [app_macro.so] => (Extension Macros) == Registered application 'MacroExit' == Registered application 'MacroIf' == Registered application 'Macro' [app_waitforsilence.so] => (Wait For Silence) == Registered application 'WaitForSilence' [format_gsm.so] => (Raw GSM data) == Registered file format gsm, extension(s) gsm [app_meetme.so] => (MeetMe conference bridge) == Registered application 'MeetMeAdmin' == Registered application 'MeetMeCount' == Registered application 'MeetMe' [app_sms.so] => (SMS/PSTN handler) == Registered application 'SMS' [app_curl.so] => (Load external URL) == Registered application 'Curl' [cdr_manager.so] => (Asterisk Call Manager CDR Backend) == Parsing '/etc/asterisk/cdr_manager.conf': Found [app_queue.so] => (True Call Queueing) == Registered application 'Queue' == Manager registered action Queues == Manager registered action QueueStatus == Manager registered action QueueAdd == Manager registered action QueueRemove == Manager registered action QueuePause == Registered application 'AddQueueMember' == Registered application 'RemoveQueueMember' == Registered application 'PauseQueueMember' == Registered application 'UnpauseQueueMember' == Registered custom function QUEUEAGENTCOUNT == Parsing '/etc/asterisk/queues.conf': Found [app_verbose.so] => (Send verbose output) == Registered application 'Verbose' [app_disa.so] => (DISA (Direct Inward System Access) Application) == Registered application 'DISA' [app_while.so] => (While Loops and Conditional Execution) == Registered application 'While' == Registered application 'ExecIf' == Registered application 'EndWhile' [codec_ulaw.so] => (Mu-law Coder/Decoder) == Parsing '/etc/asterisk/codecs.conf': Found -- codec_ulaw: using generic PLC == Registered translator 'ulawtolin' from format ulaw to slin, cost 1 == Registered translator 'lintoulaw' from format slin to ulaw, cost 1 [app_setcdruserfield.so] => (CDR user field apps) == Registered application 'SetCDRUserField' == Registered application 'AppendCDRUserField' == Manager registered action SetCDRUserField [app_transfer.so] => (Transfer) == Registered application 'Transfer' [app_striplsd.so] => (Strip trailing digits) == Registered application 'StripLSD' [app_zapscan.so] => (Scan Zap channels application) == Registered application 'ZapScan' [app_privacy.so] => (Require phone number to be entered, if no CallerID sent) == Registered application 'PrivacyManager' [cdr_custom.so] => (Customizable Comma Separated Values CDR Backend) == Parsing '/etc/asterisk/cdr_custom.conf': Found [app_url.so] => (Send URL Applications) == Registered application 'SendURL' [app_sayunixtime.so] => (Say time) == Registered application 'SayUnixTime' == Registered application 'DateTime' [app_parkandannounce.so] => (Call Parking and Announce Application) == Registered application 'ParkAndAnnounce' [app_echo.so] => (Simple Echo Application) == Registered application 'Echo' [cdr_csv.so] => (Comma Separated Values CDR Backend) [app_flash.so] => (Flash zap trunk application) == Registered application 'Flash' [app_readfile.so] => (Stores output of file into a variable) == Registered application 'ReadFile' [app_waitforring.so] => (Waits until first ring after time) == Registered application 'WaitForRing' [app_getcpeid.so] => (Get ADSI CPE ID) == Registered application 'GetCPEID' [app_softhangup.so] => (Hangs up the requested channel) == Registered application 'SoftHangup' [app_festival.so] => (Simple Festival Interface) == Registered application 'Festival' [app_setrdnis.so] => (Set RDNIS Number) == Registered application 'SetRDNIS' [skipping chan_oss.so] [app_zapbarge.so] => (Barge in on Zap channel application) == Registered application 'ZapBarge' [app_md5.so] => (MD5 checksum applications) == Registered application 'MD5Check' == Registered application 'MD5' [codec_a_mu.so] => (A-law and Mulaw direct Coder/Decoder) == Registered translator 'alawtoulaw' from format alaw to ulaw, cost 1 == Registered translator 'ulawtoalaw' from format ulaw to alaw, cost 1 [format_g729.so] => (Raw G729 data) == Registered file format g729, extension(s) g729 [format_ogg_vorbis.so] => (OGG/Vorbis audio) == Registered file format ogg_vorbis, extension(s) ogg [app_voicemail.so] => (Comedian Mail (Voicemail System)) == Registered application 'VoiceMail' == Registered application 'VoiceMailMain' == Registered application 'MailboxExists' == Registered application 'VMAuthenticate' == Parsing '/etc/asterisk/voicemail.conf': Found [app_playback.so] => (Sound File Playback Application) == Registered application 'Playback' [app_mp3.so] => (Silly MP3 Application) == Registered application 'MP3Player' [app_externalivr.so] => (External IVR Interface Application) == Registered application 'ExternalIVR' [app_nbscat.so] => (Silly NBS Stream Application) == Registered application 'NBScat' [format_au.so] => (Sun Microsystems AU format (signed linear)) == Registered file format au, extension(s) au [codec_g726.so] => (ITU G.726-32kbps G726 Transcoder) == Parsing '/etc/asterisk/codecs.conf': Found -- codec_g726: using generic PLC == Registered translator 'g726tolin' from format g726 to slin, cost 2 == Registered translator 'lintog726' from format slin to g726, cost 2 [format_g726.so] => (Raw G.726 (16/24/32/40kbps) data) == 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 [app_eval.so] => (Reevaluates strings) == Registered application 'Eval' [codec_lpc10.so] => (LPC10 2.4kbps (signed linear) Voice Coder) == Parsing '/etc/asterisk/codecs.conf': Found -- codec_lpc10: using generic PLC == Registered translator 'lpc10tolin' from format lpc10 to slin, cost 3 == Registered translator 'lintolpc10' from format slin to lpc10, cost 5 [app_txtcidname.so] => (TXTCIDName) == Registered application 'TXTCIDName' == Parsing '/etc/asterisk/enum.conf': Found [app_random.so] => (Random goto) == Registered application 'Random' [format_pcm_alaw.so] => (Raw aLaw 8khz PCM Audio support) == Registered file format alaw, extension(s) alaw|al [cdr_odbc.so] => (ODBC CDR Backend) == Parsing '/etc/asterisk/cdr_odbc.conf': Found [app_adsiprog.so] => (Asterisk ADSI Programming Application) == Registered application 'ADSIProg' [app_setcidname.so] => (Set CallerID Name) == Registered application 'SetCIDName' [app_zapras.so] => (Zap RAS Application) == Registered application 'ZapRAS' [app_exec.so] => (Executes applications) == Registered application 'Exec' [app_alarmreceiver.so] => (Alarm Receiver for Asterisk) == Parsing '/etc/asterisk/alarmreceiver.conf': Found == Registered application 'AlarmReceiver' [skipping chan_alsa.so] [app_lookupcidname.so] => (Look up CallerID Name from local database) == Registered application 'LookupCIDName' [app_directory.so] => (Extension Directory) == Registered application 'Directory' [app_controlplayback.so] => (Control Playback Application) == Registered application 'ControlPlayback' [app_senddtmf.so] => (Send DTMF digits Application) == Registered application 'SendDTMF' [app_image.so] => (Image Transmission Application) == Registered application 'SendImage' [app_realtime.so] => (Realtime Data Lookup/Rewrite) == Registered application 'RealTimeUpdate' == Registered application 'RealTime' [app_setcidnum.so] => (Set CallerID Number) == Registered application 'SetCIDNum' [format_ilbc.so] => (Raw iLBC data) == Registered file format iLBC, extension(s) ilbc [app_userevent.so] => (Custom User Event Application) == Registered application 'UserEvent' [codec_ilbc.so] => (iLBC/PCM16 (signed linear) Codec Translator) == Registered translator 'ilbctolin' from format ilbc to slin, cost 3 == Registered translator 'lintoilbc' from format slin to ilbc, cost 21 == Manager registered action DBGet == Manager registered action DBPut == Parsing '/etc/asterisk/enum.conf': Found Asterisk Ready. *CLI> sip debug sip debug SIP Debugging enabled *CLI> <-- SIP read from 192.168.125.45:5068: INVITE sip:2@kha SIP/2.0 Via: SIP/2.0/UDP 192.168.125.45:5068;rport;branch=z9hG4bK1653610192 From: ;tag=196575185 To: Call-ID: 1533649758@192.168.125.45 CSeq: 20 INVITE Contact: Max-Forwards: 5 User-Agent: Linphone-1.0.1/eXosip Subject: Phone call Expires: 120 Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, REFER, SUBSCRIBE, NOTIFY, MESSAGE Content-Type: application/sdp Content-Length: 356 v=0 o=1236 123456 654321 IN IP4 192.168.125.45 s=A conversation c=IN IP4 192.168.125.45 t=0 0 m=audio 7078 RTP/AVP 0 3 8 110 111 115 101 b=AS:20 a=rtpmap:0 PCMU/8000/1 a=rtpmap:3 GSM/8000/1 a=rtpmap:8 PCMA/8000/1 a=rtpmap:110 speex/8000/1 a=rtpmap:111 speex/16000/1 a=rtpmap:115 1015/8000/1 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11 --- (14 headers 15 lines)--- Using INVITE request as basis request - 1533649758@192.168.125.45 Sending to 192.168.125.45 : 5068 (non-NAT) Reliably Transmitting (no NAT) to 192.168.125.45:5068: SIP/2.0 407 Proxy Authentication Required Via: SIP/2.0/UDP 192.168.125.45:5068;branch=z9hG4bK1653610192 From: ;tag=196575185 To: ;tag=as7ef9275f Call-ID: 1533649758@192.168.125.45 CSeq: 20 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Contact: Proxy-Authenticate: Digest realm="asterisk", nonce="5ace1f6f" Content-Length: 0 --- Scheduling destruction of call '1533649758@192.168.125.45' in 15000 ms Found user '1236' <-- SIP read from 192.168.125.45:5068: ACK sip:2@kha SIP/2.0 Via: SIP/2.0/UDP 192.168.125.45:5068;rport;branch=z9hG4bK1653610192 From: ;tag=196575185 To: ;tag=as7ef9275f Call-ID: 1533649758@192.168.125.45 CSeq: 20 ACK Content-Length: 0 --- (7 headers 0 lines)--- <-- SIP read from 192.168.125.45:5068: INVITE sip:2@kha SIP/2.0 Via: SIP/2.0/UDP 192.168.125.45:5068;rport;branch=z9hG4bK676966446 From: ;tag=196575185 To: Call-ID: 1533649758@192.168.125.45 CSeq: 21 INVITE Contact: Proxy-Authorization: Digest username="1236", realm="asterisk", nonce="5ace1f6f", uri="sip:2@kha", response="fe2187ac32cc7239f322fc3acd40de81", algorithm=MD5 Max-Forwards: 5 User-Agent: Linphone-1.0.1/eXosip Subject: Phone call Expires: 120 Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, REFER, SUBSCRIBE, NOTIFY, MESSAGE Content-Type: application/sdp Content-Length: 356 v=0 o=1236 123456 654321 IN IP4 192.168.125.45 s=A conversation c=IN IP4 192.168.125.45 t=0 0 m=audio 7078 RTP/AVP 0 3 8 110 111 115 101 b=AS:20 a=rtpmap:0 PCMU/8000/1 a=rtpmap:3 GSM/8000/1 a=rtpmap:8 PCMA/8000/1 a=rtpmap:110 speex/8000/1 a=rtpmap:111 speex/16000/1 a=rtpmap:115 1015/8000/1 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-11 --- (15 headers 15 lines)--- Using INVITE request as basis request - 1533649758@192.168.125.45 Sending to 192.168.125.45 : 5068 (non-NAT) Found user '1236' Found RTP audio format 0 Found RTP audio format 3 Found RTP audio format 8 Found RTP audio format 110 Found RTP audio format 111 Found RTP audio format 115 Found RTP audio format 101 Peer audio RTP is at port 192.168.125.45:7078 Found description format PCMU Found description format GSM Found description format PCMA Found description format speex Found description format speex Found description format 1015 Found description format telephone-event Capabilities: us - 0xc (ulaw|alaw), peer - audio=0x20e (gsm|ulaw|alaw|speex)/video=0x0 (nothing), combined - 0xc (ulaw|alaw) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Looking for 2 in default list_route: hop: Transmitting (no NAT) to 192.168.125.45:5068: SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.125.45:5068;branch=z9hG4bK676966446 From: ;tag=196575185 To: Call-ID: 1533649758@192.168.125.45 CSeq: 21 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Contact: Content-Length: 0 --- -- Executing Dial("SIP/1236-c047", "SIP/1238|20|tT") in new stack We're at 192.168.125.45 port 15686 Answering/Requesting with root capability 0x4 (ulaw) Answering with capability 0x8 (alaw) Answering with non-codec capability 0x1 (telephone-event) 12 headers, 11 lines Reliably Transmitting (no NAT) to 192.168.125.46:5060: INVITE sip:1238@192.168.125.46:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.125.45:5060;branch=z9hG4bK01d6c5b3;rport From: "Knut-Haavard Aksnes" ;tag=as3c5eaec5 To: Contact: Call-ID: 511d8680305bf8f51c1720361b9c76b3@192.168.125.45 CSeq: 102 INVITE User-Agent: Asterisk PBX Date: Wed, 24 Aug 2005 10:03:31 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Content-Type: application/sdp Content-Length: 244 v=0 o=root 12153 12153 IN IP4 192.168.125.45 s=session c=IN IP4 192.168.125.45 t=0 0 m=audio 15686 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- -- Called 1238 <-- SIP read from 192.168.125.46:5060: SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.125.45:5060;branch=z9hG4bK01d6c5b3;rport=5060 From: "Knut-Haavard Aksnes" ;tag=as3c5eaec5 To: Call-ID: 511d8680305bf8f51c1720361b9c76b3@192.168.125.45 CSeq: 102 INVITE Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, MESSAGE, INFO Content-Length: 0 --- (8 headers 0 lines)--- <-- SIP read from 192.168.125.46:5060: SIP/2.0 101 Dialog Establishement Via: SIP/2.0/UDP 192.168.125.45:5060;branch=z9hG4bK01d6c5b3;rport=5060 From: "Knut-Haavard Aksnes" ;tag=as3c5eaec5 To: ;tag=785849013 Call-ID: 511d8680305bf8f51c1720361b9c76b3@192.168.125.45 CSeq: 102 INVITE Contact: Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, MESSAGE, INFO Content-Length: 0 --- (9 headers 0 lines)--- -- SIP/1238-81b4 is making progress passing it to SIP/1236-c047 We're at 192.168.125.45 port 13350 Answering with preferred capability 0x4 (ulaw) Answering with preferred capability 0x8 (alaw) Answering with non-codec capability 0x1 (telephone-event) Transmitting (no NAT) to 192.168.125.45:5068: SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 192.168.125.45:5068;branch=z9hG4bK676966446 From: ;tag=196575185 To: ;tag=as663cbd9e Call-ID: 1533649758@192.168.125.45 CSeq: 21 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Contact: Content-Type: application/sdp Content-Length: 244 v=0 o=root 12153 12153 IN IP4 192.168.125.45 s=session c=IN IP4 192.168.125.45 t=0 0 m=audio 13350 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- <-- SIP read from 192.168.125.46:5060: SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.125.45:5060;branch=z9hG4bK01d6c5b3;rport=5060 From: "Knut-Haavard Aksnes" ;tag=as3c5eaec5 To: ;tag=785849013 Call-ID: 511d8680305bf8f51c1720361b9c76b3@192.168.125.45 CSeq: 102 INVITE Contact: Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, MESSAGE, INFO Content-Length: 0 --- (9 headers 0 lines)--- -- SIP/1238-81b4 is ringing Transmitting (no NAT) to 192.168.125.45:5068: SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.125.45:5068;branch=z9hG4bK676966446 From: ;tag=196575185 To: ;tag=as663cbd9e Call-ID: 1533649758@192.168.125.45 CSeq: 21 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Contact: Content-Length: 0 --- <-- SIP read from 192.168.125.46:5060: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.125.45:5060;branch=z9hG4bK01d6c5b3;rport=5060 From: "Knut-Haavard Aksnes" ;tag=as3c5eaec5 To: ;tag=785849013 Call-ID: 511d8680305bf8f51c1720361b9c76b3@192.168.125.45 CSeq: 102 INVITE Contact: Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, MESSAGE, INFO Content-Type: application/sdp Content-Length: 225 v=0 o=1238 123456 654321 IN IP4 192.168.125.46 s=A conversation c=IN IP4 192.168.125.46 t=0 0 m=audio 7078 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 --- (10 headers 10 lines)--- Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 101 Peer audio RTP is at port 192.168.125.46:7078 Found description format PCMU Found description format PCMA Found description format telephone-event Capabilities: us - 0xc (ulaw|alaw), peer - audio=0xc (ulaw|alaw)/video=0x0 (nothing), combined - 0xc (ulaw|alaw) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.125.46, port 5060 Transmitting (no NAT) to 192.168.125.46:5060: ACK sip:1238@192.168.125.46:5060 SIP/2.0 Via: SIP/2.0/UDP 192.168.125.45:5060;branch=z9hG4bK3370a2e4;rport From: "Knut-Haavard Aksnes" ;tag=as3c5eaec5 To: ;tag=785849013 Contact: Call-ID: 511d8680305bf8f51c1720361b9c76b3@192.168.125.45 CSeq: 102 ACK User-Agent: Asterisk PBX Content-Length: 0 --- -- SIP/1238-81b4 answered SIP/1236-c047 We're at 192.168.125.45 port 13350 Answering with preferred capability 0x4 (ulaw) Answering with preferred capability 0x8 (alaw) Answering with non-codec capability 0x1 (telephone-event) Reliably Transmitting (no NAT) to 192.168.125.45:5068: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.125.45:5068;branch=z9hG4bK676966446 From: ;tag=196575185 To: ;tag=as663cbd9e Call-ID: 1533649758@192.168.125.45 CSeq: 21 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Contact: Content-Type: application/sdp Content-Length: 244 v=0 o=root 12153 12154 IN IP4 192.168.125.45 s=session c=IN IP4 192.168.125.45 t=0 0 m=audio 13350 RTP/AVP 0 8 101 a=rtpmap:0 PCMU/8000 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- -- Attempting native bridge of SIP/1236-c047 and SIP/1238-81b4 <-- SIP read from 192.168.125.45:5068: ACK sip:2@192.168.125.45 SIP/2.0 Via: SIP/2.0/UDP 192.168.125.45:5068;rport;branch=z9hG4bK1600006588 From: ;tag=196575185 To: ;tag=as663cbd9e Call-ID: 1533649758@192.168.125.45 CSeq: 21 ACK Contact: Max-Forwards: 5 User-Agent: Linphone-1.0.1/eXosip Content-Length: 0 --- (10 headers 0 lines)--- Destroying call '764869696@192.168.125.45' Destroying call '660536972@192.168.125.46' <-- SIP read from 192.168.125.48:2051: INVITE sip:1@192.168.125.45;user=phone SIP/2.0 Via: SIP/2.0/UDP 192.168.125.48:2051;branch=z9hG4bK-sewfusr4r1t2;rport From: ;tag=te9v8vdq46 To: Call-ID: 3c26b3ec1388-ppbzigkafn6q@snom360 CSeq: 1 INVITE Max-Forwards: 70 Contact: P-Key-Flags: resolution="31x13", keys="4" User-Agent: snom360/3.60n Accept: application/sdp Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO Allow-Events: talk, hold, refer Supported: timer, 100rel, replaces Session-Expires: 3600 Content-Type: application/sdp Content-Length: 368 v=0 o=root 58652918 58652918 IN IP4 192.168.125.48 s=call c=IN IP4 192.168.125.48 t=0 0 m=audio 64354 RTP/AVP 0 8 9 2 3 18 4 101 a=rtpmap:0 pcmu/8000 a=rtpmap:8 pcma/8000 a=rtpmap:9 g722/8000 a=rtpmap:2 g726-32/8000 a=rtpmap:3 gsm/8000 a=rtpmap:18 g729/8000 a=rtpmap:4 g723/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=ptime:20 a=sendrecv --- (17 headers 17 lines)--- Using INVITE request as basis request - 3c26b3ec1388-ppbzigkafn6q@snom360 Sending to 192.168.125.48 : 2051 (non-NAT) Found user '1239' Found RTP audio format 0 Found RTP audio format 8 Found RTP audio format 9 Found RTP audio format 2 Found RTP audio format 3 Found RTP audio format 18 Found RTP audio format 4 Found RTP audio format 101 Peer audio RTP is at port 192.168.125.48:64354 Found description format pcmu Found description format pcma Found description format g722 Found description format g726-32 Found description format gsm Found description format g729 Found description format g723 Found description format telephone-event Capabilities: us - 0x8 (alaw), peer - audio=0x11f (g723|gsm|ulaw|alaw|g726|g729)/video=0x0 (nothing), combined - 0x8 (alaw) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) Looking for 1 in default list_route: hop: Transmitting (no NAT) to 192.168.125.48:2051: SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.125.48:2051;branch=z9hG4bK-sewfusr4r1t2 From: ;tag=te9v8vdq46 To: Call-ID: 3c26b3ec1388-ppbzigkafn6q@snom360 CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Contact: Content-Length: 0 --- -- Executing Dial("SIP/1239-4477", "SIP/1236|20|tT") in new stack We're at 192.168.125.45 port 10758 Answering/Requesting with root capability 0x8 (alaw) Answering with capability 0x4 (ulaw) Answering with non-codec capability 0x1 (telephone-event) 12 headers, 11 lines Reliably Transmitting (no NAT) to 192.168.125.45:5068: INVITE sip:1236@192.168.125.45:5068 SIP/2.0 Via: SIP/2.0/UDP 192.168.125.45:5060;branch=z9hG4bK209f0525;rport From: "SNOM 360" ;tag=as5623414f To: Contact: Call-ID: 334c32b351fe071f0c1130b43dd1eb3a@192.168.125.45 CSeq: 102 INVITE User-Agent: Asterisk PBX Date: Wed, 24 Aug 2005 10:03:39 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Content-Type: application/sdp Content-Length: 244 v=0 o=root 12153 12153 IN IP4 192.168.125.45 s=session c=IN IP4 192.168.125.45 t=0 0 m=audio 10758 RTP/AVP 8 0 101 a=rtpmap:8 PCMA/8000 a=rtpmap:0 PCMU/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- <-- SIP read from 192.168.125.45:5068: SIP/2.0 100 Trying Via: SIP/2.0/UDP 192.168.125.45:5060;branch=z9hG4bK209f0525;rport=5060 From: "SNOM 360" ;tag=as5623414f To: Call-ID: 334c32b351fe071f0c1130b43dd1eb3a@192.168.125.45 CSeq: 102 INVITE Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, MESSAGE, INFO Content-Length: 0 --- (8 headers 0 lines)--- -- Called 1236 <-- SIP read from 192.168.125.45:5068: SIP/2.0 101 Dialog Establishement Via: SIP/2.0/UDP 192.168.125.45:5060;branch=z9hG4bK209f0525;rport=5060 From: "SNOM 360" ;tag=as5623414f To: ;tag=868667120 Call-ID: 334c32b351fe071f0c1130b43dd1eb3a@192.168.125.45 CSeq: 102 INVITE Contact: Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, MESSAGE, INFO Content-Length: 0 --- (9 headers 0 lines)--- -- SIP/1236-433b is making progress passing it to SIP/1239-4477 We're at 192.168.125.45 port 19178 Answering with preferred capability 0x8 (alaw) Answering with non-codec capability 0x1 (telephone-event) Transmitting (no NAT) to 192.168.125.48:2051: SIP/2.0 183 Session Progress Via: SIP/2.0/UDP 192.168.125.48:2051;branch=z9hG4bK-sewfusr4r1t2 From: ;tag=te9v8vdq46 To: ;tag=as572a19f0 Call-ID: 3c26b3ec1388-ppbzigkafn6q@snom360 CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Contact: Content-Type: application/sdp Content-Length: 220 v=0 o=root 12153 12153 IN IP4 192.168.125.45 s=session c=IN IP4 192.168.125.45 t=0 0 m=audio 19178 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- <-- SIP read from 192.168.125.45:5068: SIP/2.0 486 Busy Here Via: SIP/2.0/UDP 192.168.125.45:5060;branch=z9hG4bK209f0525;rport=5060 From: "SNOM 360" ;tag=as5623414f To: ;tag=868667120 Call-ID: 334c32b351fe071f0c1130b43dd1eb3a@192.168.125.45 CSeq: 102 INVITE Allow: INVITE, ACK, OPTIONS, CANCEL, BYE, SUBSCRIBE, NOTIFY, MESSAGE, INFO Content-Length: 0 --- (8 headers 0 lines)--- -- Got SIP response 486 "Busy Here" back from 192.168.125.45 Transmitting (no NAT) to 192.168.125.45:5068: ACK sip:1236@192.168.125.45:5068 SIP/2.0 Via: SIP/2.0/UDP 192.168.125.45:5060;branch=z9hG4bK209f0525;rport From: "SNOM 360" ;tag=as5623414f To: ;tag=868667120 Contact: Call-ID: 334c32b351fe071f0c1130b43dd1eb3a@192.168.125.45 CSeq: 102 ACK User-Agent: Asterisk PBX Content-Length: 0 --- -- SIP/1236-433b is busy == Everyone is busy/congested at this time (1:1/0/0) -- Executing Goto("SIP/1239-4477", "dial-BUSY|1|1") in new stack -- Goto (dial-BUSY,1,1) -- Executing Goto("SIP/1239-4477", "default|4|1") in new stack -- Goto (default,4,1) -- Executing Dial("SIP/1239-4477", "SIP/1240|20|tT") in new stack Destroying call '334c32b351fe071f0c1130b43dd1eb3a@192.168.125.45' We're at 192.168.125.45 port 19078 Answering/Requesting with root capability 0x8 (alaw) Answering with non-codec capability 0x1 (telephone-event) 12 headers, 10 lines Reliably Transmitting (no NAT) to 192.168.125.49:2051: INVITE sip:1240@192.168.125.49:2051;line=4xgcbr7c SIP/2.0 Via: SIP/2.0/UDP 192.168.125.45:5060;branch=z9hG4bK101fe4a4 From: "SNOM 360" ;tag=as6031442f To: Contact: Call-ID: 25472bdd242a588a5319689a7574173f@192.168.125.45 CSeq: 102 INVITE User-Agent: Asterisk PBX Date: Wed, 24 Aug 2005 10:03:39 GMT Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Content-Type: application/sdp Content-Length: 220 v=0 o=root 12153 12153 IN IP4 192.168.125.45 s=session c=IN IP4 192.168.125.45 t=0 0 m=audio 19078 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- -- Called 1240 <-- SIP read from 192.168.125.49:2051: SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.125.45:5060;branch=z9hG4bK101fe4a4 From: "SNOM 360" ;tag=as6031442f To: ;tag=85lla8n5ee Call-ID: 25472bdd242a588a5319689a7574173f@192.168.125.45 CSeq: 102 INVITE Contact: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO Allow-Events: talk, hold, refer Content-Length: 0 --- (10 headers 0 lines)--- -- SIP/1240-eb04 is ringing Transmitting (no NAT) to 192.168.125.48:2051: SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.125.48:2051;branch=z9hG4bK-sewfusr4r1t2 From: ;tag=te9v8vdq46 To: ;tag=as572a19f0 Call-ID: 3c26b3ec1388-ppbzigkafn6q@snom360 CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Contact: Content-Length: 0 X-Asterisk-HangupCause:: User busy --- <-- SIP read from 192.168.125.49:2051: SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.125.45:5060;branch=z9hG4bK101fe4a4 From: "SNOM 360" ;tag=as6031442f To: ;tag=85lla8n5ee Call-ID: 25472bdd242a588a5319689a7574173f@192.168.125.45 CSeq: 102 INVITE Contact: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO Allow-Events: talk, hold, refer Content-Length: 0 --- (10 headers 0 lines)--- -- SIP/1240-eb04 is ringing <-- SIP read from 192.168.125.49:2051: SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.125.45:5060;branch=z9hG4bK101fe4a4 From: "SNOM 360" ;tag=as6031442f To: ;tag=85lla8n5ee Call-ID: 25472bdd242a588a5319689a7574173f@192.168.125.45 CSeq: 102 INVITE Contact: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO Allow-Events: talk, hold, refer Content-Length: 0 --- (10 headers 0 lines)--- -- SIP/1240-eb04 is ringing <-- SIP read from 192.168.125.49:2051: SIP/2.0 180 Ringing Via: SIP/2.0/UDP 192.168.125.45:5060;branch=z9hG4bK101fe4a4 From: "SNOM 360" ;tag=as6031442f To: ;tag=85lla8n5ee Call-ID: 25472bdd242a588a5319689a7574173f@192.168.125.45 CSeq: 102 INVITE Contact: Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO Allow-Events: talk, hold, refer Content-Length: 0 --- (10 headers 0 lines)--- -- SIP/1240-eb04 is ringing <-- SIP read from 192.168.125.49:2051: SIP/2.0 200 Ok Via: SIP/2.0/UDP 192.168.125.45:5060;branch=z9hG4bK101fe4a4 From: "SNOM 360" ;tag=as6031442f To: ;tag=85lla8n5ee Call-ID: 25472bdd242a588a5319689a7574173f@192.168.125.45 CSeq: 102 INVITE Contact: User-Agent: snom320/3.60p Allow: INVITE, ACK, CANCEL, BYE, REFER, OPTIONS, NOTIFY, SUBSCRIBE, PRACK, MESSAGE, INFO Allow-Events: talk, hold, refer Supported: timer, 100rel, replaces Content-Type: application/sdp Content-Length: 212 v=0 o=root 1159212805 1159212806 IN IP4 192.168.125.49 s=call c=IN IP4 192.168.125.49 t=0 0 m=audio 62930 RTP/AVP 8 101 a=rtpmap:8 pcma/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-15 a=sendrecv --- (13 headers 10 lines)--- Found RTP audio format 8 Found RTP audio format 101 Peer audio RTP is at port 192.168.125.49:62930 Found description format pcma Found description format telephone-event Capabilities: us - 0x8 (alaw), peer - audio=0x8 (alaw)/video=0x0 (nothing), combined - 0x8 (alaw) Non-codec capabilities: us - 0x1 (telephone-event), peer - 0x1 (telephone-event), combined - 0x1 (telephone-event) list_route: hop: set_destination: Parsing for address/port to send to set_destination: set destination to 192.168.125.49, port 2051 Transmitting (no NAT) to 192.168.125.49:2051: ACK sip:1240@192.168.125.49:2051;line=4xgcbr7c SIP/2.0 Via: SIP/2.0/UDP 192.168.125.45:5060;branch=z9hG4bK2248bf12 From: "SNOM 360" ;tag=as6031442f To: ;tag=85lla8n5ee Contact: Call-ID: 25472bdd242a588a5319689a7574173f@192.168.125.45 CSeq: 102 ACK User-Agent: Asterisk PBX Content-Length: 0 --- -- SIP/1240-eb04 answered SIP/1239-4477 We're at 192.168.125.45 port 19178 Answering with preferred capability 0x8 (alaw) Answering with non-codec capability 0x1 (telephone-event) Reliably Transmitting (no NAT) to 192.168.125.48:2051: SIP/2.0 200 OK Via: SIP/2.0/UDP 192.168.125.48:2051;branch=z9hG4bK-sewfusr4r1t2 From: ;tag=te9v8vdq46 To: ;tag=as572a19f0 Call-ID: 3c26b3ec1388-ppbzigkafn6q@snom360 CSeq: 1 INVITE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Contact: Content-Type: application/sdp Content-Length: 220 v=0 o=root 12153 12154 IN IP4 192.168.125.45 s=session c=IN IP4 192.168.125.45 t=0 0 m=audio 19178 RTP/AVP 8 101 a=rtpmap:8 PCMA/8000 a=rtpmap:101 telephone-event/8000 a=fmtp:101 0-16 a=silenceSupp:off - - - - --- -- Attempting native bridge of SIP/1239-4477 and SIP/1240-eb04 <-- SIP read from 192.168.125.48:2051: ACK sip:1@192.168.125.45 SIP/2.0 Via: SIP/2.0/UDP 192.168.125.48:2051;branch=z9hG4bK-0v2k51swrt6w;rport From: ;tag=te9v8vdq46 To: ;tag=as572a19f0 Call-ID: 3c26b3ec1388-ppbzigkafn6q@snom360 CSeq: 1 ACK Max-Forwards: 70 Contact: Content-Length: 0 --- (9 headers 0 lines)--- <-- SIP read from 127.0.0.1:5068: SUBSCRIBE sip:1238@127.0.0.1 SIP/2.0 Via: SIP/2.0/UDP 127.0.0.1:5068;rport;branch=z9hG4bK1562432833 From: ;tag=1265125155 To: Knut-Håvard Aksnes Call-ID: 2010299813@127.0.0.1 CSeq: 20 SUBSCRIBE Contact: Max-Forwards: 5 Event: presence User-Agent: Linphone-1.0.1/eXosip Expires: 3600 Allow: INVITE, ACK, CANCEL, BYE, OPTIONS, REFER, SUBSCRIBE, NOTIFY, MESSAGE Accept: application/pidf+xml Content-Length: 0 --- (14 headers 0 lines)--- Using latest SUBSCRIBE request as basis request Sending to 127.0.0.1 : 5068 (non-NAT) Found no matching peer or user for '127.0.0.1:5068' Looking for 1238 in default Transmitting (no NAT) to 127.0.0.1:5068: SIP/2.0 404 Not Found Via: SIP/2.0/UDP 127.0.0.1:5068;branch=z9hG4bK1562432833 From: ;tag=1265125155 To: Knut-Håvard Aksnes ;tag=as0066c3d9 Call-ID: 2010299813@127.0.0.1 CSeq: 20 SUBSCRIBE User-Agent: Asterisk PBX Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, NOTIFY Contact: Content-Length: 0 --- Destroying call '2010299813@127.0.0.1'