From: Jo-Philipp Wich Date: Thu, 25 Sep 2008 14:25:18 +0000 (+0000) Subject: * luci/app/asterisk: add english translations X-Git-Tag: 0.9.0~1194 X-Git-Url: https://git.archive.openwrt.org/?p=project%2Fluci.git;a=commitdiff_plain;h=37ff3ad073935bc0ee4e05a2aef28dddc6a9d56f * luci/app/asterisk: add english translations --- diff --git a/applications/luci-asterisk/luasrc/i18n/asterisk.en.lua b/applications/luci-asterisk/luasrc/i18n/asterisk.en.lua new file mode 100644 index 000000000..6c59e6b72 --- /dev/null +++ b/applications/luci-asterisk/luasrc/i18n/asterisk.en.lua @@ -0,0 +1,274 @@ +asterisk_asterisk = "Asterisk General Options" +asterisk_asterisk_agidir = "AGI directory" +asterisk_asterisk_cacherecordfiles = "Cache recorded sound files during recording" +asterisk_asterisk_debug = "Debug Level" +asterisk_asterisk_dontwarn = "Disable some warnings" +asterisk_asterisk_dumpcore = "Dump core on crash" +asterisk_asterisk_highpriority = "High Priority" +asterisk_asterisk_initcrypto = "Initialise Crypto" +asterisk_asterisk_internaltiming = "Use Internal Timing" +asterisk_asterisk_logdir = "Log directory" +asterisk_asterisk_maxcalls = "Maximum number of calls allowed" +asterisk_asterisk_maxload = "Maximum load to stop accepting new calls" +asterisk_asterisk_nocolor = "Disable console colors" +asterisk_asterisk_recordcachedir = "Sound files Cache directory" +asterisk_asterisk_rungroup = "The Group to run as" +asterisk_asterisk_runuser = "The User to run as" +asterisk_asterisk_spooldir = "Voicemail Spool directory" +asterisk_asterisk_systemname = "Prefix UniquID with system name" +asterisk_asterisk_transcodeviasln = "Build transcode paths via SLINEAR, not directly" +asterisk_asterisk_transmitsilenceduringrecord = "Transmit SLINEAR silence while recording a channel" +asterisk_asterisk_verbose = "Verbose Level" +asterisk_asterisk_zone = "Time Zone" + +asterisk_dialplan = "Section dialplan" +asterisk_dialplan_include = "include" + +asterisk_dialplanexten = "Dialplan Extension" + +asterisk_dialplangeneral = "Dialplan General Options" +asterisk_dialplangeneral_allowtransfer = "Allow transfer" +asterisk_dialplangeneral_canreinvite = "Reinvite/redirect media connections" +asterisk_dialplangeneral_clearglobalvars = "Clear global vars" + +asterisk_dialplangoto = "Dialplan Goto" + +asterisk_dialplanmeetme = "Dialplan Conference" + +asterisk_dialplansaytime = "Dialplan Time" + +asterisk_dialplanvoice = "Dialplan Voicemail" + +asterisk_dialzone = "Dial Zones for Dialplan" +asterisk_dialzone_addprefix = "Prefix to add matching dialplans" +asterisk_dialzone_international = "Match International prefix" +asterisk_dialzone_localprefix = "Prefix (0) to add/remove to/from international numbers" +asterisk_dialzone_localzone = "localzone" +asterisk_dialzone_match = "Match plan" +asterisk_dialzone_uses = "Connection to use" + +asterisk_featuremap = "Feature Key maps" +asterisk_featuremap_atxfer = "Attended transfer key" +asterisk_featuremap_blindxfer = "Blind transfer key" +asterisk_featuremap_disconnect = "Key to Disconnect call" +asterisk_featuremap_parkcall = "Key to Park call" + +asterisk_featurepark = "Parking Feature" +asterisk_featurepark_adsipark = "ADSI Park" +asterisk_featurepark_atxfernoanswertimeout = "Attended transfer timeout (sec)" +asterisk_featurepark_automon = "One touch record key" +asterisk_featurepark_context = "Name of call context for parking" +asterisk_featurepark_courtesytone = "Sound file to play to parked caller" +asterisk_featurepark_featuredigittimeout = "Max time (ms) between digits for feature activation" +asterisk_featurepark_findslot = "Method to Find Parking slot" +asterisk_featurepark_parkedmusicclass = "parkedmusicclass" +asterisk_featurepark_parkedplay = "Play courtesy tone to" +asterisk_featurepark_parkenabled = "Enable Parking" +asterisk_featurepark_parkext = "Extension to dial to park" +asterisk_featurepark_parkingtime = "Parking time (secs)" +asterisk_featurepark_parkpos = "Range of extensions for call parking" +asterisk_featurepark_pickupexten = "Pickup extension" +asterisk_featurepark_transferdigittimeout = "Seconds to wait bewteen digits when transferring" +asterisk_featurepark_xferfailsound = "sound when attended transfer is complete" +asterisk_featurepark_xfersound = "Sound when attended transfer fails" + +asterisk_hardwarereboot = "Reload Hardware Config" +asterisk_hardwarereboot_method = "Reboot Method" +asterisk_hardwarereboot_param = "Parameter" + +asterisk_iax = "SIP Connection" +asterisk_iax_alwaysinternational = "Always Dial International" +asterisk_iax_context = "context" +asterisk_iax_countrycode = "Country Code for connection" +asterisk_iax_extension = "Add as Extension" +asterisk_iax_host = "Host name (or blank)" +asterisk_iax_internationalprefix = "International Dial Prefix" +asterisk_iax_prefix = "Dial Prefix (for external line)" +asterisk_iax_secret = "Secret" +asterisk_iax_timeout = "Dial Timeout (sec)" +asterisk_iax_type = "Option type" +asterisk_iax_username = "User name" + +asterisk_iaxgeneral = "IAX General Options" +asterisk_iaxgeneral_allow = "Allow Codecs" +asterisk_iaxgeneral_canreinvite = "Reinvite/redirect media connections" +asterisk_iaxgeneral_static = "Static" +asterisk_iaxgeneral_writeprotect = "Write Protect" + +asterisk_meetme = "Meetme Conference" +asterisk_meetme_adminpin = "Admin PIN" +asterisk_meetme_pin = "Meeting PIN" + +asterisk_meetmegeneral = "Meetme Conference General Options" +asterisk_meetmegeneral_audiobuffers = "Number of 20ms audio buffers to be used" + +asterisk_module = "Modules" +asterisk_module_appalarmreceiver = "Alarm Receiver Application" +asterisk_module_appauthenticate = "Authentication Application" +asterisk_module_appcdr = "Make sure asterisk doesnt save CDR" +asterisk_module_appchanisavail = "Check if channel is available" +asterisk_module_appchanspy = "Listen in on any channel" +asterisk_module_appcontrolplayback = "Control Playback Application" +asterisk_module_appcut = "Cuts up variables" +asterisk_module_appdb = "Database access functions" +asterisk_module_appdial = "Dialing Application" +asterisk_module_appdictate = "Virtual Dictation Machine Application" +asterisk_module_appdirectedpickup = "Directed Call Pickup Support" +asterisk_module_appdirectory = "Extension Directory" +asterisk_module_appdisa = "DISA (Direct Inward System Access) Application" +asterisk_module_appdumpchan = "Dump channel variables Application" +asterisk_module_appecho = "Simple Echo Application" +asterisk_module_appenumlookup = "ENUM Lookup" +asterisk_module_appeval = "Reevaluates strings" +asterisk_module_appexec = "Executes applications" +asterisk_module_appexternalivr = "External IVR application interface" +asterisk_module_appforkcdr = "Fork The CDR into 2 seperate entities" +asterisk_module_appgetcpeid = "Get ADSI CPE ID" +asterisk_module_appgroupcount = "Group Management Routines" +asterisk_module_appices = "Encode and Stream via icecast and ices" +asterisk_module_appimage = "Image Transmission Application" +asterisk_module_applookupblacklist = "Look up Caller*ID name/number from black" +asterisk_module_applookupcidname = "Look up CallerID Name from local databas" +asterisk_module_appmacro = "Extension Macros" +asterisk_module_appmath = "A simple math Application" +asterisk_module_appmd5 = "MD5 checksum Application" +asterisk_module_appmilliwatt = "Digital Milliwatt (mu-law) Test Application" +asterisk_module_appmixmonitor = "Record a call and mix the audio during the recording" +asterisk_module_appparkandannounce = "Call Parking and Announce Application" +asterisk_module_appplayback = "Trivial Playback Application" +asterisk_module_appprivacy = "Require phone number to be entered" +asterisk_module_appqueue = "True Call Queueing" +asterisk_module_apprandom = "Random goto" +asterisk_module_appread = "Read Variable Application" +asterisk_module_appreadfile = "Read in a file" +asterisk_module_apprealtime = "Realtime Data Lookup/Rewrite" +asterisk_module_apprecord = "Trivial Record Application" +asterisk_module_appsayunixtime = "Say time" +asterisk_module_appsenddtmf = "Send DTMF digits Application" +asterisk_module_appsendtext = "Send Text Applications" +asterisk_module_appsetcallerid = "Set CallerID Application" +asterisk_module_appsetcdruserfield = "CDR user field apps" +asterisk_module_appsetcidname = "load => .so ; Set CallerID Name" +asterisk_module_appsetcidnum = "load => .so ; Set CallerID Number" +asterisk_module_appsetrdnis = "Set RDNIS Number" +asterisk_module_appsettransfercapability = "Set ISDN Transfer Capability" +asterisk_module_appsms = "SMS/PSTN handler" +asterisk_module_appsofthangup = "Hangs up the requested channel" +asterisk_module_appstack = "Stack Routines" +asterisk_module_appsystem = "Generic System() application" +asterisk_module_apptalkdetect = "Playback with Talk Detection" +asterisk_module_apptest = "Interface Test Application" +asterisk_module_apptransfer = "Transfer" +asterisk_module_apptxtcidname = "TXTCIDName" +asterisk_module_appurl = "Send URL Applications" +asterisk_module_appuserevent = "Custom User Event Application" +asterisk_module_appverbose = "Send verbose output" +asterisk_module_appvoicemail = "Voicemail" +asterisk_module_appwaitforring = "Waits until first ring after time" +asterisk_module_appwaitforsilence = "Wait For Silence Application" +asterisk_module_appwhile = "While Loops and Conditional Execution" +asterisk_module_cdrcsv = "Comma Separated Values CDR Backend" +asterisk_module_cdrcustom = "Customizable Comma Separated Values CDR Backend" +asterisk_module_cdrmanager = "Asterisk Call Manager CDR Backend" +asterisk_module_cdrmysql = "MySQL CDR Backend" +asterisk_module_cdrpgsql = "PostgreSQL CDR Backend" +asterisk_module_cdrsqlite = "SQLite CDR Backend" +asterisk_module_chanagent = "Agent Proxy Channel" +asterisk_module_chanalsa = "Channel driver for GTalk" +asterisk_module_changtalk = "Channel driver for GTalk" +asterisk_module_chaniax2 = "Option chan_iax2" +asterisk_module_chanlocal = "Local Proxy Channel" +asterisk_module_chansip = "Session Initiation Protocol (SIP)" +asterisk_module_codecamu = "A-law and Mulaw direct Coder/Decoder" +asterisk_module_codecadpcm = "Adaptive Differential PCM Coder/Decoder" +asterisk_module_codecalaw = "A-law Coder/Decoder" +asterisk_module_codecg726 = "ITU G.726-32kbps G726 Transcoder" +asterisk_module_codecgsm = "GSM/PCM16 (signed linear) Codec Translation" +asterisk_module_codecspeex = "Speex/PCM16 (signed linear) Codec Translator" +asterisk_module_codeculaw = "Mu-law Coder/Decoder" +asterisk_module_formatau = "Sun Microsystems AU format (signed linear)" +asterisk_module_formatg723 = "G.723.1 Simple Timestamp File Format" +asterisk_module_formatg726 = "Raw G.726 (16/24/32/40kbps) data" +asterisk_module_formatg729 = "Raw G729 data" +asterisk_module_formatgsm = "Raw GSM data" +asterisk_module_formath263 = "Raw h263 data" +asterisk_module_formatjpeg = "JPEG (Joint Picture Experts Group) Image" +asterisk_module_formatpcm = "Raw uLaw 8khz Audio support (PCM)" +asterisk_module_formatpcmalaw = "load => .so ; Raw aLaw 8khz PCM Audio support" +asterisk_module_formatsln = "Raw Signed Linear Audio support (SLN)" +asterisk_module_formatvox = "Dialogic VOX (ADPCM) File Format" +asterisk_module_formatwav = "Microsoft WAV format (8000hz Signed Line" +asterisk_module_formatwavgsm = "Microsoft WAV format (Proprietary GSM)" +asterisk_module_funccallerid = "Caller ID related dialplan functions" +asterisk_module_funcenum = "ENUM Functions" +asterisk_module_funcuri = "URI encoding / decoding functions" +asterisk_module_pbxael = "Asterisk Extension Language Compiler" +asterisk_module_pbxconfig = "Text Extension Configuration" +asterisk_module_pbxfunctions = "load => .so ; Builtin dialplan functions" +asterisk_module_pbxloopback = "Loopback Switch" +asterisk_module_pbxrealtime = "Realtime Switch" +asterisk_module_pbxspool = "Outgoing Spool Support" +asterisk_module_pbxwilcalu = "Wil Cal U (Auto Dialer)" +asterisk_module_resconfigmysql = "MySQL Config Resource" +asterisk_module_resconfigodbc = "ODBC Config Resource" +asterisk_module_resconfigpgsql = "PGSQL Module" +asterisk_module_rescrypto = "Cryptographic Digital Signatures" +asterisk_module_resfeatures = "Call Parking Resource" +asterisk_module_resindications = "Indications Configuration" +asterisk_module_resmonitor = "Call Monitoring Resource" +asterisk_module_resmusiconhold = "Music On Hold Resource" +asterisk_module_resodbc = "ODBC Resource" +asterisk_module_ressmdi = "SMDI Module" +asterisk_module_ressnmp = "SNMP Module" + +asterisk_moh = "Music On Hold" +asterisk_moh_application = "Application" +asterisk_moh_directory = "Directory of Music" +asterisk_moh_mode = "Option mode" +asterisk_moh_random = "Random Play" + +asterisk_sip = "SIP Connection" +asterisk_sip_alwaysinternational = "Always Dial International" +asterisk_sip_canreinvite = "Reinvite/redirect media connections" +asterisk_sip_context = "context" +asterisk_sip_countrycode = "Country Code for connection" +asterisk_sip_dtmfmode = "DTMF mode" +asterisk_sip_extension = "Add as Extension" +asterisk_sip_fromdomain = "Primary domain identity for From: headers" +asterisk_sip_fromuser = "From user (required by many SIP providers)" +asterisk_sip_host = "Host name (or blank)" +asterisk_sip_incoming = "Ring on incoming dialplan contexts" +asterisk_sip_insecure = "Allow Insecure for" +asterisk_sip_internationalprefix = "International Dial Prefix" +asterisk_sip_mailbox = "Mailbox for MWI" +asterisk_sip_nat = "NAT between phone and Asterisk" +asterisk_sip_pedantic = "Check tags in headers" +asterisk_sip_port = "SIP Port" +asterisk_sip_prefix = "Dial Prefix (for external line)" +asterisk_sip_qualify = "Reply Timeout (ms) for down connection" +asterisk_sip_register = "Register connection" +asterisk_sip_secret = "Secret" +asterisk_sip_selfmailbox = "Dial own extension for mailbox" +asterisk_sip_timeout = "Dial Timeout (sec)" +asterisk_sip_type = "Client Type" +asterisk_sip_username = "Username" + +asterisk_sipgeneral = "Section sipgeneral" +asterisk_sipgeneral_allow = "Allow codecs" +asterisk_sipgeneral_port = "SIP Port" +asterisk_sipgeneral_realm = "SIP realm" + +asterisk_voicegeneral = "Voicemail general options" +asterisk_voicegeneral_serveremail = "From Email address of server" + +asterisk_voicemail = "Voice Mail boxes" +asterisk_voicemail_attach = "Email contains attachment" +asterisk_voicemail_email = "Email" +asterisk_voicemail_name = "Display Name" +asterisk_voicemail_password = "Password" +asterisk_voicemail_zone = "zone" + +asterisk_voicezone = "Voice Zone settings" +asterisk_voicezone_message = "Message Format" +asterisk_voicezone_zone = "Time Zone" + diff --git a/applications/luci-asterisk/luasrc/i18n/asterisk.en.xml b/applications/luci-asterisk/luasrc/i18n/asterisk.en.xml new file mode 100644 index 000000000..13cba13e5 --- /dev/null +++ b/applications/luci-asterisk/luasrc/i18n/asterisk.en.xml @@ -0,0 +1,257 @@ + + + + +Asterisk General Options +AGI directory +Cache recorded sound files during recording +Debug Level +Disable some warnings +Dump core on crash +High Priority +Initialise Crypto +Use Internal Timing +Log directory +Maximum number of calls allowed +Maximum load to stop accepting new calls +Disable console colors +Sound files Cache directory +The Group to run as +The User to run as +Voicemail Spool directory +Prefix UniquID with system name +Build transcode paths via SLINEAR, not directly +Transmit SLINEAR silence while recording a channel +Verbose Level +Time Zone +Section dialplan +include +Dialplan Extension +Dialplan General Options +Allow transfer +Reinvite/redirect media connections +Clear global vars +Dialplan Goto +Dialplan Conference +Dialplan Time +Dialplan Voicemail +Dial Zones for Dialplan +Prefix to add matching dialplans +Match International prefix +Prefix (0) to add/remove to/from international numbers +localzone +Match plan +Connection to use +Feature Key maps +Attended transfer key +Blind transfer key +Key to Disconnect call +Key to Park call +Parking Feature +ADSI Park +Attended transfer timeout (sec) +One touch record key +Name of call context for parking +Sound file to play to parked caller +Max time (ms) between digits for feature activation +Method to Find Parking slot +parkedmusicclass +Play courtesy tone to +Enable Parking +Extension to dial to park +Parking time (secs) +Range of extensions for call parking +Pickup extension +Seconds to wait bewteen digits when transferring +sound when attended transfer is complete +Sound when attended transfer fails +Reload Hardware Config +Reboot Method +Parameter +SIP Connection +Always Dial International +context +Country Code for connection +Add as Extension +Host name (or blank) +International Dial Prefix +Dial Prefix (for external line) +Secret +Dial Timeout (sec) +Option type +User name +IAX General Options +Allow Codecs +Reinvite/redirect media connections +Static +Write Protect +Meetme Conference +Admin PIN +Meeting PIN +Meetme Conference General Options +Number of 20ms audio buffers to be used +Modules +Alarm Receiver Application +Authentication Application +Make sure asterisk doesnt save CDR +Check if channel is available +Listen in on any channel +Control Playback Application +Cuts up variables +Database access functions +Dialing Application +Virtual Dictation Machine Application +Directed Call Pickup Support +Extension Directory +DISA (Direct Inward System Access) Application +Dump channel variables Application +Simple Echo Application +ENUM Lookup +Reevaluates strings +Executes applications +External IVR application interface +Fork The CDR into 2 seperate entities +Get ADSI CPE ID +Group Management Routines +Encode and Stream via icecast and ices +Image Transmission Application +Look up Caller*ID name/number from black +Look up CallerID Name from local databas +Extension Macros +A simple math Application +MD5 checksum Application +Digital Milliwatt (mu-law) Test Application +Record a call and mix the audio during the recording +Call Parking and Announce Application +Trivial Playback Application +Require phone number to be entered +True Call Queueing +Random goto +Read Variable Application +Read in a file +Realtime Data Lookup/Rewrite +Trivial Record Application +Say time +Send DTMF digits Application +Send Text Applications +Set CallerID Application +CDR user field apps +load => .so ; Set CallerID Name +load => .so ; Set CallerID Number +Set RDNIS Number +Set ISDN Transfer Capability +SMS/PSTN handler +Hangs up the requested channel +Stack Routines +Generic System() application +Playback with Talk Detection +Interface Test Application +Transfer +TXTCIDName +Send URL Applications +Custom User Event Application +Send verbose output +Voicemail +Waits until first ring after time +Wait For Silence Application +While Loops and Conditional Execution +Comma Separated Values CDR Backend +Customizable Comma Separated Values CDR Backend +Asterisk Call Manager CDR Backend +MySQL CDR Backend +PostgreSQL CDR Backend +SQLite CDR Backend +Agent Proxy Channel +Channel driver for GTalk +Channel driver for GTalk +Option chan_iax2 +Local Proxy Channel +Session Initiation Protocol (SIP) +Adaptive Differential PCM Coder/Decoder +A-law Coder/Decoder +A-law and Mulaw direct Coder/Decoder +ITU G.726-32kbps G726 Transcoder +GSM/PCM16 (signed linear) Codec Translation +Speex/PCM16 (signed linear) Codec Translator +Mu-law Coder/Decoder +Sun Microsystems AU format (signed linear) +G.723.1 Simple Timestamp File Format +Raw G.726 (16/24/32/40kbps) data +Raw G729 data +Raw GSM data +Raw h263 data +JPEG (Joint Picture Experts Group) Image +Raw uLaw 8khz Audio support (PCM) +load => .so ; Raw aLaw 8khz PCM Audio support +Raw Signed Linear Audio support (SLN) +Dialogic VOX (ADPCM) File Format +Microsoft WAV format (8000hz Signed Line +Microsoft WAV format (Proprietary GSM) +Caller ID related dialplan functions +ENUM Functions +URI encoding / decoding functions +Asterisk Extension Language Compiler +Text Extension Configuration +load => .so ; Builtin dialplan functions +Loopback Switch +Realtime Switch +Outgoing Spool Support +Wil Cal U (Auto Dialer) +MySQL Config Resource +ODBC Config Resource +PGSQL Module +Cryptographic Digital Signatures +Call Parking Resource +Indications Configuration +Call Monitoring Resource +Music On Hold Resource +ODBC Resource +SMDI Module +SNMP Module +Music On Hold +Application +Directory of Music +Option mode +Random Play +SIP Connection +Always Dial International +Reinvite/redirect media connections +context +Country Code for connection +DTMF mode +Add as Extension +Primary domain identity for From: headers +From user (required by many SIP providers) +Host name (or blank) +Ring on incoming dialplan contexts +Allow Insecure for +International Dial Prefix +Mailbox for MWI +NAT between phone and Asterisk +Check tags in headers +SIP Port +Dial Prefix (for external line) +Reply Timeout (ms) for down connection +Register connection +Secret +Dial own extension for mailbox +Dial Timeout (sec) +Client Type +Username +Section sipgeneral +Allow codecs +SIP Port +SIP realm +Voicemail general options +From Email address of server +Voice Mail boxes +Email contains attachment +Email +Display Name +Password +zone +Voice Zone settings +Message Format +Time Zone + +