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спросил 2014-01-24 17:18:52 +0400

Diana92 Gravatar Diana92

канальные переменные

Помогите пожалуйста. Смотрю список канальных переменных. Не могу найти переменную которую показывает на какой номер ты звонишь. Не внутренние номера, а именно исходящий ИД номер?Просто создаю таблицу в базе который хранит DTMF выборы, нужно еще там хранить на какой номер был звонок. Эти номеры будут только для ИВР, для голосования. Подскажите? Поискала в гугле. Нашла такой список

Predefined Channel Variables
There are some channel variables set by Asterisk that you can refer to in your dialplan definitions. Asterisk-defined variables, in contrast to user-defined variables, are case sensitive. Note: Several of these builtin variables have been converted to functions in 1.2, to allow setting their values.
${ACCOUNTCODE}: Account code, if specified - see Asterisk billing (DEPRECATED in 1.2.0 and removed in 1.4. Use ${CDR(accountcode)}
${ANSWEREDTIME}: This is the amount of time(in seconds) for actual call.
${BLINDTRANSFER}: The active SIP channel that dialed the number. This will return the SIP Channel that dialed the number when doing blind transfers - see BLINDTRANSFER
${CALLERID(all)}: The current Caller ID name and number - See Setting Callerid for usage in Asterisk 1.4
${CALLERID(name)}: The current Caller ID name - ${CALLERIDNAME} was used in versions of Asterisk prior to 1.2.0, it was DEPRECATED in 1.2.0 and removed in 1.4.
${CALLERID(num)}: The current Caller ID number - ${CALLERIDNUM} was used in versions of Asterisk prior to 1.2.0, it was DEPRECATED in 1.2.0 and removed in 1.4.
(Note: this is not necessarily numeric as the name would indicate and can legitimately contain the space character. Commands acting on this variable (such as 'GotoIf', for example) should be aware of this).
${CALLINGPRES}: PRI Call ID Presentation variable for incoming calls (See callingpres )
${CHANNEL}: Current channel name
${CONTEXT}: The name of the current context
${DATETIME}: Current date time in the format: DDMMYYYY-HH:MM:SS This is deprecated in Asterisk 1.2, instead use :${STRFTIME(${EPOCH},,%d%m%Y-%H:%M:%S)})
${DIALEDPEERNAME}: Name of the called party. Broken for now, see DIALEDPEERNAME
${DIALEDPEERNUMBER}: Number of the called party. Broken for now, see DIALEDPEERNUMBER
${DIALEDTIME}: Time since the number was dialed (only works when dialed party answers the line?!)
${DIALSTATUS}: Status of the call. See DIALSTATUS (note: In the current SVN release, DIALSTATUS seems to have been removed. Now you should use the DEVSTATE function. Try in astersisk console "core show function DEVSTATE" for more informations)
${DNID}: Dialed Number Identifier. Limitations apply, see DNID
${EPOCH}: The current UNIX-style epoch (number of seconds since 1 Jan 1970)
${EXTEN}: The current extension - cannot be modified with the set command- just use the GoTo to change the EXTEN variable!
${HANGUPCAUSE}: The last hangup return code on a Zap channel connected to a PRI interface
${INVALID_EXTEN}: The extension asked for when redirected to the i (invalid) extension
${LANGUAGE}: The current language setting. See Asterisk multi-language
${MEETMESECS}: Number of seconds a user participated in a MeetMe conference
${PRIORITY}: The current priority
${RDNIS}: The current redirecting DNIS, Caller ID that redirected the call. Limitations apply, see RDNIS
${SIPDOMAIN}: SIP destination domain of an inbound call (if appropriate)
${SIP_CODEC}: Set the SIP codec for the inbound (=first) call leg (see channelvariables.txt or README.variables in 1.2); Asterisk 1.6.2 also comes with SIP_CODEC_OUTBOUND for the remote (=second) call leg.
${SIPCALLID}: The SIP dialog Call-ID: header
${SIPUSERAGENT}: The SIP user agent header
${TIMESTAMP}: Current date time in the format: YYYYMMDD-HHMMSS This is deprecated as of Asterisk 1.4, instead use :${STRFTIME(${EPOCH},,%Y%m%d-%H%M%S)})
${TRANSFERCAPABILITY}: Type of Channel
${TXTCIDNAME}: Result of application TXTCIDName (see below)
${UNIQUEID}: Current call unique identifier
${TOUCH_MONITOR}: used for "one touch record" (see features.conf, and wW dial flags). If is set on either side of the call then that var contains the app_args for app_monitor otherwise the default of WAV||m is used
${TOUCH_MONITOR_PREFIX}: used for "one touch record" (see features.conf, and wW dial flags). This set Prefix to ${TOUCH_MONITOR} default: auto "New in 1.8"

Но там не вижу то что мне нужно(

канальные переменные

Помогите пожалуйста. Смотрю список канальных переменных. Не могу найти переменную которую показывает на какой номер ты звонишь. Не внутренние номера, а именно исходящий ИД номер?Просто создаю таблицу в базе который хранит DTMF выборы, нужно еще там хранить на какой номер был звонок. Эти номеры номера будут предназначаться только для ИВР, для голосования. никто телефон поднимать не будет, люди звонят, будет приветствие и меню, где нужно нажать за что ты голосуешь и все. Подскажите? Поискала в гугле. Нашла такой список

Predefined Channel Variables
There are some channel variables set by Asterisk that you can refer to in your dialplan definitions. Asterisk-defined variables, in contrast to user-defined variables, are case sensitive. Note: Several of these builtin variables have been converted to functions in 1.2, to allow setting their values.
${ACCOUNTCODE}: Account code, if specified - see Asterisk billing (DEPRECATED in 1.2.0 and removed in 1.4. Use ${CDR(accountcode)}
${ANSWEREDTIME}: This is the amount of time(in seconds) for actual call.
${BLINDTRANSFER}: The active SIP channel that dialed the number. This will return the SIP Channel that dialed the number when doing blind transfers - see BLINDTRANSFER
${CALLERID(all)}: The current Caller ID name and number - See Setting Callerid for usage in Asterisk 1.4
${CALLERID(name)}: The current Caller ID name - ${CALLERIDNAME} was used in versions of Asterisk prior to 1.2.0, it was DEPRECATED in 1.2.0 and removed in 1.4.
${CALLERID(num)}: The current Caller ID number - ${CALLERIDNUM} was used in versions of Asterisk prior to 1.2.0, it was DEPRECATED in 1.2.0 and removed in 1.4.
(Note: this is not necessarily numeric as the name would indicate and can legitimately contain the space character. Commands acting on this variable (such as 'GotoIf', for example) should be aware of this).
${CALLINGPRES}: PRI Call ID Presentation variable for incoming calls (See callingpres )
${CHANNEL}: Current channel name
${CONTEXT}: The name of the current context
${DATETIME}: Current date time in the format: DDMMYYYY-HH:MM:SS This is deprecated in Asterisk 1.2, instead use :${STRFTIME(${EPOCH},,%d%m%Y-%H:%M:%S)})
${DIALEDPEERNAME}: Name of the called party. Broken for now, see DIALEDPEERNAME
${DIALEDPEERNUMBER}: Number of the called party. Broken for now, see DIALEDPEERNUMBER
${DIALEDTIME}: Time since the number was dialed (only works when dialed party answers the line?!)
${DIALSTATUS}: Status of the call. See DIALSTATUS (note: In the current SVN release, DIALSTATUS seems to have been removed. Now you should use the DEVSTATE function. Try in astersisk console "core show function DEVSTATE" for more informations)
${DNID}: Dialed Number Identifier. Limitations apply, see DNID
${EPOCH}: The current UNIX-style epoch (number of seconds since 1 Jan 1970)
${EXTEN}: The current extension - cannot be modified with the set command- just use the GoTo to change the EXTEN variable!
${HANGUPCAUSE}: The last hangup return code on a Zap channel connected to a PRI interface
${INVALID_EXTEN}: The extension asked for when redirected to the i (invalid) extension
${LANGUAGE}: The current language setting. See Asterisk multi-language
${MEETMESECS}: Number of seconds a user participated in a MeetMe conference
${PRIORITY}: The current priority
${RDNIS}: The current redirecting DNIS, Caller ID that redirected the call. Limitations apply, see RDNIS
${SIPDOMAIN}: SIP destination domain of an inbound call (if appropriate)
${SIP_CODEC}: Set the SIP codec for the inbound (=first) call leg (see channelvariables.txt or README.variables in 1.2); Asterisk 1.6.2 also comes with SIP_CODEC_OUTBOUND for the remote (=second) call leg.
${SIPCALLID}: The SIP dialog Call-ID: header
${SIPUSERAGENT}: The SIP user agent header
${TIMESTAMP}: Current date time in the format: YYYYMMDD-HHMMSS This is deprecated as of Asterisk 1.4, instead use :${STRFTIME(${EPOCH},,%Y%m%d-%H%M%S)})
${TRANSFERCAPABILITY}: Type of Channel
${TXTCIDNAME}: Result of application TXTCIDName (see below)
${UNIQUEID}: Current call unique identifier
${TOUCH_MONITOR}: used for "one touch record" (see features.conf, and wW dial flags). If is set on either side of the call then that var contains the app_args for app_monitor otherwise the default of WAV||m is used
${TOUCH_MONITOR_PREFIX}: used for "one touch record" (see features.conf, and wW dial flags). This set Prefix to ${TOUCH_MONITOR} default: auto "New in 1.8"

Но там не вижу то что мне нужно(

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Asterisk® и Digium® являются зарегистрированными торговыми марками компании Digium, Inc., США.
IP АТС Asterisk распространяется под лицензией GNU GPL.