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分类: 系统运维

2010-03-05 10:02:33

标准通道变量(Asterisk standard channel variables注:变量后面就不翻译了,保留英文更好理解使用



${CDR(accountcode)}    * Account code (if specified)

${BLINDTRANSFER}         The name of the channel on the other side of a blind transfer

${BRIDGEPEER}            Bridged peer

${BRIDGEPVTCALLID}       Bridged peer PVT call ID (SIP Call ID if a SIP call)

${CALLERID(ani)}       * Caller ANI (PRI channels)

${CALLERID(ani2)}      * ANI2 (Info digits) also called Originating line information or OLI

${CALLERID(all)}       * Caller ID

${CALLERID(dnid)}      * Dialed Number Identifier

${CALLERID(name)}      * Caller ID Name only

${CALLERID(num)}       * Caller ID Number only

${CALLERID(rdnis)}     * Redirected Dial Number ID Service

${CALLINGANI2}         * Caller ANI2 (PRI channels)

${CALLINGPRES}         * Caller ID presentation for incoming calls (PRI channels)

${CALLINGTNS}          * Transit Network Selector (PRI channels)

${CALLINGTON}          * Caller Type of Number (PRI channels)

${CHANNEL}             * Current channel name

${CONTEXT}             * Current context

${DATETIME}            * Current date time in the format: DDMMYYYY-HH:MM:SS

                         (Deprecated; use ${STRFTIME(${EPOCH},,%d%m%Y-%H:%M:%S)})

${DB_RESULT}             Result value of DB_EXISTS() dial plan function

${EPOCH}               * Current unix style epoch

${EXTEN}               * Current extension

${ENV(VAR)}              Environmental variable VAR

${GOTO_ON_BLINDXFR}      Transfer to the specified context/extension/priority

                         after a blind transfer (use ^ characters in place of

                         | to separate context/extension/priority when setting

                         this variable from the dialplan)

${HANGUPCAUSE}         * Asterisk cause of hangup (inbound/outbound)

${HINT}                * Channel hints for this extension

${HINTNAME}            * Suggested Caller*ID name for this extension

${INVALID_EXTEN}         The invalid called extension (used in the "i" extension)

${LANGUAGE}            * Current language (Deprecated; use ${LANGUAGE()})

${LEN(VAR)}            * String length of VAR (integer)

${PRIORITY}            * Current priority in the dialplan

${PRIREDIRECTREASON}     Reason for redirect on PRI, if a call was directed

${TIMESTAMP}           * Current date time in the format: YYYYMMDD-HHMMSS

                         (Deprecated; use ${STRFTIME(${EPOCH},,%Y%m%d-%H%M%S)})

${TRANSFER_CONTEXT}      Context for transferred calls

${FORWARD_CONTEXT}       Context for forwarded calls

${UNIQUEID}            * Current call unique identifier

${SYSTEMNAME}          * value of the systemname option of asterisk.conf

${ENTITYID}            * Global Entity ID set automatically, or from asterisk.conf

的返回值(Application return values


${AGISTATUS}         * agi()

${AQMSTATUS}         * addqueuemember()

${AVAILSTATUS}       * chanisavail()

${CHECKGROUPSTATUS}  * checkgroup()

${CHECKMD5STATUS}    * checkmd5()

${CPLAYBACKSTATUS}   * controlplayback()

${DIALSTATUS}        * dial()

${DBGETSTATUS}       * dbget()

${ENUMSTATUS}        * enumlookup()

${HASVMSTATUS}       * hasnewvoicemail()

${LOOKUPBLSTATUS}    * lookupblacklist()

${OSPAUTHSTATUS}     * ospauth()

${OSPLOOKUPSTATUS}   * osplookup()

${OSPNEXTSTATUS}     * ospnext()

${OSPFINISHSTATUS}   * ospfinish()

${PARKEDAT}          * parkandannounce()

${PLAYBACKSTATUS}    * playback()

${PQMSTATUS}         * pausequeuemember()

${PRIVACYMGRSTATUS}  * privacymanager()

${QUEUESTATUS}       * queue()

${RQMSTATUS}         * removequeuemember()

${SENDIMAGESTATUS}   * sendimage()

${SENDTEXTSTATUS}    * sendtext()

${SENDURLSTATUS}     * sendurl()

${SYSTEMSTATUS}      * system()

${TRANSFERSTATUS}    * transfer()

${TXTCIDNAMESTATUS}  * txtcidname()

${UPQMSTATUS}        * unpausequeuemember()

${VMSTATUS}          * voicmail()

${VMBOXEXISTSSTATUS} * vmboxexists()

${WAITSTATUS}        * waitforsilence()

${CURL}                 * Resulting page content for curl()

${ENUM}                 * Result of application EnumLookup

${EXITCONTEXT}            Context to exit to in IVR menu (app background())

                          or in the RetryDial() application

${MONITOR}              * Set to "TRUE" if the channel is/has been monitored (app monitor())

${MONITOR_EXEC}           Application to execute after monitoring a call

${MONITOR_EXEC_ARGS}      Arguments to application

${MONITOR_FILENAME}       File for monitoring (recording) calls in queue

${QUEUE_PRIO}             Queue priority

${QUEUE_MAX_PENALTY}      Maximum member penalty allowed to answer caller

${QUEUE_MIN_PENALTY}      Minimum member penalty allowed to answer caller

${QUEUESTATUS}            Status of the call, one of:


${RECORDED_FILE}        * Recorded file in record()

${TALK_DETECTED}        * Result from talkdetect()

${TOUCH_MONITOR}          The filename base to use with Touch Monitor (auto record)

${TOUCH_MONITOR_PREF}   * The prefix for automonitor recording filenames.

${TOUCH_MONITOR_FORMAT}   The audio format to use with Touch Monitor (auto record)

${TOUCH_MONITOR_OUTPUT} * Recorded file from Touch Monitor (auto record)

${TOUCH_MONITOR_MESSAGE_START} Recorded file to play for both channels at start of monitoring session

${TOUCH_MONITOR_MESSAGE_STOP} Recorded file to play for both channels at end of monitoring session

${TXTCIDNAME}           * Result of application TXTCIDName

${VPB_GETDTMF}            chan_vpb

会议桥[会议电话桥分器]The MeetMe Conference Bridge

${MEETME_RECORDINGFILE}      Name of file for recording a conference with the "r" option

${MEETME_RECORDINGFORMAT}    Format of file to be recorded

${MEETME_EXIT_CONTEXT}       Context for exit out of meetme meeting

${MEETME_AGI_BACKGROUND}     AGI script for Meetme (DAHDI only)

${MEETMESECS}              * Number of seconds a user participated in a MeetMe conference

${CONF_LIMIT_TIMEOUT_FILE}   File to play when time is up. Used with the L() option.

${CONF_LIMIT_WARNING_FILE}   File to play as warning if 'y' is defined. The default is to say the time remaining.  Used with the L() option.

${VM_CATEGORY}      Sets voicemail category

${VM_NAME}        * Full name in voicemail

${VM_DUR}         * Voicemail duration

${VM_MSGNUM}      * Number of voicemail message in mailbox

${VM_CALLERID}    * Voicemail Caller ID (Person leaving vm)

${VM_CIDNAME}     * Voicemail Caller ID Name

${VM_CIDNUM}      * Voicemail Caller ID Number

${VM_DATE}        * Voicemail Date

${VM_MESSAGEFILE} * Path to message left by caller

${AUTH_MAILBOX}   * Authenticated mailbox

${AUTH_CONTEXT}   * Authenticated mailbox context

${DUNDTECH}       * The Technology of the result from a call to DUNDiLookup()

${DUNDDEST}       * The Destination of the result from a call to DUNDiLookup()

${ANI2}               * The ANI2 Code provided by the network on the incoming call. (ie, Code 29 identifies call as a Prison/Inmate Call)

${CALLTYPE}           * Type of call (Speech, Digital, etc)

${CALLEDTON}          * Type of number for incoming PRI extension i.e. 0=unknown, 1=international, 2=domestic, 3=net_specific, 4=subscriber, 6=abbreviated, 7=reserved

${CALLINGSUBADDR}     * Called PRI Subaddress

${FAXEXTEN}           * The extension called before being redirected to "fax"

${PRIREDIRECTREASON}  * Reason for redirect, if a call was directed

${SMDI_VM_TYPE}       * When an call is received with an SMDI message, the 'type' of message 'b' or 'u'

${SIPCALLID}         * SIP Call-ID: header verbatim (for logging or CDR matching)

${SIPDOMAIN}         * SIP destination domain of an inbound call (if appropriate)

${SIPUSERAGENT}      * SIP user agent (deprecated)

${SIPURI}            * SIP uri

${SIP_CODEC}           Set the SIP codec for a call

${SIP_URI_OPTIONS}   * additional options to add to the URI for an outgoing call

${RTPAUDIOQOS}         RTCP QoS report for the audio of this call

${RTPVIDEOQOS}         RTCP QoS report for the video of this call

${AGENTMAXLOGINTRIES}  Set the maximum number of failed logins

${AGENTUPDATECDR}      Whether to update the CDR record with Agent channel data

${AGENTGOODBYE}        Sound file to use for "Good Bye" when agent logs out

${AGENTACKCALL}        Whether the agent should acknowledge the incoming call

${AGENTAUTOLOGOFF}     Auto logging off for an agent

${AGENTWRAPUPTIME}     Setting the time for wrapup between incoming calls

${AGENTNUMBER}       * Agent number (username) set at login

${AGENTSTATUS}       * Status of login ( fail | on | off )

${AGENTEXTEN}        * Extension for logged in agent

${DIALEDPEERNAME}     * Dialed peer name

${DIALEDPEERNUMBER}   * Dialed peer number

${DIALEDTIME}         * Time for the call (seconds). Is only set if call was answered.

${ANSWEREDTIME}       * Time from answer to hangup (seconds)


${DYNAMIC_FEATURES}   * The list of features (from the [applicationmap] section of  features.conf) to activate during the call, with feature names separated by '#' characters

${LIMIT_PLAYAUDIO_CALLER}  Soundfile for call limits

${LIMIT_PLAYAUDIO_CALLEE}  Soundfile for call limits

${LIMIT_WARNING_FILE}      Soundfile for call limits

${LIMIT_TIMEOUT_FILE}      Soundfile for call limits

${LIMIT_CONNECT_FILE}      Soundfile for call limits

${OUTBOUND_GROUP}          Default groups for peer channels (as in SetGroup)  * See "show application dial" for more information

${AVAILCHAN}          * the name of the available channel if one was found

${AVAILORIGCHAN}      * the canonical channel name that was used to create the channel

${AVAILSTATUS}        * Status of requested channel

${MACRO_EXTEN}        * The calling extensions

${MACRO_CONTEXT}      * The calling context

${MACRO_PRIORITY}     * The calling priority

${MACRO_OFFSET}         Offset to add to priority at return from macro

${SPYGROUP}           * A ':' (colon) separated list of group names. (To be set on spied on channel and matched against the g(grp) option)

${OSPINHANDLE}          OSP handle of in_bound call

${OSPINTIMELIMIT}       Duration limit for in_bound call

${OSPOUTHANDLE}         OSP handle of out_bound call

${OSPTECH}              OSP technology

${OSPDEST}              OSP destination

${OSPCALLING}           OSP calling number

${OSPOUTTOKEN}          OSP token to use for out_bound call

${OSPOUTTIMELIMIT}      Duration limit for out_bound call

${OSPRESULTS}           Number of remained destinations

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