http://www.voip-info.org/wiki/view/Asterisk+Detailed+Variable+List

在编写asterisk拨号规则时，需要使用到各种系统变量和程序变量。

这个教程将讲述这些变量的使用方式。

 

原文见asterisk官方文档。

Asterisk standard channel variables 

There are a number of variables that are defined or read 

by Asterisk. Here is a list of them. More information is 

available in each application's help text. All these variables 

are in UPPER CASE only. 


to diaplay all channel variable use this command 


DumpChan(<min_verbose_level>) 


Variables marked with a * are builtin functions and can't be set, 

only read in the dialplan. Writes to such variables are silently 

ignored. 


${ACCOUNTCODE} * Account code (if specified) (Deprecated; use ${CDR(accountcode)}) 

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

${BRIDGEPEER} Bridged peer 

${CALLERANI} * Caller ANI (PRI channels) (Deprecated; use ${CALLERID(ani)}) 

${CALLERID} * Caller ID (Deprecated; use ${CALLERID(all)}) 

${CALLERIDNAME} * Caller ID Name only (Deprecated; use ${CALLERID(name)}) 

${CALLERIDNUM} * Caller ID Number only (Deprecated; use ${CALLERID(num)}) 

${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  //主号通道号。

${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 

${DNID} * Dialed Number Identifier (Deprecated; use ${CALLERID(dnid)}) 

${EPOCH} * Current unix style epoch 

${EXTEN} * Current extension   //当前分机号或拨号。

${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 

${RDNIS} * Redirected Dial Number ID Service (Deprecated; use ${CALLERID(rdnis)}) 

${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 


Application return values 

In Asterisk 1.2, many applications return the result in a variable 

instead of, as in Asterisk 1.0, changing the dial plan priority (+101). 

For the various status values, see each application's help text. 


${AGISTATUS} * agi() 

${AQMSTATUS} * addqueuemember() 

${AVAILSTATUS} * chanisavail() 

${CHECKGROUPSTATUS} * checkgroup() 

${CHECKMD5STATUS} * checkmd5() 

${CPLAYBACKSTATUS} * controlplayback() 

${DIALSTATUS} * dial() - see also ${HANGUPCAUSE} 

${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() 



Various application variables 

${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 

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

(TIMEOUT | FULL | JOINEMPTY | LEAVEEMPTY | JOINUNAVAIL | LEAVEUNAVAIL) 

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

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

${TXTCIDNAME} * Result of application TXTCIDName 

${VPB_GETDTMF} chan_vpb 


The MeetMe Conference Bridge uses the following variables: 

${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 (zap only) 

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


The VoiceMail() application uses the following variables: 

${VM_CATEGORY} * Sets voicemail category 

${VM_NAME} * Full name in voicemail 

${VM_MAILBOX} * Mailbox Number 

${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 


The VMAuthenticate() application uses the following variables: 

${AUTH_MAILBOX} * Authenticated mailbox 

${AUTH_CONTEXT} * Authenticated mailbox context 


DUNDiLookup() uses the following variables 

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

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


The Zaptel channel sets the following variables: 

${ANI2} * The ANI2 Code provided by the network on the incoming call. (ie, Code 29 identifies call as a Prison/Inmate Call) See also: NANPA ANI II Digits Assignments 

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

${CALLEDTON} * Dialplan for called number on PRI/BRI calls (17=international, 33=national, 65=local, 73=private, 0=unknown). Note: this is a misnomer, TON != dialplan. 

${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' 


The SIP channel uses the following variables: 

${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 

${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 


The Agent channel uses the following variables: 

${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 


The Dial() application uses the following variables: 

${DIALEDPEERNAME} * Dialed peer name 

${DIALEDPEERNUMBER} * Dialed peer number 

${DIALEDTIME} * Time for the call (seconds) 

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

${DIALSTATUS} * Status of the call, one of: 

                (CHANUNAVAIL | CONGESTION | BUSY | NOANSWER  

| ANSWER | CANCEL | DONTCALL | TORTURE) 

${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


The chanisavail() application sets the following variables: 

${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 


When using macros in the dialplan, these variables are available 

${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 


The ChanSpy() application uses the following variables: 

${SPYGROUP} * A ':' (colon) separated list of group names. 

(To be set on spied on channel and matched against the g(grp) option) 


If you compile with OSP support, these variables are used: 

${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 


Call File extension variables: 

${REASON} The reason why an auto-dialout call failed 





CDR Variables 


If the channel has a cdr, that cdr record has it's own set of variables which 

can be accessed just like channel variables. The following builtin variables 

are available and, unless specified, read-only. 


${CDR(clid)} Caller ID 

${CDR(src)} Source 

${CDR(dst)} Destination 

${CDR(dcontext)} Destination context 

${CDR(channel)} Channel name 

${CDR(dstchannel)} Destination channel 

${CDR(lastapp)} Last app executed 

${CDR(lastdata)} Last app's arguments 

${CDR(start)} Time the call started. 

${CDR(answer)} Time the call was answered. 

${CDR(end)} Time the call ended. 

${CDR(duration)} Duration of the call. 

${CDR(billsec)} Duration of the call once it was answered. 

${CDR(disposition)} ANSWERED, NO ANSWER, BUSY 

${CDR(amaflags)} DOCUMENTATION, BILL, IGNORE etc 

${CDR(accountcode)} The channel's account code (read-write). 

${CDR(uniqueid)} The channel's unique id. 

${CDR(userfield)} The channels uses specified field (read-write). 



In addition, you can set your own extra variables with a traditional 

Set(CDR(var)=val) to anything you want. 


NOTE Some CDR values (eg: duration & billsec) can't be accessed until the call has terminated. As of 91617, those values will be calculated on-demand if requested. Until that makes it into a stable release, you can set endbeforehexten=yes in cdr.conf, and then use the "hangup" context to wrap up your call. 


Certain functional variables may be accessed with ${foo(<args>)}. A list 

of these functional variables may be found by typing "show functions" 

at the Asterisk CLI.