${DIALSTATUS}
CHANUNAVAIL Channel unavailable. On SIP, peer may not be registered.
CONGESTION Congestion. This status is usually a sign that the dialled number is not recognised.
NOANSWER No answer. The dial command reached its number, the number rang for too long, then the dial timed out.
BUSY Busy signal. The dial command reached its number but the number is busy.
ANSWER Call is answered. A successful dial. The caller reached the callee.
CANCEL Call is cancelled. The dial command reached its number but the caller hung up before the callee picked up.
DONTCALL Privacy mode, callee rejected the call
TORTURE Privacy mode, callee chose to send caller to torture menu
INVALIDARGS Error parsing Dial command arguments (added for Asterisk 1.4.1, SVN r53135-53136)
在Privacy和Screening模式中,被叫选择发送主叫到'Go Away'脚本时状态变为DONTCALL;发送到'torture'脚本时状态变为TORTURE。
被*标记的变量是内建函数,不能在拨号方案中被设置,只能被读取。对这些变量的赋值将被忽略。
${CDR(accountcode)} * Account co
${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 on
${CALLERID(num)} * Caller ID Number on
${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
Application的返回值(Application return values)
当你调用有些application的时候,它们会返回一个值供你读取。对于每一个application,这些状态字段是唯一的。各种状态值,前参考每个application的帮助文档。
${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()
各种application的相关变量(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
${QUEUE_MIN_PENALTY} Minimum member penalty allowed to answer caller
${QUEUESTATUS} Status of the call, on
(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_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
MeetMe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 on
${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.
The VoiceMail() application
${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
The VMAuthenticate() application
${AUTH_MAILBOX} * Authenticated mailbox
${AUTH_CONTEXT} * Authenticated mailbox context
DUNDiLookup()
${DUNDTECH} * The Technology of the result from a call to DUNDiLookup()
${DUNDDEST} * The Destination of the result from a call to DUNDiLookup()
chan_dahdi
${ANI2} * The ANI2 Co
${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'
chan_sip
${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
chan_agent
${AGENTMAXLOGINTRIES} Set the maximum number of failed logins
${AGENTUPDATECDR} Whether to update the CDR record with Agent channel da
${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
${DIALEDPEERNAME} * Dialed peer name
${DIALEDPEERNUMBER} * Dialed peer number
${DIALEDTIME} * Time for the call (seconds). Is on
${ANSWEREDTIME} * Time from answer to hangup (seconds)
${DIALSTATUS} * Status of the call, on
${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
${AVAILCHAN} * the name of the available channel if on
${AVAILORIGCHAN} * the canonical channel name that was used to create the channel
${AVAILSTATUS} * Status of requested channel
拨号方案宏(Dialplan Macros)
${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
${SPYGROUP} * A ':' (colon) separated list of group names. (To be set on spied on channel and matched against the g(grp) option)
OSP
${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