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asterisk App Dial 拨号 发起外呼 -
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asterisk App Dial 拨号 发起外呼 -
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这种做法在Server端getValue()方法是能获得修改后 ...
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Asterisk Detailed Variable List
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
${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
${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_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)
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.
发表评论
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解决elastix登录显示非常慢以及禁用新闻的展示的问题
2011-06-10 11:12 1253解决登录显示非常慢的问题以及禁用新闻的展示 ... -
FreePBX 2.7.0.3 汉化
2011-06-09 15:46 18191,FreePBX 2.7.0.3 右侧语言选项,默认无 ... -
Asterisk 1.6的配置文件:chan_dahdi.conf
2010-12-24 13:56 2165Asterisk 1.6的配置文件:chan_dahdi.co ... -
Asterisk 安装笔记(2)- Zaptel 和 Dahdi 的配置
2010-12-24 13:53 2028Zap Channel Module Configurat ... -
Asterisk Extension中的Application命令详解七
2010-12-22 09:51 1172StripLSD( ) ... -
Asterisk Extension中的Application命令详解六
2010-12-22 09:50 1587MailboxExists() ... -
Asterisk Extension中的Application命令详解五
2010-12-22 09:46 1947Math( ) Performs mathema ... -
Asterisk Extension中的Application命令详解四
2010-12-22 09:44 2108Hangup( ) Unconditionall ... -
Asterisk Extension中的Application命令详解三
2010-12-22 09:43 3263CheckGroup( )检查特定组中的信道数。CheckG ... -
Asterisk Extension中的Application命令详解二
2010-12-22 09:41 2103AgentLogin( )允许呼叫代理 ... -
Asterisk Extension中的Application命令详解一
2010-12-22 09:39 1671AbsoluteTimeout() 设置呼叫最大呼叫时长 A ... -
拨号方案基础
2010-12-22 09:28 1266ApplicationAnswer(),Playback()和 ... -
模拟卡的疑难杂症
2010-12-22 09:23 4936, 不能编译zaptel和asterisk ... -
Asterisk的拨号计划命令
2010-12-22 09:21 1795Asterisk的拨号计划命 ... -
Asterisk 配置文详解和Freepbx功能键逐个数
2010-12-22 09:19 12899Asterisk 配置文详解 转自:http://www.ha ... -
Asterisk manager API(AMI)文档(中文版
2010-12-22 03:04 1505Asterisk控制接口(AMI)允许管理客户端程序连接到一个 ... -
SIP 中文翻译
2010-12-22 03:01 29861.介绍 extensions.conf中使用sip设备的语 ... -
队列振铃方式
2010-12-22 02:58 981队列振铃方式有:ringall,roundrobin,leas ... -
asterisk App Dial 拨号 发起外呼
2010-12-22 02:54 2887Synopsis Attempts to establi ... -
asterisk 集群配置的完全解决方案
2010-12-22 02:53 1196IAX 设置详细两台Asterisk服务器设置步骤如下:* 设 ...
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