asterisk-通道变量列表
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${ACCOUNTCODE}: 用戶計費帳號 sip.conf 里的 account=XXXX?
${ANSWEREDTIME}: 通話時長(秒)?
${BLINDTRANSFER}: 通道是否為轉接類型
${CALLERID(all)}: 主叫用戶名(主叫ID) 格式 name(123454)
${CALLERID(name)}:?主叫用戶名?sip.conf 里的 username=XXXX?
${CALLERID(num)}: 主叫號碼sip.conf 里的 callerid=XXXX?
${CALLINGPRES}: PRI Call ID Presentation variable for incoming calls (See?callingpres?)?
${CHANNEL}: 當前通道標識?
${CONTEXT}: 當前context
${DATETIME}: 當前日期時間?
${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}: 當前通道狀態?
${DNID}: 用戶所撥打的號碼?
${EPOCH}: The current UNIX-style epoch (number of seconds since 1 Jan 1970)?
${EXTEN}: 當前所撥打分機號碼?
${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}: 提示語言?
${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}: Used to set the SIP codec for a call (apparently broken in Ver 1.0.1, ok in Ver. 1.0.3 & 1.0.4, not sure about 1.0.2)?
${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}: 通道類型。是否可以轉接?
${TXTCIDNAME}: Result of application TXTCIDName (see below)
?${UNIQUEID}: 當前唯一標識
?${TOUCH_MONITOR}: used for "one touch record" (see features.conf, and wW dial flags).
轉載于:https://my.oschina.net/u/1537782/blog/222847
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