asterisk-通道變量列表

${ACCOUNTCODE}: 用戶計費賬號 sip.conf 裏的 account=XXXX 
${ANSWEREDTIME}: 通話時長(秒) 
${BLINDTRANSFER}: 通道是否爲轉接類型
${CALLERID(all)}: 主叫用戶名(主叫ID) 格式 name(123454)
${CALLERID(name)}: 主叫用戶名 sip.conf 裏的 username=XXXX 
${CALLERID(num)}: 主叫號碼sip.conf 裏的 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。是否能夠轉接 
${TXTCIDNAME}: Result of application TXTCIDName (see below)
 ${UNIQUEID}: 當前惟一標識
 ${TOUCH_MONITOR}: used for "one touch record" (see features.conf, and wW dial flags).