asterisk 通道变量

${ACCOUNTCODE}: 用户计费帐号 sip.conf 里的 account=XXXX

${ANSWEREDTIME}: 通话时长(秒)

${BLINDTRANSFER}: 通道是否为转接类型

${CALLERID(all)}: 主叫用户名(主叫ID) 格式 name(123454)

${CALLERID(name)}: 主叫用户名 sip.conf 里的 username=XXXX

${CALLERID(num)}: 主叫号码sip.conf 里的 callerid=XXXX

${CALLINGPRES}: PRI Call ID Presentation variable for incoming calls (See callingpres )

${CHANNEL}: 当前通道标识

${CONTEXT}: 当前context

${DATETIME}: 当前日期时间

${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}: 当前通道状态 ${DNID}: 用户所拨打的号码

${EPOCH}: The current UNIX-style epoch (number of seconds since 1 Jan 1970)

${EXTEN}: 当前所拨打分机号码

${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}: 提示语言

${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}: 通道类型。是否可以转接

${TXTCIDNAME}: Result of application TXTCIDName (see below)  

${UNIQUEID}: 当前唯一标识  

${TOUCH_MONITOR}: used for "one touch record" (see features.conf, and wW dial flags).