转自：http://hi.baidu.com/dashboard/blog/item/b858461eabe203f01ad576f3.html
- ${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
- ${TOUCH_MONITOR}: used for "one touch record" (see features.conf, and wW dial flags). If is set
1.4版本已经没有${TIMESTAMP}这个变量了～今天就是因为它耽误我N多时间~日～
