Asterisk 通道变量

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). If is set