Asterisk使用命令

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Asterisk CLI 命令列表

以下的命令是在CLI控制台上用help命令列出的集合：!: 执行一个shell command用法：!<command>说明：执行一个shell命令abort halt: 取消一个暂停用法：abort shutdown说明：让Asterisk放弃一个正在执行的shutdown或者restart，正常继续运行呼叫操作add extension: 增加新的extension到context用法：add extension <exten>,<priority>,<app>,<app-data> into <context> [replace]说明：这个命令会增加一个新的extension到<context>。

如果已经有了同样的priority的extension‘，最后的replace’参数可以替代这个extension例子：add extension 6123,1,Dial,IAX/216.207.245.56/6123 into local现在，你可以拨打6123，并且和Markster交谈了。

add ignorepat：增加新的匹配模式用法：add ignorepat <pattern> into <context>说明：这个命令增加一个新的匹配模式到context <context>例子：add ignorepat _3XX into localadd queue member: 向指定的队列增加一个通道用法：add queue member <channel> to <queue> [penalty <penalty>]说明：向一个队列增加一个通道成员，[penalty <penalty>]指明这个通道成员的优先级ael debug contexts:启用AEL contexts调试ael debug macros:启用AEL macros调试ael debug read：启用AEL read调试ael debug tokens:启用AEL tokens调试ael no debug：关闭AEL调试信息ael reload：重载AEL配置agent logoff：让一个座席离线用法：agent logoff <channel> [soft]说明：让一个座席不在登录进来，如果指定了soft选项，不会强行挂机退出通话agi debug：启用AGI调试用法：agi debug说明：启用为调试目的而将AGI传输的内容输出agi no debug：关闭AGI调试用法：agi no debug说明：关闭为调试目的而将AGI传输的内容输出到控制台cb mysql status：显示CBMySQL的连接状态信息用法：cb mysql status说明：列出当前CBMySQL的mysql连接状态结果例子：Connected to conference@localhost, port 3306 with username mvb2000user for 6 days, 10 hours, 11 minutes, 40 seconds.Wrote 0 records since last restart.cdr mysql status：显示cdr_mysql的连接状态信息用法：cdr mysql status说明：显示cdr_mysql的当前连接状态例子：Connected to mvb2000cdrdb@localhost, port 3306 using table cdr for 6 days, 10 hours, 14 minutes, 46 seconds.Wrote 7 records since last restart.cdr status：Display the CDR status用法：cdr status说明：显示通话详细记录引擎的系统状态例子：CDR logging: enabledCDR mode: simpleCDR registered backend: csvCDR registered backend: mysqlCDR registered backend: cdr_managerdatabase del：删除数据库的key/value用法：database del <family> <key>说明：删除Asterisk数据库中指定的family和key的记录database deltree: Removes database keytree/values用法：database deltree <family> [keytree]说明：在Asterisk数据库中删除一个family或者指定的在family中的keytreedatabase get: Gets database value用法：database get <family> <key>说明：在Asterisk数据库中为指定的family和key检索记录database put: Adds/updates database value用法：database put <family> <key> <value>说明：在asterisk数据库中以指定的famliy、key、value新增或者修改一个记录database show显示数据库内容用法：database show [family [keytree]]说明：显示Asterisk数据库的内容，可选的关联一个指定的family或者family和keytreedatabase showkey显示数据库内容用法：database showkey <keytree>说明：显示Asterisk数据库内容，关联到指定的keydebug channel：启用在一个通道上的调试信息用法：debug channel <channel>说明：在指定的通道上打开调试信息debug level:设置全局调试信息的等级用法：debug level <level> [filename]说明：设置调试为指定的等级（0为关闭调试）。

《Asterisk 使用资料》

Asterisk目录及配置文件/etc/asterisk/Asterisk主目录,包含其它关于Asterisk的配置文件;*zaptel.conf这个配置文件放在/etc,因为其它软件也可以使用Zaptel这个硬件及其驱动,所以不是放在/etc/asterisk里./usr/lib/asterisk/modules/这个目录包含所有可以加载Asterisk模块(应用程序\编辑器\格式和有用通道),在Asterisk启运时会加载这些模块(可以编辑modules.conf)./var/lib/asterisk/比较重要的是astdb文件与agi-bin目录;astdb包含Asterisk当地数据库信息.sounds/所有声音提示的文件在里面,包括Asterisk原代码中的sounds.txt文件mohmp3/如果配置了音乐保持,应用程序会在这个目录下查找mp3(用CBR从文件中去除身份标签).keys/使用公钥和私钥系统认证与一个由RSA数字签名形成的一对等连接.公钥和私钥的扩展名分别为.pub和.keyfirmware/这个目录含了很多Asterisk相兼容的设备固件,它只有iax/这个子目录,其中有Digium的IAXy的二进制固件镜像.images/只有在较多的支持并且利用图解式的图像设备被发布,这个目录将会与相应的目录有更大的关联./var/lib/asterisk/agi-bin agi-bin包含所有脚本,可以通过许多已经建立的AGI应用程序与Asterisk连接./var/spool/asteriskoutgoing/gcall/tmp/voicemail/================配置文件================/etc/asterisk/asterisk.conf主要配置文件，/etc/zaptel.conf硬件接口的基本层.修改这个配置文件要用modprobe装载Linux Kernel使用模块./etc/asterisk/zapata.conf为硬件配置Asterisk的接口./etc/asterisk/extensions.conf拨号方案./etc/asterisk/sip.conf SIP协议配置文件/etc/asterisk/iax.conf呼入和呼出IAX通道/etc/asterisk/extensions.conf拨号方案配置文件/etc/asterisk/moduprobe.conf加载模块配置文件Asterisk developer\'s documentation 翻译计划by serva今天，Asterisk已经成为一个VOIP业界使用最广泛的一个集成电信级别P BX的工具，IPPBX如今已经成为VOIP商家争夺的一块重要市场，在我们越来越熟悉使用Asterisk的同时，我们希望能够越来越深入的了解asterisk，特别是对于开发人员，如果有自己特定的需求，在复杂的asterisk文件夹和asteri sk的源代码文件中迷失了方向，不知所措。

《Asterisk 使用资料》..

Asterisk 入门一 --- Asterisk 1.6.2.6 安装

Asterisk 入门一 --- Asterisk 1.6.2.6 安装2010-04-28 16:27其实asterisk安装灰常简单只要跟着安装包中的readme 文档一步一步走就可以了：1.安装测试环境虚拟机：Sun VirtualBox 3.1X操作系统：CentOS 52.Astrisk各稳定版本以及附加工具/downloads改页面有很多下载选项大概的介绍一下1.AsteriskAsterisk 源码压缩包,这就是我们要用的2.AsteriskNOWAsteriskNOW 是一个光盘镜像包含了linux 系统，一旦安装完成就有一套简单的PBX 系统了3.Asterisk Add-OnsAsterisk Add-Ons 是Asterisk 和Asterisk 一些相关的驱动和组件等。

3.下载安装# wget/pub/telephony/asterisk/releases/asteris k-1.6.2.6.tar.gz#tar -zxvf asterisk-1.6.2.6.tar.gz# cd asterisk-1.6.2.6运行配置脚本检测依赖关系并设置安装目录# ./configure --prefix=/usr/local/asterisk看到硕大的星号了嘛哈哈配置成功.$$$$$$$$$$$$$$$=...$7$7.. .7$$7:..$$:. ,$7.7.$7. 7$$$$ .$$77..$$. $$$$$ .$$$7..7$ .?. $$$$$ .?. 7$$$.$.$. .$$$7. $$$$7 .7$$$. .$$$..777. .$$$$$$77$$$77$$$$$7. $$$,$$$~ .7$$$$$$$$$$$$$7. .$$$..$$7 .7$$$$$$$7: ?$$$.$$$ ?7$$$$$$$$$$I .$$$7$$$ .7$$$$$$$$$$$$$$$$ :$$$.$$$ $$$$$$7$$$$$$$$$$$$ .$$$.$$$ $$$ 7$$$7 .$$$ .$$$.$$$$ $$$$7 .$$$.7$$$7 7$$$$ 7$$$$$$$$ $$$$$$$7. $$ (TM)$$$$$$$. .7$$$$$$ $$$$$$$$$$$$$$7$$$$$$$$$.$$$$$$ $$$$$$$$$$$$$$$$.该步骤可选，该步骤用于选择一些模块#make menuselect编译安装#make#make install安装一些简单的配置文件#make samples好了现在都安装结束了--------------------------------------------------------------------------------------------启动守护进程# safe_asterisk进入CLI#asteriks -r更多的命令可以使用 #man asterisk查看按q退出Asterisk 入门二 --- Asterisk 配置以及控制台命令2010-04-28 16:52接下来我们开始学习asterisk先提供个学习的教材和网址大部分的问题可以在/中找到解答其次提供一本 Asterisk 入门教程叫Asterisk，电话未来之路第二书签版.pdf自个摆渡下吧-------------------------------------------------------------------------------------Asterisk 几个相关目录配置文件/usr/local/asterisk/etc/asterisk可执行文件目录 /usr/local/asterisk/sbin先启动asterisk守护进程#safe_asterisk进入控制台#asterisk -rAsterisk 1.6.2.6, Copyright (C) 1999 - 2010 Digium, Inc. and others.Created by Mark Spencer <markster@>Asterisk comes with ABSOLUTELY NO WARRANTY; type 'core show warranty' for details. This is free software, with components licensed under the GNU General PublicLicense version 2 and other licenses; you are welcome to redistribute it under certain conditions. Type 'core show license' for details.=========================================================================Connected to Asterisk 1.6.2.6 currently running on localhost (pid = 18023)Verbosity is at least 3localhost*CLI>看看有啥命令>core show help或者具体的命令帮助>core show help <command>Asterisk app命令中文翻译2008-11-06 15:15常规指令Authenticate:鉴别用户VMAuthenticate:根据“voicemail.conf”鉴别用户Curl:接受外接URLs的修复。

Asterisk系统的安装与配置

Asterisk系统的安装与配置一、安装包装备asterisk1.8.7.1.tar.gzdahdi-linux-complete-2.6.1+2.6.1.tar.gzdahdi是Asterisk管理硬件的插件(中间件)二、安装dahdi 安装时需要下载东西，所以需要互联网1.tar zxvf dahdi-linux-complete-2.6.1+2.6.1.tar.gz2.cd dahdi-linux-complete-2.6.1+2.6.13.make all4.make install5.make config ；生成启动脚本/etc/init.d/dahdi6./etc/init.d/dahdi restart ；启动dahdi7.chkconfig dahdi on ；检查编译DAHDI时CONFLICTING TYPES FOR ‘BOOL’解决办法，google上面有，说是linux版本宏引起的。

三、安装asterisk1.tar zxvf asterisk1.8.7.1.tar.gz2.cd asterisk1.8.7.13../configure --build=i3864.；指定安装路径 ./configure --prefix=/usr/local5.make menuselect ;在配置界面选中meete应用6.make ;编译7.make install 安装Asterisk8.make samples ；安装配置文件模版/etc/asterisk下9.make progdocs ;安装Asterisk程序文档10.asterisk 启动四、配置SIP通道默认语言 /etc/asterisk/sip.conf[general]language=cn/enbindaddr=0.0.0.0 服务器IP五、配置基本SIP账户 /etc/asterisk/sip.conf1.CLI命令sip show userssip show peerssip show settingssip set debug on/offsip reload2.分机号码模板[SIPPHONE](!)type=friend 呼入呼出均可host=dynamic 分机号注册时获取contex=geeyavoip ; 分机号呼入时使用的上下文extensions.conf中的上下文nat=no 不支持natqualify=20003.使用号码模板定义分机号[301](SIPPHONE) ;继承SIPPHONE模板username=301accountcode=301directrtpsetup=yes aster工作在proxy，不修改sdp实现rtp透传;secret=301 ;无需密码则注释掉这句;directmidia=yes 媒体穿越nat NAT=yes;canreinvite = yes 与上面一行等效的to disable re-invites if you had NAT=yes六、配置电话会议室 /etc/asterisk/meete.conf[rooms] ；会议室号将作为拨号计划中 meetme参数conf => 1001 ;定义会议室号1001，无密码conf => 1002，918 ;定义会议室密码为918备注：拨号计划中调用meetme函数时，将进入会议桥，如需要密码，系统将提示输入会议室密码。

Asterisk安装和配置

1.安裝所需套件yum install gcc gcc-c++ kernel-devel zlib zlib-devel openssl openssl-devel2.下载asterisk、libpri、zaptelcd /usr/local/src/wgetwgetwget3.安装zapteltar -zxvf zaptel-.1.tar.gzcd /usr/local/src/zaptel-.1./configuremakemake installmodprobe zaptel4.安装libpritar -zxvf libpri-.tar.gzcd /usr/local/src/libpri-makemake install5.安装asterisktar -zxvf asterisk-.tar.gzcd /usr/local/src/asterisk-./configuremakemake installmake samplesasterisk -vvvc6.分机配置vim /etc/asterisk/sip.confSIP分机常用参数配置：[101] ;SIP分机注册账号callerid=ABC ;分机显示名称(可不设定)username=101 ;SIP再注册时要使用的账号(可不设定)type=friend ;连线的模式，一般话机设成friendsecret=101 ;分机注册密码qualify=yes ;验证模式，只有在type=peer时生效(可不设定)nat=yes ;是否在NAT下host=dynamic ;搜寻Client的模式，dynamic由话机主动去注册或者输入Hostname、IP由SIP Server去连线dtmfmode=rfc2833 ;按键信号模式，预设为rfc2833context=internal ;设定要用extensions.conf哪一组的动作canreinvite=no ;分机直通或由SIP连接callgroup=0 ;定义群组，可利用于群组广播....等(可不设定)pickupgroup=0 ;定义代接群组(可不设定)[102]username=102type=friendsecret=102qualify=yesnat=yeshost=dynamicdtmfmode=rfc2833context=internalcanreinvite=no7.设定接通动作vim /etc/asterisk/extensions.conf最后面加放设定[internal]exten => _X.,1,Dial(SIP/${EXTEN}|30) exten => _X.,n,Hangup()说明：_：代表开头X：代表0-9.：代表任意长度的字元_X.：电话以数字开始不限制长度Dial：接通SIP：线路模式SIP协议${EXTEN}：收到的号码30：只振玲等待30秒Hangup：挂断8.设定远端管理账号(依需求设定)vim /etc/asterisk/manager.confenabled = yes最后面增加[admin] secret=admin deny=.0/0.0.0.0 permit=127.0.0.1/255.255.255.0 read = system,call,log,verbose,command,agent,user write =system,call,log,verbose,command,agent,user用telnet localhost 5038到asterisk的管理界面出现Asterisk Call Manager/1.0输入action: login <enter>username: admin <enter>secret: admin <enter><e。

asterisk常用命令

asterisk常用命令Asterisk基本配置Asterisk译为星号(*)在很多应用中被用做通配符,Astrisk做为PBX系统的完美名称,原因之一是Asterisk可以连接数目庞大的接口类型,包括:1模拟接接口,如你的电话线或模拟电话.2数字线路,如T-1和E-1线路3V oip协议,如SIP和IAX目前正式版本是IAX2,但是IAX1的所有格式都已经停掉.所以当说到IAX都是特拽IAX2使用接口配置文件1.Zaptel.conf:硬件接口基层配置,我们将建立一个FX0和一个FXS通道.2.zapata.conf:硬件配置Asterisk的接口.3.extension.conf:拨号方案文件.4.sip.conf:配置SIP协议的文件5.iax.conf:配置呼入和呼出IAX通道的文件******编辑完zaptel.conf文件,需要使用/sbin/ztcfg -vv装载配置到硬件(不须要详细输出可以省略-vv);修改信令方法需要重启.在编辑sip.conf和iax.conf文档后,分别需要装载chan_iax2.so和chan_sip.soAsterisk译为星号(*)在很多应用中被用做通配符,Astrisk做为PBX系统的完美名称,原因之一是Asterisk可以连接数目庞大的接口类型,包括:1模拟接接口,如你的电话线或模拟电话.2数字线路,如T-1和E-1线路3V oip协议,如SIP和IAX目前正式版本是IAX2,但是IAX1的所有格式都已经停掉.所以当说到IAX都是特拽IAX2使用接口配置文件1.Zaptel.conf:硬件接口基层配置,我们将建立一个FX0和一个FXS通道.2.zapata.conf:硬件配置Asterisk的接口.3.extension.conf:拨号方案文件.4.sip.conf:配置SIP协议的文件5.iax.conf:配置呼入和呼出IAX通道的文件******编辑完zaptel.conf文件,需要使用/sbin/ztcfg -vv装载配置到硬件(不须要详细输出可以省略-vv);修改信令方法需要重启.在编辑sip.conf和iax.conf文档后,分别需要装载chan_iax2.so和chan_sip.so******SIPSIP(会话初始协议),通常用于VOIP电话,进行呼叫建立,呼叫协商,呼叫结束.它帮助两个端互相通认,但它不处理媒体;当呼叫建立后,他通过实时传输协议(RTP)在电话A到电话B直接传输媒体SIP和RTPSIP是一个应用层的信令协议.他使用的端口是5060(通常)进行通信.SIP可以通过UDP 和TCP传输层协议进行传输.Asterisk目前没有TCP用于传输SIP信息.RTP用于端点间传输媒体(语音),Asterisk中RTP使用大数字的无特权端口(默认10,000到20,000)SIP优点:普通的被接受和结构灵活.其它的VOIP协议还有H.323,IAX,MGCP.SIP配置/etc/asterisk/sip.conf文件中:[general]srvlookup=yet[10000]type=friend;可以定义用户\端\朋友asterisk常用命令1. 运行调试常用命令./asterisk & 在后台启动asterisk./asterisk –vvvc 启动asterisk并尽量多的在控制台显示调试信息./asterisk –r 连接asterisk控制台set verbose 9999 设置打印详细信息2.asterisk控制台下常用命令sip reload 重新加载sip配置信息sip set debug 设置显示更多的sip信息sip set debug off关闭显示更多的sip信息sip show channels 显示所有活动的SIP通道sip show peers 显示所有已定义的SIP peerdialplan show 显示拨号方案dialplan reload重新加载拨打方案stop gracefully 优雅地停止asteriskstop now 立即停止运行asteriskmodule show 显示所有已加载模块module load cdr_radius.so 加载模块cdr_radius.somodule reload cdr_radius.so 重新加载模块cdr_radius.somodule unload cdr_radius.so 卸载模块cdr_radius.socore show applications 显示所有的应用，比如Dial, Radius, Hangup, Answer等core show channels 显示所有的通话频道core show codecs 显示所有支持的编解码器core show functions 显示所有的功能函数asterisk -r之后敲restart now,好像。

asterisk配置文件列表及常用指令

配置文件说明列表：常规指令Authenticate:鉴别用户VMAuthenticate:根据“voicemail.conf”鉴别用户Curl:接受外接URLs的修复。

支持POSTingDUNDiLookup:用DUNDi查寻号码SendDTMF:发送独裁的DTMF数据SendImage:发送图像档案SendText:发送给客户正文消息SendURL:发送给客户USL以显示System:执行系统命令Transfer:将访客转移到另一个扩展名TrySystem:0回复执行系统命令Wait:稍等WaitExten稍等:WaitForRing:等待呼叫申请WaitMusicOnHold:等待，在线音乐BillingForkCDR:将CDR分成两个独立的实体NoCDR:确保 asterisk没有把CRD保存为特定的呼叫ResetCDR:重新安装CDR数据SetAccount:设置帐簿号码SetAMAflags:开通AMA标志开户的渠道SetCDRUserField:建立CDR用户领域AppendCDRUserField:给CDR用户附加数据Call management (hangup, answer, dial, etc)Answer:如果呼叫回复一个信道Busy:显示忙的情形等待挂断ChanIsAvail:如果信道可以利用即核查Congestion:显示拥塞等待挂断Dial:设置一个指令连接当前的信道DISA:直接访问内在系统Hangup: 无条件挂断Ringing显示呼叫Caller presentation (ID, Name etc)CallingPres:更改呼叫者的身份介绍LookupBlacklist: 从黑名单数据库上查询访客的身份姓名和号码LookupCIDName: 从当地的数据库查询访客的身份和姓名PrivacyManager:如果没有CallerID过来，要求输入电话号码SetCallerID: 建立CallerID，发生矛盾服从CallerIDSetCallerPres:建立访客资料的独立信道SetCIDName:建立呼叫者身份姓名，发生矛盾服从CallerID SetCIDNum: 只建立访客的身份号码（没有姓名），发生矛盾服从CallerID SoftHangup: 请求挂断转到另一个信道Zapateller:拥SIT阻止远程商人ADSI• ADSIProg:将 Asterisk ADSI Scripts 加载到电话• GetCPEID: 获得 ADSI CPE IDDatabase handling• DBdel: 从数据库中删除一个键. 发生矛盾服从 DB.• DBdeltree: 从数据库中删除群. 发生矛盾服从 DB.• DBget:从数据库中找回数值. 发生矛盾服从DB.• DBput: 在数据库中储存一个数值. 发生矛盾服从DB.• MYSQL:执行不同的 mySQL 数据库行为查 Asterisk database 找更多的信息.Application integration• AGI: 执行一套AGI 适应程序• DeadAGI: 在中断的信道上执行AGI• EAGI: 执行一套 AGI 适应程序• EnumLookup:在 ENUM 中查找号码• Macro: 执行较大的程序• NoOp: 不进行操作，可将数值打印到控制太进行调试• Perl: res perl 是Apache 的 mod_perl, 只适用于 Asterisk • PHP: res_php 不需要就可将PHP与进行一体化处理• Read: 用DTMF读变量• TXTCIDName: 从 TXT 记录中查找访客的姓名• UserEvent:发送任意结果给界面处理器Control flow & timeouts• AbsoluteTimeout: 建立呼叫最长时间• DigitTimeout: 建立数据间的最长时间间隔• Goto:转到优先码，扩展名和过渡段• GotoIf: 条件展转接• GotoIfTime: 当前的条件转接• Random: 在拨号栏中随机跳动• ResponseTimeout: 设置等候回应的最长时间• While: 开始一个当循环- *1.2beta• EndWhile: 结束一个当循环 - *1.2beta• ExecIf: 条件执行 - *1.2betaString & variable manipulation• Cut: 串处理功能，发生矛盾支持 CUT.• ImportVar: 赋数值变量• Math: 进行简单的计算• Prefix: 预设主要的数据• SetGlobalVar:赋数值变量• Set: 设信道变量和功能值• SetVar: 赋数值变量(如果不允许，使用 Set)• StripLSD: 消除附加的数据• StripMSD: 消除主要的数据• SubString: 在一个预设的变量中保存子链数据• Suffix:添加附加数据Sounds:录音重放• Background: 当执行其它命令的时候播放声音文件• BackgroundDetect: 用谈话探测背景文件• ControlPlayback: 用快进，重绕和离开当前命令控制器播放声音文件• DateTime: 显示日期/和时间• Echo: 重复声音回复到永福端• Festival: 使用欢快的声音合成器朗诵课文• Milliwatt: 在0分贝产生持续的1000赫兹的音调• MP3Player: 播放MP3声音文件• MusicOnHold: 无限度地播放音乐• Playback: 播放一个声音文件• Playtones: 当执行其它命令的时候播放声音列表• SayUnixTime: 显示日期/和时间• SayAlpha: 显示字母• SayDigits: 显示阿拉伯数字• SayNumber: 显示数字• SayPhonetic:显示语音• SetMusicOnHold: 在控制系统上显示默认的音乐• SetLanguage: 为录音重放显示语言转换• StopPlaytones: 停止播放音调列表查看Asterisk sound files 获取更多信息语音: 录音和监测• ALSAMonitor: 监测ALSA控制台• ChangeMonitor: 转换监测的路线文件名• ChanSpy:通用的路线干涉• Dictate: 听写的录音和播放• Monitor: 录音电话会话到声音文件• MuxMon:记录和混合电话支架• Record: 记录电话会话到语音文件• StopMonitor: 停止监测路线SIP 命令• SIPdtmfMode: 在SIP呼叫过程中转换DTMF模式• SIPGetHeader: 从SIP邀请信息中获取标题• SIPAddHeader: 填标题到外部的SIP邀请• suplimental resource SIPCallPickup:在获取组中获取打入的电话• sipredirect: 发送SIP 302信息到呼叫者ZAP 命令• Flash: 编辑到Zap主页• ZapBarge: 干涉Zap路线• ZapRAS: 提供ISDN数据服务• ZapScan: 扫描Zap路线到监督程序调用•查看 Asterisk zap channels, zapata.conf 获取等多信息.语音邮件和会议• Directory: 规定语音邮件扩展目录• HasNewVoicemail: 有条件的分枝到优先权＋101• MailboxExists: 检查邮箱是否存在• MeetMe:简单的 MeetMe 会议桥• MeetMeAdmin: MeetMe 会议管理• MeetMeCount: MeetMe参与者计算• VoiceMail: 语音留言• VoiceMailMain: 进入语音系统• VMAuthenticate: 基于语音邮箱鉴别用户查看 voicemail.conf 获取更多的信息列队和ACD操纵• AddQueueMember: 自动添加排队成员• AgentCallbackLogin: 呼叫代理商回叫注册• AgentLogin: 呼叫代理商注册• AgentMonitorOutgoing: 记录代理商的外出呼叫• ParkAndAnnounce: 停放和宣告• ParkedCall: 回复被停放的呼叫• PauseQueueMemeber: 中止代理• Queue: 排队等代呼叫• RemoveQueueMember: 自动移交列队成员• UnpauseQueueMemeber: 重新开始代理警报监测/中心站警报接受站：仿效一个Ademco 连接ID的警报接受站业余无线电接受装置/转发器连接Rpt：支持业余无线电接受装置和商业化的两种转发器连接外部应用（不是在CVS里）Asterisk app_dbodc：用unixODBC进行拨号方案修正DynExtenDB：数据库中的存储扩充Iconv：字符切换设置app Prepaid：为Postgres 所设计的PPPD：邮件后台程序连接器Bristuff & zaphfc applications• PickUp: 大多数路线是独立的；是bristuff片的一部分• PickUpChan:接收指定路线• PickupSIPuri• PickDown: 挂断长途电话• Steal: 接受中转电话• Devstate: 产生一种安装设备驱动程序状态来改变事件（忙，打电话…）• Segfault:用 segfault 撞击Asterisk• ZapEC: 对于Zap能或者不能的回波消除• Autoanswer: 在指定范围内自动回话• AutoanswerLogin: 登陆到自动回话应用程序vISDN applicationsVISDN交叠拨号Sirrix 路径的应用• SrxEchoCan: 不能/能回波消除• SrxDeflect: 偏转进入的电话• SrxMWI:在Sirrix组上设置/重新设置信息等待指示（MWI）。

Asterisk的连接与管理

Asterisk的管理方法很晚了，刚才稍微看了一下Asterisk的管理方法。

现在想做一个记录。

Asterisk有三种管理方法，最常用的当然是通过asterisk命令进入CLI模式对Asterisk进行管理。

另外还有两种方法：1）用TCP协议，连接到5038端口，然后用AMI命令来管理Asterisk;2）用HTTP协议，连接到8080端口，通过WEB的形式对Asterisk进行管理；下面是基本的URL：http://xxx.xxx.xxx.xxx:8088/static/config/index.html1.Asterisk命令行对应的套接字文件是：/var/run/asterisk/asterisk.ctl管理员可以配置cli_permissions.conf对操作系统用户登录asterisk进行授权。

具体的授权方法就不在这里介绍，有兴趣的人可以看看cli_premissions.conf中的说明。

rasterisk的作用基本上和asterisk –r是一致的，不作特别介绍，只对Asterisk命令进行分2.使用AMI方法AMI方法是通过socket登录到Asterisk中，然后用AMI命令对Asterisk进行管理。

通过这个管理接口，我们可以用某种编程语言开发出自己的Asterisk管理平台。

实际上，很多基于GUI的管理平台也是通过这种方法对Asterisk进行管理。

这种方法的配置文件是：manager.conf3.WEB管理Web管理的配置文件是：http.conf浏览器。

具体的登录路径类似于：http:// xxx.xxx.xxx.xxx:8088/static/config/index.html 使用的帐号密码是manager.conf中定义的帐号密码。

登录后的管理终端如下：这个终端的静态页面会保存在下面地方：。

Asterisk 安装和配置

准备工作：（1）安装WM ware虚拟机（2）操作系统：内核2.6的Linux 系统，建议使用Fedora 。

（3）安装软件：asterisk.tar.gz , libpri.tar.gz, zaptel.tar.gz。

路径：\\10.7.1.7\TeamShare\AlexJunkieGrp\TEMP。

（将这三个文件放到Linux 系统的/opt文件夹下。

）（4） SSH终端。

步骤（1）：安装虚拟机（略）步骤（2）：安装fedora 8 操作系统（也可以使用已经安装过的操作系统）安装操作系统 fedora 8（1）最好完全安装。

（2）在安装过程中配置自己的硬盘大小，如果完全安装的话一般4.0G是不够的。

所以请避免安装时引起的很多不必要的麻烦Attention: 保证网络流通，关闭防火墙。

安装Asterisk依赖包清单包－－依赖关系gcc--libpri, zaptel, asterisklibtermcap-devel--asteriskkernel-devel--zaptelkernel-smp-devel--zaptelgcc-c++--asteriskopenssl-devel--asterisknewt-devel--zaptelzlib-devel--asteriskunixODBC-devel--asterisklibtool--asteriskmake—asterisk（1）检查是否已经安装kernel-devel# rpm -q kernel-devel例如：kernel-devel-2.6.18-92.1.10.el5，表示已经安装，如果没有安装，要用yum安装一下，如下 yum install kernel-devel（2）检查是否已经安装以下的辅助软件包rpm -q bisonrpm -q bison-develrpm -q ncursesrpm -q ncurses-develrpm -q zlibrpm -q zlib-develrpm -q opensslrpm -q openssl-develrpm -q gnutls-develrpm -q gccrpm -q gcc-c++rpm -q mysql-devel（4）如果没有安装则用yum安装yum install bisonyum install bison-develyum install ncursesyum install ncurses-develyum install zlibyum install zlib-develyum install opensslyum install openssl-develyum install gnutls-develyum install gccyum install gcc-c++yum install mysql-devel准备工作完毕步骤（3）：安装asterisk （1）：解压缩软件包＃ cd /opt解压缩三个文件包：＃tar -xvzf libpri.tar.gz＃tar -xvzf zaptel.tar.gz＃tar -xvzf asterisk.tar.gz删除文件包：＃rm -r *.gz（2）：安装zaptel# cd zaptel# ./configure# make （显示编译成功）# make install （显示安装成功）# make configAttention: 安装过程中不会出现错误信息（3）：安装libpri退出zaptel文件夹＃cd ..# cd libpri# make# make install（4）：安装asterisk如果没有出现错误，安装是正确的# cd ..# cd asterisk# ./configure# make （显示编译成功）# make install （显示安装成功）# make samples测试一下asterisk是否安装成功可以执行asterisk的主命令# asterisk –vvvvvvvvc（启动asterisk）如果这个时候提示cli> 表示已经成功cli> stop now（退出asterisk）# make config（配置启动）安装asterisk完成步骤（4）：asterisk 配置在安装asterisk成功后，用两个sip电话通过asterisk服务互通打电话，需要配置sip.conf, extension.conf。

queue命令 asterisk

Queues.confQueues.conf-[general];; Global settings for call queues;全局设置; Persistent Members; Store each dynamic member in each queue in the astdb so that; when asterisk is restarted, each member will be automatically; read into their recorded queues. Default is 'yes'.;把每一个QUEUE里面的成员都存到astDB里面,这样当asterisk重启的时候,每一个成员都能够自动的恢复到Queue当中,default='yes'persistentmembers = yes;; AutoFill Behavior; The old/current behavior of the queue has a serial type behavior; in that the queue will make all waiting callers wait in the queue; even if there is more than one available member ready to take; calls until the head caller is connected with the member they; were trying to get to. The next waiting caller in line then; becomes the head caller, and they are then connected with the; next available member and all available members and waiting callers; waits while this happens. The new behavior, enabled by setting; autofill=yes makes sure that when the waiting callers are connecting; with available members in a parallel fashion until there are; no more available members or no more waiting callers. This is; probably more along the lines of how a queue should work and; in most cases, you will want to enable this behavior. If you; do not specify or comment out this option, it will default to no; to keep backward compatibility with the old behavior.;会记住每一个成员和用户当前的状态,自动去尝试连接坐席,并自动更新用户当前的状态.比如之前还有几个人在等待.下一个等待的用户将会变成第一个用户，然后他们会连接到第一个可用的分机。

asterisk命令

asterisk是开源的VOIP系统。

是目前最流行的VOIP开源软件。

在使用过程中可以使用下述的命令，对系统进行监控调试。

下文来自网络，非本人原创1. 运行调试常用命令./asterisk & 在后台启动asterisk./asterisk –vvvc 启动asterisk并尽量多的在控制台显示调试信息./asterisk –r 连接asterisk控制台set verbose 9999 设置打印详细信息2.asterisk控制台下常用命令sip reload 重新加载sip配置信息sip set debug on 设置显示更多的sip信息ssip set debug off关闭显示更多的sip信息sip show channels 显示所有活动的SIP通道sip show peers 显示所有已定义的SIP peerdialplan show 显示拨号方案dialplan reload重新加载拨打方案stop gracefully 优雅地停止asteriskstop now 立即停止运行asteriskmodule show 显示所有已加载模块module load cdr_radius.so 加载模块cdr_radius.somodule reload cdr_radius.so 重新加载模块cdr_radius.somodule unload cdr_radius.so 卸载模块cdr_radius.socore show applications 显示所有的应用，比如Dial, Radius, Hangup, Answer等core show channels 显示所有的通话频道core show codecs 显示所有支持的编解码器core show functions 显示所有的功能函数ps -aux ：显示进程命令。

asterisk -rvc 进入到asterisk。

输入: stop now 停止关闭asterisk。

queue命令 asterisk

Asterisk常用调试命令

Asterisk基本配置Asterisk译为星号(*在很多应用中被用做通配符,Astrisk做为PBX系统的完美名称,原因之一是Asterisk可以连接数目庞大的接口类型,包括:1模拟接接口,如你的电话线或模拟电话.2数字线路,如T-1和E-1线路3Voip协议,如SIP和IAX目前正式版本是IAX2,但是IAX1的所有格式都已经停掉.所以当说到IAX都是特拽IAX2使用接口配置文件1.Zaptel.conf:硬件接口基层配置,我们将建立一个FX0和一个FXS通道.2.zapata.conf:硬件配置Asterisk的接口.3.extension.conf:拨号方案文件.4.sip.conf:配置SIP协议的文件5.iax.conf:配置呼入和呼出IAX通道的文件******编辑完zaptel.conf文件,需要使用/sbin/ztcfg -vv装载配置到硬件(不须要详细输出可以省略-vv;修改信令方法需要重启.在编辑sip.conf和iax.conf文档后,分别需要装载chan_iax2.so和chan_sip.soAsterisk译为星号(*在很多应用中被用做通配符,Astrisk做为PBX系统的完美名称,原因之一是Asterisk可以连接数目庞大的接口类型,包括:1模拟接接口,如你的电话线或模拟电话.2数字线路,如T-1和E-1线路3Voip协议,如SIP和IAX目前正式版本是IAX2,但是IAX1的所有格式都已经停掉.所以当说到IAX都是特拽IAX2使用接口配置文件1.Zaptel.conf:硬件接口基层配置,我们将建立一个FX0和一个FXS通道.2.zapata.conf:硬件配置Asterisk的接口.3.extension.conf:拨号方案文件.4.sip.conf:配置SIP协议的文件5.iax.conf:配置呼入和呼出IAX通道的文件******编辑完zaptel.conf文件,需要使用/sbin/ztcfg -vv装载配置到硬件(不须要详细输出可以省略-vv;修改信令方法需要重启.在编辑sip.conf和iax.conf文档后,分别需要装载chan_iax2.so和chan_sip.so******SIPSIP(会话初始协议,通常用于VOIP电话,进行呼叫建立,呼叫协商,呼叫结束.它帮助两个端互相通认,但它不处理媒体;当呼叫建立后,他通过实时传输协议(RTP在电话A到电话B直接传输媒体SIP和RTPSIP是一个应用层的信令协议.他使用的端口是5060(通常进行通信.SIP可以通过UDP和TCP传输层协议进行传输.Asterisk目前没有TCP用于传输SIP信息.RTP用于端点间传输媒体(语音,Asterisk中RTP使用大数字的无特权端口(默认10,000到20,000SIP优点:普通的被接受和结构灵活.其它的VOIP协议还有H.323,IAX,MGCP. SIP配置/etc/asterisk/sip.conf文件中:[general]srvlookup=yet[10000]type=friend;可以定义用户\端\朋友asterisk常用命令1. 运行调试常用命令./asterisk & 在后台启动asterisk./asterisk –vvvc 启动asterisk并尽量多的在控制台显示调试信息./asterisk –r 连接asterisk控制台set verbose 9999 设置打印详细信息2.asterisk控制台下常用命令sip reloadsip set debug 设置显示更多的sip信息sip set debug off关闭显示更多的sip信息sip show channels 显示所有活动的SIP通道sip show peers 显示所有已定义的SIP peersip show channels 查看当前SIP协议的并发。

Asterisk 入门教程

Asterisk 入门教程subscriber：加入某个group，从而遵从某一个call restricionsubscriber group：可以使用1种call restriction。

call restriction：将numbering plan中各项取出，各自组成成不同的呼叫限制方案numbering plan:通用，是一个整体的计划，里面包括长途，本地，等各种形式的方案-------------------------------------------------------------以上，用于拨入号码的预处理，实际上是一种过滤，下面才是真正的呼叫处理。

--------------------------------------------------------------route:一种策略，根据号码匹配的模式，选择走哪个trunk。

trunk:实际上做通一个链路，供route选择。

analog的，一个POTS就是一条analog trunkanalog的数量由设备定，ip trunk没有限制数量，并且一个ip trunk连到对方IPPBX后，对方的subscriber 数量是不确定的，由那个IP PBX决定。

Asterisk，开始吧--------------------------------------------------------------什么是asterisk?开源电话平台Asterisk 通过了电话的开源平台。

基本上就是一个软件的PBX。

最初是Digium 公司的Mark Spencer 编写的，这个公司就是他创立的，专门生产并销售Asterisk使用的硬件。

Asterisk 简直就是一场电话的革命。

为什么使用Asterisk？现在电信行业的混乱，因为各种专利技术应用在各种层次的网络上，导致很大的混乱。

Asterisk 就是答案——给电信行业普遍的混乱环境带来秩序的革命。

asterisk sip配置说明..

asterisk sip配置说明1. 介绍extensions.conf中使用sip设备的语法是SIP/devicename，devicename名在下一节中说明。

如果用户在Internet上，可以使用SIP/username@domain形式,同时不要忘记打开D NS SRV功能。

如果定义了一个SIP代理，可以使用SIP/proxyhostname/user或者SIP/user@proxy hostname形式，proxyhostname的定义在下面的章节说明。

2. 在CLI中的查询命令：u sip show peers 显示所有的SIP peers(包括friends)u sip show users 显示所有的SIP users(包括friends)u sip show registry 显示注册到的主机状态u sip debug 显示所有的Sip信息u module reload chan_sip.so 重新装载sip的配置文件3. 设备命名命名一个设备之前，要先理解Asterisk是怎么处理呼入电话的：1) Asterisk取出SIP From: address中的username，使用它来匹配系统中定义的type=user的的设备名。

2) Asterisk检查INVITE请求中的IP地址和端口号，使用它来匹配系统中定义的t ype=peer的设备。

不要混淆extensions和设备名的概念。

设备需要一个唯一的名字。

设备名不能用于电话号码。

电话号码是拨号计划中的声明的一个extension。

注意：参数username并不是用户名的意思，在大多数情况下，它根本不是必须的。

在后边，我们把它重命名做defaultuser，因为他与defaultip合并使用。

4. 通用属性说明[general] ;通用属性context=default ;呼入电话默认使用的contextallowguest=no ; 允许或拒绝客户呼叫 (默认 yes)allowoverlap=no ; 允许重叠拨号(默认yes)allowtransfer=no ; 允许呼叫转接(默认yes)realm=mydomain.tld ; 认证的领域。

Asterisk使用AGI脚本执行控制台命令获取通道参数

Asterisk使⽤AGI脚本执⾏控制台命令获取通道参数脚本如下require_once("/var/lib/asterisk/agi-bin/phpagi-asmanager.php");require_once("/var/lib/asterisk/agi-bin/phpagi.php");//使⽤Command必须先使⽤phpagi-asmanager的connect函数连接⾄后台$host = '127.0.0.1';$username = 'admin';$secret = '29666985';define('MAX_TRY_TIMES',3); //定义重试次数define('TIMEOUT',4000); //定义超时时间，单位毫秒$agi = new AGI(); //调⽤phpagi脚本中的函数$ami = new AGI_AsteriskManager();//调⽤phpagi-asmanager脚本中的函数$conn = $ami->connect($host,$username,$secret);if( $conn == false ){ //连接失败echo "Connection failed.\n";die;}$channel = $agi->get_variable("CDR(channel)"); //获取通道值$channel = $channel["data"];$arrReturn = $ami->Command("core show channel $channel");//在脚本中执⾏asterisk后台命令获取通道参数$lines=explode("\n", str_replace("\r\n", "\n", $arrReturn["data"]));$ary = array();$ary['Channel'] = substr($lines[2],strpos($lines[2],":")+2); //客户通道$ary['Extension'] = substr($lines[8],strpos($lines[8],":")+2); //分机号$ary['ExtenChannel'] = substr($lines[41],strpos($lines[41],"=")+1); //坐席通道$agi->set_variable("WKK",$ary['Extension']); //通道变量-分机号$agi->set_variable("ExtenChannel",$ary['ExtenChannel']); //坐席通道$ami->disconnect();。

asterisk安装和asterisk_sip控制台信息

asterisk安装和asterisk_sip控制台信息1. 检查依赖的包是否安装：依赖的包有gcc--libpri,zaptelasteriskncurses-devellibtermcap-develkernel-develkernel-smp-develgcc-c++openssl-develnewt-develzlib-develunixODBC-devellibtoolmake利用 rpm －q 名称进行查询看是否安装，无时进行安装，当然用源安装快！2.asterisk 的安装和环境的搭建首先在FTP上（自己Google)得到压缩包，或者ssh 远程得到命令为 get 名称自己的主机名称：放置目录安装到/usr/src目录下解压缩、切换目录（/usr/src/asterisk-xxx）、执行（sudo ./configure)、sudo make、sudo make install 、sudo make samples、sudo make progdocs修改配置： 1.1 在/etc/asterisk/sip.conf 中添加用户：[1001]type=friendsecret=testhost=dynamic[1002]type=friendsecret=testhost=dynamic1.2 /etc/asterisk/extensions.conf 中添加：[test]exten => 1001,1,Dial(SIP/1001)exten => 1002,1,Dial(SIP/1002)===============================各项讲解=================================== sip环境的简易配置：（所有的配置文件都在/etc/asterisk目录下，其它地方的配置无效） sip.conf[general] 所有用户和对端的默认选项allowoverlap=nobindport=5060 监听端口bindaddr=0.0.0.0 监听所有过来的电话srvlookup=yes 使用 DNS 服务qualify=yescontext=test 拨号方案[1001]type=friend 类型用户类型的用于认证呼入呼叫端Peer 是用于呼出呼叫朋友类型两者都有secret=test 密码host=dynamic 动态IP[1002]type=friendsecret=testhost=dynamicextensions.conf在文档的最后添加：[test]exten => 1001,1,Dial(SIP/1001) 呼叫超时秒数exten => 1002,1,Dial(SIP/1002)====================================== ===================================运行asterisk的命令为：asterisk & or sudo /usr/sbin/asterisk -vvvvvvcg停止命令为：stop now连接到命令行：asterisk －rvvvdialplan reloadsip reload显示支持的通信类型： show channeltypes打开调试信息： sip set debug现在astersik下的sip环境就搭建完成了，开启astersik ================================== sudo /usr/sbin/asterisk -vvvvvvcg =============================== ==登录到Astersk服务器运行控制台: asterisk -crvvv查看登录用户 : sip shwo peers*CLI> sip show peersName/username Host Dyn Nat ACL Port Status1004 (Unspecified) D 0 UNKNOW N1003 (Unspecified) D 0 UNKNOW N1002 (Unspecified) D 0 UNKNOW N1001/1001 192.168.123.58 D 39478 OK (101 ms)4 sip peers [Monitored: 1 online, 3 offline Unmonitored: 0 online, 0 offline]*CLI>查看详细记录： sip show peer 1001*CLI> sip show peer 1002 ////注意与查看全部的区别 peers peer* Name : 1002Secret : <Set>MD5Secret : <Not set>Context : testSubscr.Cont. : <Not set>Language :AMA flags : UnknownTransfer mode: openCallingPres : Presentation Allowed, Not ScreenedCallgroup :Pickupgroup :Mailbox :VM Extension : asteriskLastMsgsSent : 32767/65535Call limit : 0Dynamic : YesCallerid : "" <>MaxCallBR : 384 kbpsExpire : -1Insecure : noNat : RFC3581ACL : NoT38 pt UDPTL : NoCanReinvite : YesPromiscRedir : NoUser=Phone : NoVideo Support: NoTrust RPID : NoSend RPID : NoSubscriptions: YesOverlap dial : NoDTMFmode : rfc2833LastMsg : 0ToHost :Addr->IP : (Unspecified) Port 0 Defaddr->IP : 0.0.0.0 Port 5060Def. Username:SIP Options : (none)Codecs : 0x8000e (gsm|ulaw|alaw|h263) Codec Order : (none)Auto-Framing: NoStatus : UNKNOWNUseragent :Reg. Contact :*CLI>重新加载拨号方案：dialplan reload*CLI> dialplan reload== Parsing '/etc/asterisk/extensions.conf': Found== Setting global variable 'CONSOLE' to 'Console/dsp'== Setting global variable 'IAXINFO' to 'guest'== Setting global variable 'TRUNK' to 'Zap/G2'== Setting global variable 'TRUNKMSD' to '1'-- Registered extension context 'dundi-e164-canonical'-- Registered extension context 'dundi-e164-customers'-- Registered extension context 'dundi-e164-via-pstn'-- Registered extension context 'dundi-e164-local'-- Including context 'dundi-e164-canonical' in context 'dundi-e164-local'-- Including context 'dundi-e164-customers' in context 'dundi-e164-local'-- Including context 'dundi-e164-via-pstn' in context 'dundi-e164-local'-- Registered extension context 'dundi-e164-switch'-- Including switch 'DUNDi/e164' in context 'dundi-e164-switch'-- Registered extension context 'dundi-e164-lookup'-- Including context 'dundi-e164-local' in context 'dundi-e164-lookup'-- Including context 'dundi-e164-switch' in context 'dundi-e164-lookup'-- Registered extension context 'macro-dundi-e164'-- Added extension 's' priority 1 to macro-dundi-e164-- Including context 'dundi-e164-lookup' in context 'macro-dundi-e164'-- Registered extension context 'iaxtel700'-- Added extension '_91700XXXXXXX' priority 1 to iaxtel700 -- Registered extension context 'iaxprovider'-- Registered extension context 'trunkint'-- Added extension '_9011.' priority 1 to trunkint-- Added extension '_9011.' priority 2 to trunkint-- Registered extension context 'trunkld'-- Added extension '_91NXXNXXXXXX' priority 1 to trunkld -- Added extension '_91NXXNXXXXXX' priority 2 to trunkld -- Registered extension context 'trunklocal'-- Added extension '_9NXXXXXX' priority 1 to trunklocal-- Registered extension context 'trunktollfree'-- Added extension '_91800NXXXXXX' priority 1 to trunktollfree-- Added extension '_91888NXXXXXX' priority 1 to trunktollfree-- Added extension '_91877NXXXXXX' priority 1 to trunktollfree-- Added extension '_91866NXXXXXX' priority 1 to trunktollfree-- Registered extension context 'international'-- Including context 'longdistance' in context 'international' -- Including context 'trunkint' in context 'international'-- Registered extension context 'longdistance'-- Including context 'local' in context 'longdistance'-- Including context 'trunkld' in context 'longdistance'-- Registered extension context 'local'-- Including context 'default' in context 'local'-- Including context 'trunklocal' in context 'local'-- Including context 'iaxtel700' in context 'local'-- Including context 'trunktollfree' in context 'local'-- Including context 'iaxprovider' in context 'local'-- Including context 'parkedcalls' in context 'local'-- Registered extension context 'macro-trunkdial'-- Added extension 's' priority 1 to macro-trunkdial-- Added extension 's' priority 2 to macro-trunkdial-- Added extension 's-NOANSWER' priority 1 to macro-trunkdial-- Added extension 's-BUSY' priority 1 to macro-trunkdial-- Added extension '_s-.' priority 1 to macro-trunkdial-- Registered extension context 'macro-stdexten'-- Added extension 's' priority 1 to macro-stdexten-- Added extension 's' priority 2 to macro-stdexten-- Added extension 's-NOANSWER' priority 1 to macro-stdexten-- Added extension 's-NOANSWER' priority 2 to macro-stdexten-- Added extension 's-BUSY' priority 1 to macro-stdexten-- Added extension 's-BUSY' priority 2 to macro-stdexten-- Added extension '_s-.' priority 1 to macro-stdexten-- Added extension 'a' priority 1 to macro-stdexten-- Registered extension context 'macro-stdPrivacyexten'-- Added extension 's' priority 1 to macro-stdPrivacyexten -- Added extension 's' priority 2 to macro-stdPrivacyexten -- Added extension 's-NOANSWER' priority 1 to macro-stdPrivacyexten-- Added extension 's-NOANSWER' priority 2 to macro-stdPrivacyexten-- Added extension 's-BUSY' priority 1 to macro-stdPrivacyexten-- Added extension 's-BUSY' priority 2 to macro-stdPrivacyexten-- Added extension 's-DONTCALL' priority 1 to macro-stdPrivacyexten-- Added extension 's-TORTURE' priority 1 to macro-stdPrivacyexten-- Added extension '_s-.' priority 1 to macro-stdPrivacyexten -- Added extension 'a' priority 1 to macro-stdPrivacyexten -- Registered extension context 'macro-page'-- Added extension 's' priority 1 to macro-page-- Added extension 's' priority 2 to macro-page-- Added extension 's' priority 3 to macro-page-- Added extension 's' priority 4 to macro-page-- Added extension 's' priority 5 to macro-page-- Added extension 's' priority 6 to macro-page-- Added extension 's' priority 7 to macro-page-- Registered extension context 'demo'-- Added extension 's' priority 1 to demo-- Added extension 's' priority 2 to demo-- Added extension 's' priority 3 to demo-- Added extension 's' priority 4 to demo-- Added extension 's' priority 5 to demo-- Added extension 's' priority 6 to demo-- Added extension 's' priority 7 to demo-- Added extension '2' priority 1 to demo-- Added extension '2' priority 2 to demo-- Added extension '3' priority 1 to demo-- Added extension '3' priority 2 to demo-- Added extension '1000' priority 1 to demo-- Added extension '1234' priority 1 to demo-- Added extension '1234' priority 2 to demo -- Added extension '1235' priority 1 to demo -- Added extension '1236' priority 1 to demo -- Added extension '1236' priority 2 to demo -- Added extension '#' priority 1 to demo-- Added extension '#' priority 2 to demo-- Added extension 't' priority 1 to demo-- Added extension 'i' priority 1 to demo-- Added extension '500' priority 1 to demo -- Added extension '500' priority 2 to demo -- Added extension '500' priority 3 to demo -- Added extension '500' priority 4 to demo -- Added extension '600' priority 1 to demo -- Added extension '600' priority 2 to demo -- Added extension '600' priority 3 to demo -- Added extension '600' priority 4 to demo -- Added extension '76245' priority 1 to demo -- Added extension '_7XXX' priority 1 to demo -- Added extension '7999' priority 1 to demo -- Added extension '7999' priority 2 to demo -- Added extension '8500' priority 1 to demo -- Added extension '8500' priority 2 to demo -- Registered extension context 'page'-- Added extension '_X.' priority 1 to page-- Registered extension context 'default'-- Including context 'demo' in context 'default' -- Registered extension context 'test'-- Added extension '1001' priority 1 to test-- Added extension '1002' priority 1 to test-- Added extension '1003' priority 1 to test-- Added extension '1004' priority 1 to test== Parsing '/etc/asterisk/users.conf': FoundDialplan reloaded.*CLI>重新加载SIP方案：sip reload*CLI> sip reload*CLI> Reloading SIP== Parsing '/etc/asterisk/sip.conf': Found== Parsing '/etc/asterisk/users.conf': Found== Parsing '/etc/asterisk/sip_notify.conf': Found*CLI>=========================sip 控制台信息=====================================1. 运行调试常用命令./asterisk & 在后台运行./asterisk -vvvc 启动并尽量多的在控制台显示调试信息./asterisk -r 连接控制台set verbose 9999 设置打印详细信息2. 控制台下常用命令sip reload 重新加载sip信息sip set debug 设置显示更多的sip信息sip set debug off 关闭显示更多的sip信息sip show channels 显示所有活动的sip通道sip show users 显示用户情况sip show user xxx 显示用户xxx的情况dialplan show 显示拨号方案dialplan reload 重新加载拨号方案stop gracefully 优雅的停止asteriskstop now 立即停止运行module show 显示已有的加载module load cdr_radius.so 加载模块 cdrxxxmodule reload cdr_radius.so 重新加载模块 cdrxxxmodule unload cdr_radius.so 卸载模块 cdrxxxcore show applications 显示所有的应用比如Dial Radius Hangup Answer等core show channels 显示所有的通话通道core show codecs 显示所有支持的编码core show functions 显示所有的功能函数-------------------------------------------------------------------------*CLI> help! Execute a shell commandabort halt Cancel a running haltael debug contexts Enable AEL contexts debug (does nothing)ael debug macros Enable AEL macros debug (does nothing) ael debug read Enable AEL read debug (does nothing)ael debug tokens Enable AEL tokens debug (does nothing) ael nodebug Disable AEL debug messagesael reload Reload AEL configurationagent logoff Sets an agent offlineagent show Show status of agentsagent show online Show all online agentsagi debug Enable AGI debuggingagi debug off Disable AGI debuggingagi dumphtml Dumps a list of agi commands in html format agi show List AGI commands or specific helpcdr status Display the CDR statusconsole active Sets/displays active consoleconsole answer Answer an incoming console callconsole autoanswer Sets/displays autoanswerconsole boost Sets/displays mic boost in dBconsole dial Dial an extension on the consoleconsole flash Flash a call on the consoleconsole hangup Hangup a call on the consoleconsole mute Disable mic inputconsole send text Send text to the remote deviceconsole transfer Transfer a call to a different extensionconsole unmute Enable mic inputcore clear profile Clear profiling infocore set debug channel Enable/disable debugging on a channelcore set debug Set level of debug chattinesscore set debug off Turns off debug chattinesscore set global Set global dialplan variablecore set verbose Set level of verbosenesscore show applications Shows registered dialplan applicationscore show application Describe a specific dialplan applicationcore show audio codecs Displays a list of audio codecscore show channels Display information on channelscore show channel Display information on a specific channel core show channeltypes List available channel typescore show channeltype Give more details on that channeltypecore show codecs Displays a list of codecscore show codec Shows a specific codeccore show config mappings Display config mappings (file names to config engines)core show file formats Displays file formatscore show file version List versions of files used to build Asteriskcore show functions Shows registered dialplan functionscore show function Describe a specific dialplan functioncore show globals Show global dialplan variablescore show hints Show dialplan hintscore show image codecs Displays a list of image codecscore show image formats Displays image formatscore show license Show the license(s) for this copy of Asteriskcore show profile Display profiling infocore show switches Show alternative switchescore show threads Show running threadscore show translation Display translation matrixcore show uptime Show uptime informationcore show version Display version infocore show video codecs Displays a list of video codecscore show warranty Show the warranty (if any) for this copy of Asteriskdatabase del Removes database key/valuedatabase deltree Removes database keytree/valuesdatabase get Gets database valuedatabase put Adds/updates database valuedatabase show Shows database contentsdatabase showkey Shows database contentsdialplan add extension Add new extension into contextdialplan add ignorepat Add new ignore patterndialplan add include Include context in other contextdialplan reload Reload extensions and *only* extensionsdialplan remove extension Remove a specified extensiondialplan remove ignorepat Remove ignore pattern from contextdialplan remove include Remove a specified include from contextdialplan save Save dialplandialplan show Show dialplandnsmgr reload Reloads the DNS manager configurationdnsmgr status Display the DNS manager statusdundi debug Enable DUNDi debuggingdundi flush Flush DUNDi cachedundi lookup Lookup a number in DUNDidundi no debug Disable DUNDi debuggingdundi no store history Disable DUNDi historic recordsdundi precache Precache a number in DUNDidundi query Query a DUNDi EIDdundi show entityid Display Global Entity IDdundi show mappings Show DUNDi mappingsdundi show peers Show defined DUNDi peersdundi show peer Show info on a specific DUNDi peerdundi show precache Show DUNDi precachedundi show requests Show DUNDi requestsdundi show trans Show active DUNDi transactionsdundi store history Enable DUNDi historic recordsfeature show Lists configured featuresfile convert Convert audio filegroup show channels Display active channels with group(s) help Display help list, or specific help on a commandhttp show status Display HTTP server statusiax2 provision Provision an IAX deviceiax2 prune realtime Prune a cached realtime lookupiax2 reload Reload IAX configurationiax2 set debug Enable IAX debuggingiax2 set debug jb Enable IAX jitterbuffer debuggingiax2 set debug jb off Disable IAX jitterbuffer debuggingiax2 set debug off Disable IAX debuggingiax2 set debug trunk Enable IAX trunk debuggingiax2 set debug trunk off Disable IAX trunk debuggingiax2 show cache Display IAX cached dialplaniax2 show channels List active IAX channelsiax2 show firmware List available IAX firmwaresiax2 show netstats List active IAX channel netstatsiax2 show peers List defined IAX peersiax2 show peer Show details on specific IAX peeriax2 show provisioning Display iax provisioningiax2 show registry Display IAX registration statusiax2 show stats Display IAX statisticsiax2 show threads Display IAX helper thread infoiax2 show users List defined IAX usersiax2 test losspct Set IAX2 incoming frame loss percentage indication add Add the given indication to the countryindication remove Remove the given indication from the countryindication show Display a list of all countries/indicationskeys init Initialize RSA key passcodeskeys show Displays RSA key informationlocal show channels List status of local channelslogger mute Toggle logging output to a consolelogger reload Reopens the log fileslogger rotate Rotates and reopens the log fileslogger show channels List configured log channelsmanager show command Show a manager interface commandmanager show commands List manager interface commandsmanager show connected List connected manager interface usersmanager show eventq List manager interface queued events manager show users List configured manager usersmanager show user Display information on a specific manager usermgcp audit endpoint Audit specified MGCP endpointmgcp reload Reload MGCP configurationmgcp set debug Enable MGCP debuggingmgcp set debug off Disable MGCP debuggingmgcp show endpoints List defined MGCP endpointsmixmonitor Execute a MixMonitor command.module load Load a module by namemodule reload Reload configurationmodule show List modules and infomodule show like List modules and infomodule unload Unload a module by namemoh reload Music On Holdmoh show classes List MOH classesmoh show files List MOH file-based classesno debug channeloriginate Originate a callqueue add member Add a channel to a specified queuequeue remove member Removes a channel from a specified queuequeue show Show status of a specified queuerealtime load Used to print out RealTime variables.realtime pgsql status Shows connection information for the Postgresql RealTime driverrealtime update Used to update RealTime variables.restart gracefully Restart Asterisk gracefullyrestart now Restart Asterisk immediatelyrestart when convenient Restart Asterisk at empty call volumertcp debug ip Enable RTCP debugging on IPrtcp debug Enable RTCP debuggingrtcp debug off Disable RTCP debuggingrtcp stats Enable RTCP statsrtcp stats off Disable RTCP statsrtp debug ip Enable RTP debugging on IPrtp debug Enable RTP debuggingrtp debug off Disable RTP debuggingsay load set/show the say modeshow parkedcalls Lists parked callsshow queuessip history Enable SIP historysip history off Disable SIP historysip notify Send a notify packet to a SIP peersip prune realtime Prune cached Realtime object(s)sip prune realtime peer Prune cached Realtime peer(s)sip prune realtime user Prune cached Realtime user(s)sip reload Reload SIP configurationsip set debug Enable SIP debuggingsip set debug ip Enable SIP debugging on IPsip set debug off Disable SIP debuggingsip set debug peer Enable SIP debugging on Peernamesip show channels List active SIP channelssip show channel Show detailed SIP channel infosip show domains List our local SIP domains.sip show history Show SIP dialog historysip show inuse List all inuse/limitssip show objects List all SIP object allocationssip show peers List defined SIP peerssip show peer Show details on specific SIP peersip show registry List SIP registration statussip show settings Show SIP global settingssip show subscriptions List active SIP subscriptionssip show users List defined SIP userssip show user Show details on specific SIP userskinny reset Reset Skinny device(s)skinny set debug Enable Skinny debuggingskinny set debug off Disable Skinny debuggingskinny show devices List defined Skinny devicesskinny show lines List defined Skinny lines per devicesoft hangup Request a hangup on a given channelstop gracefully Gracefully shut down Asteriskstop now Shut down Asterisk immediatelystop when convenient Shut down Asterisk at empty call volumestun debug Enable STUN debuggingstun debug off Disable STUN debuggingudptl debug Enable UDPTL debuggingudptl debug ip Enable UDPTL debugging on IPudptl debug off Disable UDPTL debuggingvoicemail show users List defined voicemail boxesvoicemail show zones List zone message formats*CLI> [Jan 20 16:15:05] NOTICE[6579]: chan_sip.c:15094 handle_request_subscribe: Received SIP subscribe for peer without mailbox: 1001。

asterisk常用命令

asterisk常⽤命令Asterisk常⽤命令1：amportal命令，这个东东是freepbx搞的，它是负责管控asterisk。

我们看⼀下系统的进程如上图所⽰，asterisk就是咱们的主⾓，⽽safe_asterisk就是垂帘听政，管它的。

当asterisk进程被kill了时，safe_asterisk会试图重新启动它。

这对于稳定性⾄关重要哦。

amportal start 启动asteriskamportal restart 重启asteriskamportal stop 关闭asterisk2：asterik命令Asterisk 命令可以启动asterisk程序，后⾯可以加参数，⽐如asterisk -vvvvvv v越多，debug的等级越⾼，控制台输出越详细。

如果asterisk已经启动，可以⽤asterisk -r命令连接到它的控制台，在这⾥⾯可以执⾏CLI的命令，管控asterisk3:CLI命令sip show peers 查看sip电话设备的注册情况，如下图这⾥显⽰了分机号、ip、端⼝以及状态，“ok”表⽰正常iax2 show peers 查看iax电话设备的注册情况，如下图show hints 查看⼯号和分机设备的对应绑定情况，如下图这⾥显⽰了⼯号、设备类型/设备号、状态。

状态有以下⼏种Unavailable: 不可⽤，⼀般是由于设备没注册上Idle :空闲InUse :正在使⽤show queues 查看队列情况，包括了队列分机列表、状态（这个状态有时候貌似不准确，应该以show hints的为准----个⼈观点）。

reload 重装配置⽂件。

module load xxx载⼊xxx模块，xxx是模块的名称，如果不知道，可以按tab补全。

⽐如载⼊cdr_addon_mysql.so模块，这个模块负责将保存通话记录到mysql数据库⾥，它是通过asterisk-addon软件包安装的，在/etc/asterisk/cdr_mysql.conf这个⽂件中配置数据库的对接信息。

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Asterisk使用命令收藏运行调试Asterisk 1. 运行调试常用命令./asterisk & 在后台启动asterisk ./asterisk –vvvc 启动asterisk并尽量多的在控制台显示调试信息./asterisk –r 连接asterisk控制台set verbose 9999 设置打印详细信息/usr/sbin/asterisk -cvvv 启动并连接到CLI，使用3级复杂度调试/usr/sbin/asterisk -vvvr 连接到CLI，使用3级复杂度调试/usr/sbin/asterisk命令参数* -c console 控制以，用于连接到CLI界面* -v 简繁程度，设置CLI的调试信息的多少* -g 核心。

如果asterisk突然瘫痪，在接下来的时间，为了用gdb来做trace，可以产生一个核心文件* -r 远程。

用来远程重新连接到一个已经运行的asterisk进程中。

* -rx “restart now”执行一个CLI命令，而不需要连接到CLI并手动输入命令 2.asterisk控制台下常用命令sip show subscriptions sip notify Send a notify packet to a SIP peer sip prune realtime [peer|user| Prune cached Realtime users/peers sip reload Reload SIP configuration sip set debug {on|off|ip|peer} Enable/Disable SIP debugging sip set history {on|off} Enable/Disable SIP history sip show {channels|subscriptio List active SIP channels/subscriptions sip show channel Show detailed SIP channel info sip show domains List our local SIP domains. sip show history Show SIP dialog history sip show inuse List all inuse/limits sip show objects List all SIP object allocations sip show peers List defined SIP peers sip show peer Show details on specific SIP peer sip show registry List SIP registration status sip show settings Show SIP global settings sip show tcp List TCP Connections sip show users List defined SIP users sip show user Show details on specific SIP user sip unregister Unregister (force expiration) a SIP peer from the registery dialplan show 显示拨号方案dialplan reload重新加载拨打方案stop gracefully 优雅地停止asterisk stop now 立即停止运行asterisk module show 显示所有已加载模块module load cdr_radius.so 加载模块cdr_radius.so module reload cdr_radius.so 重新加载模块cdr_radius.so module unload cdr_radius.so 卸载模块cdr_radius.so core show applications 显示所有的应用，比如Dial, Radius, Hangup, Answer等core show channels 显示所有的通话频道core show codecs 显示所有支持的编解码器core show functions 显示所有的功能函数For more information on applications, just type "core show applications" at your ; friendly Asterisk CLI prompt. ; ; "core show application <command>" will show details of how you ; use that particular application in this file, the dial plan. ; "core show functions" will list all dialplan functions ; "core show function <COMMAND>" will show you more information about ; one function. Remember that function names are UPPER CASE.本文来自CSDN博客，转载请标明出处：/jrbu59/archive/2008/06/22/2575229.aspx2007-05-13 18:30asterisk1*CLI> help! Execute a shell commandabort halt Cancel a running haltadd extension Add new extension into contextadd ignorepat Add new ignore patternadd queue member Add a channel to a specified queueael debug contexts Enable AEL contexts debugael debug macros Enable AEL macros debugael debug read Enable AEL read debugael debug tokens Enable AEL tokens debugael no debug Disable AEL debug messagesael reload Reload AEL configurationagent logoff Sets an agent offlineagi debug Enable AGI debuggingagi no debug Disable AGI debuggingcdr mysql status Show connection status of cdr_mysqlcdr status Display the CDR statusdatabase del Removes database key/valuedatabase deltree Removes database keytree/valuesdatabase get Gets database valuedatabase put Adds/updates database valuedatabase show Shows database contentsdatabase showkey Shows database contentsdebug channel Enable debugging on a channeldebug level Set global debug leveldnsmgr reload Reloads the DNS manager configurationdnsmgr status Display the DNS manager statusdont include Remove a specified include from context dump agihtml Dumps a list of agi command in html format dundi debug Enable DUNDi debuggingdundi flush Flush DUNDi cachedundi lookup Lookup a number in DUNDidundi no debug Disable DUNDi debuggingdundi no store history Disable DUNDi historic recordsdundi precache Precache a number in DUNDidundi query Query a DUNDi EIDdundi show entityid Display Global Entity IDdundi show mappings Show DUNDi mappingsdundi show peers Show defined DUNDi peersdundi show peer Show info on a specific DUNDi peerdundi show precache Show DUNDi precachedundi show requests Show DUNDi requestsdundi show trans Show active DUNDi transactionsdundi store history Enable DUNDi historic recordsextensions reload Reload extensions and *only* extensions feature show channels Show status of feature channelsgroup show channels Show active channels with group(s)help Display help list, or specific help on a command iax2 debug Enable IAX debuggingiax2 jb debug Enable IAX jitterbuffer debuggingiax2 no debug Disable IAX debuggingiax2 no jb debug Disable IAX jitterbuffer debuggingiax2 no trunk debug Disable IAX trunk debuggingiax2 provision Provision an IAX deviceiax2 prune realtime Prune a cached realtime lookupiax2 reload Reload IAX configurationiax2 set jitter Sets IAX jitter bufferiax2 show cache Display IAX cached dialplaniax2 show channels Show active IAX channelsiax2 show firmware Show available IAX firmwaresiax2 show netstats Show active IAX channel netstatsiax2 show peer Show details on specific IAX peeriax2 show peers Show defined IAX peersiax2 show provisioning Show iax provisioningiax2 show registry Show IAX registration statusiax2 show stats Display IAX statisticsiax2 show users Show defined IAX usersiax2 test losspct Set IAX2 incoming frame loss percentage iax2 trunk debug Enable IAX trunk debugginginclude context Include context in other contextindication add Add the given indication to the country indication remove Remove the given indication from the countryinit keys Initialize RSA key passcodesload Load a dynamic module by namelocal show channels Show status of local channelslogger reload Reopens the log fileslogger rotate Rotates and reopens the log fileslogger show channels List configured log channelsmeetme Execute a command on a conference or confereemgcp audit endpoint Audit specified MGCP endpointmgcp debug Enable MGCP debuggingmgcp no debug Disable MGCP debuggingmgcp reload Reload MGCP configurationmgcp show endpoints Show defined MGCP endpointsmixmonitor Execute a MixMonitor commandmoh classes show List MOH classesmoh files show List MOH file-based classesmoh reload Music On Holdno debug channel Disable debugging on a channelodbc connect Connect to ODBC DSNodbc disconnect Disconnect from ODBC DSNodbc show Show ODBC DSN(s)pri debug span Enables PRI debugging on a spanpri intense debug span Enables REALLY INTENSE PRI debuggingpri no debug span Disables PRI debugging on a spanpri set debug file Sends PRI debug output to the specified file pri show debug Displays current PRI debug settingspri show span Displays PRI Informationpri unset debug file Ends PRI debug output to filerealtime load Used to print out RealTime variables.realtime mysql status Shows connection information for the MySQL RealTime d riverrealtime update Used to update RealTime variables.reload Reload configurationremove extension Remove a specified extensionremove ignorepat Remove ignore pattern from contextremove queue member Removes a channel from a specified queue restart gracefully Restart Asterisk gracefullyrestart now Restart Asterisk immediatelyrestart when convenient Restart Asterisk at empty call volumertp debug Enable RTP debuggingrtp debug ip Enable RTP debugging on IPrtp no debug Disable RTP debuggingset debug Set level of debug chattinessset verbose Set level of verbosenessshow agents Show status of agentsshow agi Show AGI commands or specific helpshow applications Shows registered dialplan applications show application Describe a specific dialplan application show audio codecs Shows audio codecsshow channel Display information on a specific channel show channels Display information on channelsshow channeltypes Show available channel typesshow codecs Shows codecsshow codec Shows a specific codecshow conferences Show status of conferencesshow config mappings Show Config mappings (file names to config engines)show dialplan Show dialplanshow features Lists configured featuresshow file formats Displays file formatsshow functions Shows registered dialplan functionsshow function Describe a specific dialplan functionshow hints Show dialplan hintsshow image codecs Shows image codecsshow image formats Displays image formatsshow indications Show a list of all country/indications show keys Displays RSA key informationshow license Show the license(s) for this copy of Asteriskshow manager command Show a manager interface commandshow manager commands List manager interface commandsshow manager connected Show connected manager interface users show modules List modules and infoshow modules like List modules and infoshow parkedcalls Lists parked callsshow queue Show status of a specified queueshow queues Show status of queuesshow switches Show alternative switchesshow translation Display translation matrixshow uptime Show uptime informationshow version Display version infoshow version files Show versions of files used to build Asteriskshow video codecs Shows video codecsshow voicemail users List defined voicemail boxesshow voicemail zones List zone message formatsshow warranty Show the warranty (if any) for this copy of Asterisksip debug Enable SIP debuggingsip debug ip Enable SIP debugging on IPsip debug peer Enable SIP debugging on Peernamesip history Enable SIP historysip no debug Disable SIP debuggingsip no history Disable SIP historysip notify Send a notify packet to a SIP peersip prune realtime Prune cached Realtime object(s)sip prune realtime peer Prune cached Realtime peer(s)sip prune realtime user Prune cached Realtime user(s)sip reload Reload SIP configurationsip show channels Show active SIP channelssip show channel Show detailed SIP channel infosip show domains List our local SIP domains.sip show history Show SIP dialog historysip show inuse List all inuse/limitssip show objects Show all SIP object allocationssip show peer Show details on specific SIP peersip show peers Show defined SIP peerssip show registry Show SIP registration statussip show settings Show SIP global settingssip show subscriptions Show active SIP subscriptionssip show users Show defined SIP userssip show user Show details on specific SIP userskinny debug Enable Skinny debuggingskinny no debug Disable Skinny debuggingskinny show devices Show defined Skinny devicesskinny show lines Show defined Skinny lines per devicesoft hangup Request a hangup on a given channelstop gracefully Gracefully shut down Asteriskstop now Shut down Asterisk immediatelystop when convenient Shut down Asterisk at empty call volume unload Unload a dynamic module by namezap destroy channel Destroy a channelzap show cadences List cadenceszap show channels Show active zapata channelszap show channel Show information on a channelzap show status Show all Zaptel cards status。