一个简单的Asterisk自动话务台例子.doc

一个简单的Asterisk自动话务台例子这是一个简单的Asterisk自动话务台的例子。我是根据Asterisk: The Definitive Guide中例子改写的。如有什么疑问，请给我邮件：。相互讨论，共同进步。这个文档只是介绍简单的话务台的制作过程，一些细节，由于时间有限，并没有详细介绍，请勿见怪。1. 创建菜单首先要设定好菜单的流程，包括：2. 创建对应的语音文件首先要创建对应的语音文件。语音文件的格式是：16bit单声道，8000Hz的WAV文件。我使用Audacity 2.0.2来直接录制这一格式的语音文件。Audacity是一个免费的软件，下载地址是：/我是通过360软件管家找到这个软件的。这个软件可以录音，选择录音片段，导出相应文件。录音前要选好单声道和采样频率。按照上面说的，录制下面语音文件：Sample promptFilename欢迎来到中国银行。daygreeting.wav欢迎来到中国银行，我们下班了。nightgreeting.wav如果你知道你需要拨打的分机，请直接输入。销售部，请拨1；服务支持请拨2，公司目录，请拨井号；公司地址和传真，请拨3；重复当前选项，请拨9；等待或拨0转人工。mainmenu.wav我们在为你转接到销售组，请不要挂机。holdwhileweconnect.wav我们在为你转接到支持组，请不要挂机。holdwhileweconnect_support.wav我们的地址是：广州市天河区天河北路中信大厦28楼。我们的传真机号码faxandaddress.wav转人工，请稍候。transfertoreception.wav不能解读你的输入信息，请重试。invalid.wav把做好的文件保存到/var/lib/asterisk/sounds/aa目录中，具体如下：rootkernel aa# ls /var/lib/asterisk/sounds/aa/daygreeting.wav invalid.wavfaxandaddress.wav mainmenu.wavholdwhileweconnect_support.wav nightgreeting.wavholdwhileweconnect.wav transfertoreception.wav3. 创建拨号规则我把SIP创建的帐号的context都指定到jluotest，当拨打1000号的时候进入main_menu。进入main_menu后，首先会执行s开头的拨号规则，这样就会播出欢迎辞和主菜单内容。欢迎辞会根据时间不同而不同。generalstatic=yeswriteprotect=noautofallthrough=yesclearglobalvars=noextenpatternmatchnew=noglobalsCONSOLE = Console/dspIAXINFO = guestTRUNK = DAHDI/G2TRUNKMSD = 1FEATURES =DIALOPTIONS =RINGTIME = 20FOLLOWMEOPTIONS =PAGING_HEADER = Intercommain_menuexten = #,1,Verbose(1, Caller $CALLERID(all) is entering the directory)exten = #,2,Directory(default)exten = 0,1,Verbose(1, Caller $CALLERID(all) is calling the operator)exten = 0,2,Dial(SIP/2134)exten = 1,1,Verbose(1, Caller $CALLERID(all) has entered the sales queue)exten = 1,2,Playback(aa/holdwhileweconnect)exten = 1,3,Queue(jluo)exten = 2,1,Verbose(1, Caller $CALLERID(all) has entered the service queue)exten = 2,2,Playback(aa/holdwhileweconnect_support)exten = 2,3,Queue(jluo)exten = 3,1,Verbose(1, Caller $CALLERID(all) has requested address and fax info)exten = 3,2,Playback(aa/faxandaddress)exten = 3,3,Goto(s,menuprompt)exten = i,1,Verbose(1, Caller $CALLERID(all) has entered an invalid selection)exten = i,2,Playback(invalid)exten = i,3,Goto(s,menuprompt)exten = s,1,Verbose(1, Caller $CALLERID(all) has entered the auto attendant)exten = s,2,Answer()exten = s,3,Set(TIMEOUT(digit)=10)exten = s,4,Wait(1)exten = s,5,GotoIfTime(9:00-17:00,mon-fri,*,*?daygreeting:afterhoursgreeting)exten = s,6(afterhoursgreeting),Background(aa/nightgreeting)exten = s,7,Goto(menuprompt)exten = s,8(daygreeting),Background(aa/daygreeting)exten = s,9,Goto(menuprompt)exten = s,10(menuprompt),Background(aa/mainmenu)exten = s,11,WaitExten(4)exten = s,12,Goto(0,1)exten = t,1,Verbose(1, Caller $CALLERID(all) has timed out)exten = t,2,Goto(0,1)exten = _2XXX,1,Verbose(1,Call $CALLERID(all) to an extension starting with 1)exten = _2XXX,2,Dial(SIP/$EXTEN)jluotestexten = 1000,1,Goto(main_menu,s,1)exten = 123,1,Answer()exten = 123,2,Playback(aa/mainmenu)exten = 123,4,Wait(4)exten = 123,5,Hangup()exten = 200,1,Answer()exten = 200,2,Playback(hello-world)exten = 200,3,Hangup()exten = 201,1,Answer()exten = 201,2,Set(MagicNumber=42)exten = 201,3,SayNumber($MagicNumber)exten = 201,4,Waite(3)exten = 201,5,Hangup()exten = 202,1,Answer()exten = 202,2,SayDigits($EXTEN:1)exten = 203,1,Meetme(1234)exten = 2133,hint,SIP/2133exten = 500,1,AGI(hello-world.sh)exten = 6000,1,Queue(jluo)exten = _2XXX,1,Dial(SIP/$EXTEN)exten = _2XXX,2,Hangup()4. 队列配置在上面的拨号规则中，使用到了队列。Asterisk可以定义出若干条队列，每条队列中有若干个成员。拨号规则可以使用Queue应用来拨打队列，队列自动选择空闲的用户，建立一个通道，实现语音通信。相关的配置文件如下：rootkernel # vi /etc/asterisk/queues.confjluomember = SIP/2134musicclass = defaultannounce = custom/test上面内容只是文件中的一个节选，主要是把SIP帐号2134加入到队列中。可以根据实际情况加入实际的用户信息。2134需要在sip.conf中配置。5. Sip.conf的配置。SIP帐号是我使用sip.conf例子模板做了一点修改创建出来的帐号。大概修改如下：my-codecs(!); a template for my preferred codecs; disallow=allallow = ulawallow = ilbcallow = g729allow = gsmallow = g723; Or, more simply:;allow=!all,ilbc,g729,gsm,g723,ulawulaw-phone(!); and another one for ulaw-only;disallow=allallow = ulawallow = ilbcallow = g729allow = gsmallow = g723; Again, more simply:;allow=!all,ulawjluo(!)context = jluotest