找台linux先装上用软电话玩玩吧。
如何跨入“第五代呼叫中心”
呼叫中心与人们的日常生活密切相关，人们对呼叫中心的理解也是相当宽泛的。电信运营商设立的“114”特服电话，就被认为是早期一个比较典型的呼叫中心。接着，大量声讯台、寻呼台普遍采用自动应答系统提供服务，这也被称为呼叫中心服务。现在电信运营商已建成多个呼叫中心，如10000/10001和10086等，都透过其方便快捷的服务，使呼叫中心的概念深入民心。呼叫中心已经发展成为一种产业，它不仅可以为企业创造良好的社会效益，而且还可以为企业带来巨大的经济效益。随着近年通信技术与计算机技术不断融合和发展，你会发现如今呼叫中心产业的发展已经开始跨入一个新的层次和阶段，网络技术革命的脚步已悄然走来。
呼叫中心发展历程回顾
1）第一代呼叫中心：人工热线电话系统
呼叫中心，早期是指一个由两人或更多人组成的、在一个特定地方用专用设备处理电话业务的小组。这些人就是通常所说的呼叫中心代理(人)。一个呼叫中心可以只提供信息接收服务，或者只提供信息发送服务，或者是一个混合式呼叫中心，其呼叫中心代理会负责所有这两项工作。
第一代呼叫中心的特点：硬件设备为普通电话机或小交换机（排队机），简单、造价低、功能简单、自动化程度低，一般仅用于受理用户投诉、咨询；适合小企业或业务量小、用户要求不高的企业/单位使用。目前，没有正式设立呼叫中心的企业、单位一般采用这种方式。
2）第二代呼叫中心：交互式自动语音应答系统
随着计算机技术和通信技术的发展，第一代呼叫中心由于基本靠人工操作，对话务员的要求相当高，而且劳动强度大、功能差，已明显不适应时代发展的需要。因此，功能完善的第二代呼叫中心系统随即应运而生。
第二代呼叫中心特点：广泛采用了计算机技术，如通过局域网技术实现数据库数据共享；语音自动应答技术用于减轻话务员的劳动强度，减少出错率；采用自动呼叫分配器均衡座席话务量、降低呼损，提高客户的满意度等等。但第二代呼叫中心也存在一定的缺点：它需要采用专用的硬件平台与应用软件，还需要投入大量资金用于集成和客户个性化需求，灵活性差、升级不方便、风险较大、造价也较高。
3）第三代呼叫中心：兼有自动语音和人工服务的客服系统
与第二代呼叫中心相比，第三代呼叫中心采用CTI技术实现了语音和数据同步。它主要采用软件来代替专用的硬件平台及个性化的软件，由于采用了标准化的通用的软件平台和通用的硬件平台，使得呼叫中心成为一个纯粹的数据网络。
第三代呼叫中心的优点：采用通用软硬件平台，造价较低；随着软件价格的不断下调，可以不断增加新功能，特别是中间件的采用，使系统更加灵活，系统扩容升级方便；无论是企业内部的业务系统还是企业外部的客户管理系统，不同系统间的互通性都得到了加强；同时还支持虚拟呼叫中心功能（远程代理）。
4）第四代呼叫中心：网络多媒体客服中心
接入和呼出方式多样化：电话、VOIP电话、计算机、传真机、手机短信息、WAP、寻呼机、电子邮件等。
多种沟通方式格式互换：可实现文本到语音、语音到文本、EMAIL到语音、EMAIL到短消息、EMAIL到传真、传真到EMAIL、语音到EMAIL等的自由转换。
语音自动识别技术：可自动识别语音，并实现文本与语音自动双向转换，即可实现人与系统的自动交流。
第五代呼叫中心的核心——免插件的网络实时互动服务
与第四代呼叫中心相比,免插件的网络实时互动服务,完全基于互联网传输成为了第五代呼叫中心的特点，将互联网技术与呼叫中心完整融合，在IM（即时通讯）工具C/S结构的基础上，与B/S结构进行嫁接，基于B/S构架，但不限于B/S构架；基本功能包含IM功能，但不限于IM功能；完全意义上实现了网络上的沟通服务.互联网呼叫中心的核心是增值服务功能，即在实现客户咨询的初级目标的基础上，实现直复营销这一高级目标。第五代呼叫中心在第四代的基础上大大增强了实时的交互功能，并利用最先进的网络技术开创了网站访客免安装任何插件，就可与企业后台CSR直接在线交流的先河。企业后台CSR也能随时看到所有登陆企业网站客户的浏览轨迹，并能主动邀请客户在线直接交流，彻底改变了呼叫中心只能被动等待客户呼如入的历史。换言之，呼叫中心不再仅仅是有效的用户交流工具，更是产品、服务的营销工具。第五代呼叫中心的设计重点主要集中在应用层面上，而不单单是硬件（如PABX）上，因此更能适应企业的要求，更有效的配合企业客户关系管理（CRM）的进程。
第五代呼叫中心的应用
1）互动(Interaction)–即根据客户偏好, 即时调整服务方式,包括语音、数据、视频和多媒体业务等，既从客户那里了解他们的需求，又主动去帮助他们“发现”和“启动”自己的需求，影响他们选择的基于分组技术综合开放的网络架构。第五代呼叫中心的企业和客户间交流突破了原有的语音沟通瓶颈，进入了丰富多彩的多媒体时代，通过各种文字，语音，图片，视频和客户的直接交流，更生动全面的向客户展示企业的形象和产品，解决客户的疑问，既提升了企业品牌，同时更促进了企业销售。使呼叫中心真正成为了全方位立体化的实时服务平台
2）协同(Coordination)–不仅包括人与人之间的协作，也包括不同应用系统之间、不同数据资源之间、不同终端设备之间、不同应用情景之间、人与机器之间、科技与传统之间等全方位的协同。
第五代呼叫中心的CSR通过统一的服务平台协同服务客户时，通过对服务记录的整体转移，可以对客户实现“我把相关客服转给您”而不是“我把您转给相关客服”处处做到以客户为中心的服务。通过第五代呼叫中心协同办公功能，CSR可以是在同一办公地点集中办公，也可以是分散式办公，甚至是在家办公，他们接受统一的远程管理，共享最新的公司资讯，进行有效的分工协作，充分利用公司的各项资源。
3）管理(Mangerment)–即以人为主导，利用计算机硬件、软件、网络通信设备以及其他办公设备，进行信息的收集、传输、加工、储存、更新和维护，以企业战略竞争、提高效益和效率为目的，支持企业高层决策、中层控制、基层运作的集成化的人机系统。
对客户的管理：第五代呼叫中心可以对所有网络数据进行监测和统计，对客户上网的轨迹跟踪记录，知道客户是什么地区，通过什么途径到达公司网站，第几次登陆，对哪个页面最感兴趣，通过分析了解客户的需求，有针对性的服务。并将服务过的客户资料及服务记录存储到客户数据库当中，当访客再次登陆时即可识别，并跟踪服务，提高客户满意度及忠诚度。
对客服的管理：所有CSR的服务记录都会保存在企业的服务器中，包括每天服务的客户总数，平均处理时间，述求应答数，客户满意度等，公司的管理人员可按级别查看服务记录，且能以各种报表形式输出。
通过第五代呼叫中心互动，协同，管理((Interaction Coordination Mangerment 简称为 “ICM” ) 的功能,可以使Web呼叫中心进入一个精准制导的市场营销时代:目标群体精确并且资源集优；巧妙的推广策略与潜移默化的渗透；市场情报的收集与研究；即时的各单元间协同办公；客户数据的收集、筛选、及时更新；即时的战略调整和创新。真正使Web呼叫中心形成包括线上、线下即时互动的全方位立体型的整体营销服务管理体系.
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1、安装内核源码
apt-get install linux-source-2.6.18
cd /usr/src
tar -xjf linux-source-2.6.18.tar.bz2
解压得到/usr/src/linux-source-2.6.18
2、源码连接到modules目录
cd /lib/modules/2.6.18-6-686
ln -s /usr/src/linux-source-2.6.18 build
3、进行配置和预编译
cd /usr/src/linux-source-2.6.18
vi Makefile
找到EXTRAVERSION 改为
EXTRAVERSION = -6-686
这个是 uname -r 显示的内核版本，里面的扩展字段，这个在后面安装的时候有点用吧，毕竟很多人的内核都是系统装好的，没有源代码的。
如果这里不修改，后面install的时候modules会安装在/lib/2.6.18/下面，虽然也能insert ，但是可能以后会有点问题啦，呵呵
make oldconfig
make prepare scripts
这步比较重要。。。因为用到genksyms等脚本，如果用
make prepare 或者 make prepare-all 都不会编译出来。。。这里搞得比较郁闷。。。
4、下载dahdi-linux-complete-2.1.0.4+2.1.0.2.tar.gz
解压，cd，
make all
make install
make config
然后配置好/etc/dahdi/*
/etc/init.d/dahdi start
启动，明天再搞配置文件.
Asterisk 安装笔记（2）- Zaptel 和 Dahdi 的配置
Zap Channel Module Configuration
The Zap channel module permits Asterisk to communicate with the Zaptel device driver, used to access Zaptel telephony interface cards. You configure Asterisk’s Zap channel module in the zapata.conf file.Zap channel模块允许Asterisk与zaptel驱动程序之间通讯。通过配置zapata.conf文件实现
You will need the Zaptel kernel module device driver installed. See:
Although TDMoE is not directly related to Zapata hardware, it uses a pseudo-TDM engine, and gets configured here.
Using MySQL For Zap Channel Configuration
It is possible to store configuration settings for the Zap channel driver in a MySQL table, rather than editing the zapata.conf text file. You will have to compile a version of Asterisk with this support built in. See:
可以把zap channel而配置存储在mysql表中，而不是zapatap.conf中，这需要版本支持
The rest of this page assumes you are editing the zapata.conf file by hand.
Creating Channels
The format of the zapata.conf file is unfortunately not as simple as it could be. Most keywords do not do anything by themselves; they merely set up the parameters of any channel definitions that follow. The channel keyword actually creates the channel, using the settings specified before it. For example, you might create two channels like this:zapata.conf文件，没有看上去那么复杂，大多数关键词自己不做什么，仅仅是定义通道参数，channel关键词才是真正的创建通道。
signalling=fxo_ks
language=en
context=reception
channel => 1
signalling=fxo_ks
language=fr
context=sales
channel => 2
This creates channel 1 with a default language code “en” and a context “reception”. Channel 2 has a default language code “fr” and context “sales”.
This is important, if you put something like echocancel=no before the channel definition, it will effect all channels unless you turn it on later with echocancel=yes. It progresses downward, but the definition must be above the channel=> statement.
非常重要的是，如果例如在通道前定义echocancel=no，会使影响所有通道，直到定义echocancel=yes，他会往下执行，因此，定义必须在channel=>前面进行定义
Available Settings
Signalling Type
The signalling type to use with your interface is the only mandatory setting. You must set a signalling type before allocating a channel. If you are connecting analog telephone equipment, note that analog phone signalling can be a source of some confusion. FXS channels are signalled with FXO signalling, and vice versa. Asterisk ‘talks’ to internal devices as the opposite side. An FXO interface card is signalled with FXS signalling by Asterisk, and should be configured as such.信令类型是唯一强制设置，在分配一个通道之前，必须定义信令类型。如果连接的模拟电话设备，注意模拟信令是导致混乱的来源。FXS通道采用FXO信令，反之，Asterisk通知内部设备采用相反方式。FXO接口卡采用FXS信令，同样须定义。
signalling: Sets the channel signaling type. These parameters should match the Zaptel driver configuration. The setting to use depends partly on which interface card you have. Asterisk will fail to start if a channel signaling definition is incorrect or unworkable, if the statements do not match the Zaptel driver configuration, or if the device is not present or properly configured. The correct setting to use is almost certainly one of the following four: fxs_ks, fxo_ks, pri_cpe or pri_net. This setting has no default value; you must set a value before allocating a channel. Asterisk supports the following signalling types:
signalling:设置通道信令类型，这些参数须与zaptel驱动配置匹配。设置基于采用什么样的板卡，如果通道信令设置错误，如果配置描述与zaptel驱动配置不匹配，或者如果卡不存在而配置正确，Asterisk不会工作。正确的设置通常包含下面4中信令中一种，fxs_ks, fxo_ks, pri_cpe or pri_net。该设置没有缺省值，必须在分配通道前设置信令值，下面是Asterisk支持的信令类型。
- em: E & M Immediate Start
- em_w: E & M Wink Start
- em_e1: E & M CAS signalling for E1 lines
- featd: Feature Group D (The fake, Adtran style, DTMF)
- featdmf_ta: Feature Group D (The real thing, MF (domestic, US)) through a Tandem Access point
- fgccama Feature Group C-CAMA (DP DNIS, MF ANI)
- fgccamamf Feature Group C-CAMA MF (MF DNIS, MF ANI)
- featdmf: Feature Group D (The real thing, MF (domestic, US))
- featb: Feature Group B (MF (domestic, US))
- fxs_ls: FXS (Loop Start)
- fxs_gs: FXS (Ground Start)
- fxs_ks: FXS (Kewl Start)
- fxo_ls: FXO (Loop Start)
- fxo_gs: FXO (Ground Start)
- fxo_ks: FXO (Kewl Start)
- pri_cpe: PRI signalling, CPE side
- pri_net: PRI signalling, Network side (for instance, side that provides the dialtone)
- sf: SF (Inband Tone) Signalling
- sf_w: SF Wink
- sf_featd: SF Feature Group D (The fake, Adtran style, DTMF)
- sf_featdmf: SF Feature Group D (The real thing, MF (domestic, US))
- sf_featb: SF Feature Group B (MF (domestic, US))
- e911: E911 (MF) style signalling. Originating switch goes off-hook, far-end winks, originating sends KP-911-ST, far-end gives answer supervision, Originating-end sends KP-0-ANI-ST
- The following are used for Radio interfaces:
- fxs_rx: Receive audio/COR on an FXS kewlstart interface (FXO at the channel bank)
- fxs_tx: Transmit audio/PTT on an FXS loopstart interface (FXO at the channel bank)
- fxo_rx: Receive audio/COR on an FXO loopstart interface (FXS at the channel bank)
- fxo_tx: Transmit audio/PTT on an FXO groundstart interface (FXS at the channel bank)
- em_rx: Receive audio/COR on an E&M interface (1-way)
- em_tx: Transmit audio/PTT on an E&M interface (1-way)
- em_txrx: Receive audio/COR AND Transmit audio/PTT on an E&M interface (2-way)
- em_rxtx: same as em_txrx (for our dyslexic friends)
- sf_rx: Receive audio/COR on an SF interface (1-way)
- sf_tx: Transmit audio/PTT on an SF interface (1-way)
- sf_txrx: Receive audio/COR AND Transmit audio/PTT on an SF interface (2-way)
- sf_rxtx: same as sf_txrx (for our dyslexic friends)
PRI通道存在一个空闲Extension和一个微小闲置数字，只要闲置通道是空闲的，ZAP通道模块就会尝试在该通道上进行空闲拨号，然后Asterisk就会执行定义为idelext的Context和Extension中的命令。当通道需要进行语音呼叫时，’空闲’呼叫会断开并让多数通道有效。（当然尽管有微小闲置呼叫正在运行）。主要的用途是创建动态的服务，当闲置通道绑定了multilnk
ppp协议后，将比传统的多重映射提供更有效率的提供综合的语音/数据服务。
minunused: The minimum
number of unused channels available. If there are fewer channels
available, Asterisk will not attempt to bundle any channels and give
them to the data connection. Takes an integer.minunused:最小可用闲置通道的数量。如果有很少的通道可用，Asterisk不会尝试捆绑任何通道进行数据连接。该参数需要一个整数。
minidle: The minimum number of idle channels to bundle for the data link. Asterisk will keep this number of channels open for data, rather than taking them back for voice channels when needed. Takes an integer.
minidle:最小绑定进行数据连接的通道数量，Asterisk会为数据开启这个通道数量，而不是在需要的时候返回到语音通道的使用上。该参数需要一个整数。
idledial: The number to dial as the idle number. This is typically the number to dial a Remote Access Server (RAS). Channels being idled for data will be sent to this extension. Takes an integer that does not conflict with any other extension in the Dialplan, and has been defined as an idleext.
idledial: 空闲拨号的数量，这是用于拨叫远程访问服务器最基本的一个数字，为数据预留的闲置通道被这个分机。该参数需要一个整数，与在拨号方案中定义了idleext的分机不会产生冲突。
idleext: The extension to use as the idle extension. Takes a value in the form of exten@context. Typically, the extension would be an extension to run the ZapRAS command.
idleext：用于空闲分机的extension，以exten@context的用法使用，典型的用法是被作为分机运行ZapRAS命令。
minunused=2
minidle=1
idledial=6999
idleext=6999@idle
Analog Trunk Features （模拟中继特征）
usedistinctiveringdetection: Whether or not to attempt to recognize distinctive ring styles on incoming calls. This does not require audio analyisis because rings are simple transitions of the analog line. It’s merely a matter of matching the transition pattern.usedistinctiveringdetection:是否尝试识别来电特殊铃音，这不需要音频分析，因为铃音在模拟线路上是非常简单转换，只需要匹配转换样本。缺省值：no
Default: no.
usedistinctiveringdetection=yes
dring1, dring2, dring3: If you set usedistinctiveringdetection=yes, then you may define up to three different distinctive ring styles for Asterisk to attempt to recognize. Each style is defined as a comma separated list of up to three integers. Nobody has yet documented what these numbers mean, so you’re on your own when it comes to trying to figure out what numbers to use for the distinctive ring syles used by your phone company in your country. But the tip is to use the Asterisk console in verbose mode, and apparently it reports numbers describing the ring patterns it sees. These patterns may be a starting point:
dring1, dring2, dring3：如果设置了usedistinctiveringdetection=yes，就需要定义三种不同特点的铃音风格，以便于Asterisk能够尝试识别。每种风格使用逗号分割三个整数来定义。没有文档说明三个数字的含义，因此需要自己测试鉴别在不同国家不同公司中，不同数字代表的风格。Asterisk控制台上也会显示识别的风格数字，具体风格可能会是以下一些情况。
dring1=96,0,0
dring2=325,95,0
dring3=367,0,0
dring1context, dring2context, dring3context: Along with setting up to three distinctive ring patterns with dring1, dring2 and dring3, you also set corresponding contexts for incoming calls matching those distinctive ring patterns to jump into. If an incoming call does not match any of the distinctive ring patterns defined, then of course it will enter Asterisk with the default context defined for this channel.
dring1context, dring2context, dring3context:根据三种不同的铃音风格设置不同的context进行来电跳转，如果来电没有定义的风格匹配，就会进入该通道缺省的congtext。
dring1context=line2incoming
dring2context=business
dring3context=chocolate
busydetect: If enabled, Asterisk will analyze the audio coming in on the line during a call or a dial attempt to attempt to recognize busy signals. This is useful on analog trunk interfaces both to detect a busy signal when dialing out, and for detecting when the person has hung up. See also Disconnect Supervision. Be sure that you don’t use this on digital interfaces like QuadBri cards and so on. Otherwise you will run in “broken calls” problems. default=no
busydetect：忙音检测，如果开启，Asterisk会拨号尝试或通话中分析在线的音频，从而尝试识别忙音信号。这非常在模拟中继接口上外呼时检测忙音信号非常有用，可以检测何时挂机。确认不能在例如QuadBri等卡上使用该参数，否则出现中断通话的问题，缺省值：no
busydetect=yes
busycount:
This option requires busydetect=yes. You can specify how many busy tones
to wait before hanging up. The default is 3, but better results can be
achieved if set to 6 or even 8. The higher the number, the more time is
needed to detect a disconnected channel, but the lower the probability
mistaking some other sound as being a busy tone.busycount=5
busycount:这个选项需要busydetect=yes，可以定义等待挂机的忙音信号数量，缺省值是3，但能达到的最好效果可能是设置6或者8，数字越高，检测挂机通道所需要的时间就越长，但小的数字可能会导致把其他声音错误的识别为忙音信号。
callprogress: Asterisk can attempt to monitor the state of the call to listen for a ringing tone, busy tone, congestion tone, and sounds indicating that the line has been answered. It appears that this feature is independent of the busydetect feature; it seems that both can run in parallel, and both will independently attempt to recognize a busy tone. The callprogress feature is highly experimental and can easily detect false answers, so don’t count on it being very accurate. Also, it is currently configured only for standard U.S. phone tones. Default: no.
callprogress：Asterisk可以通过尝试监控呼叫状态来侦听振铃音，忙音，拥塞音以及线路已经应答声音特征。这个特征不受busydetect特征影响，两者可以并行处理，独自尝试识别忙音信号。callprogress的特征是高实验证明更容易检测错误应答，所以不要指望它非常准确。因此，目前仅仅在标准美国电话铃音中配置，缺省值：no
callprogress = yes
pulse: The standard installation of Asterisk does not permit you to specify that a Zaptel device use pulse dialing, even though the Zaptel driver supports pulse dialing. But you can apply a patch file to enable you to specify pulse dialing with the pulse keyword. See Pulse Dialing on Zap Channels for the patch.
pulse：Asterisk标准安装中，没有允许定义Zaptel卡使用脉冲拨号，尽管Zaptel驱动支持脉冲拨号，但可以更新补丁文件，使用pulse关键字去开启脉冲拨号。
pulse=yes
Analog Handset Features 模拟电话特征
adsi: If your handset has ADSI (Analog Display Services Interface) capability, set set adsi=yes. The ADSI specification is system similar to Caller ID to pass encoded information to an analog handset. It allows the creation of interactive visual menus on a multiline display, offering access to services such as voicemail through a text interface.adsi:如果手持设备支持ADSI（模拟显示服务接口），设置set adsi=yes，ADSI类似来电显示功能，传递编码信息到手持设备。它可以在多行显示的手持设备上创建交互式可视化菜单，通过文本接口提供类似语音邮件的访问服务。
immediate: Normally (i.e. with immediate set to ‘no’, the default), when you lift an FXS handset, the Zaptel driver provides you a dialtone and listens for digits that you dial, passing them on to Asterisk. Asterisk waits until the number you’ve dialed matches an extension, and then begins executing the first command on the matching extension. If you set immediate=yes, then Asterisk will instruct the Zaptel driver to not generate a dialtone when you lift a handset, instead passing control immediately to Asterisk. Asterisk will start executing the commands for this channel’s “s” extension. This is sometimes referred to as “batphone mode”. Default: no.
immediate：通常（immediate设置为no，缺省值），当FXS话机挂机时，Zaptel驱动会马上提供拨号音，等待拨号并传递给Asterisk。Asterisk等到接收到extension匹配号码时，就会开始执行相应的命令，如果设置 immediate=yes，Asterisk会命令Zaptel驱动不要在FXS挂机时产生拨号音，而是把控制权交还给Asterisk，Asterisk会开始执行这个通道的s extension。这通常应用于batphone 模式（蝙蝠电话？），缺省No
immediate=yes
callwaiting: If enabled, Asterisk will generate “call waiting pips” when you are already in a conversation on your FXS handset when someone tries to call you. If the channel has call waiting by default, you can temporarily disable it by lifting the handset and dialing *70, whereupon you will get a dialrecall tone and may then dial the intended number. There is no corresponding way to temporarily enable call waiting for channels that have it off by default. Default: no.
callwaiting：如果开启，在通话过程中如果有来电时，Asterisk就会产生呼叫等待提示音。如果通道缺省有呼叫等待，可以临时摘机按键*70取消，这种情况下，会收到重播提示音去拨打希望拨打的号码。没有合适的方法临时开启缺省设置为关闭的通道的呼叫等待。缺省为no
callwaiting=yes
callwaitingcallerid: Sets whether Asterisk will send Caller ID data to the handset during call waiting indication. Requires also setting callwaiting=yes. Default: no.
callwaitingcallerid:设置在呼叫等待过程中是否传送主叫号码等数据，需要设置callwaiting=yes，缺省值：no
callwaitingcallerid=yes
threewaycalling: If enabled, you can place a call on hold by pressing a hook flash, whereupon you get a dialrecall tone and can make another call. Default: no.
threewaycalling：（三方通话）如果设置开启，可以在按保持键切换话路，让原通话处于保持状态，这时会收到重拨提示音，并开启另外一方通话。缺省值：no
threewaycalling=yes
transfer: This option has effect only when threewaycalling=yes. If threewaycalling=yes and transfer=yes, then once you’ve placed a call on hold with a hook flash, you can transfer that call to another extension by dialing the extension and hanging up. Default: no.
transfer:（呼叫转接）这个选项仅当三方通话=yes时有效，当设置了三方通话和呼叫转接，一旦通过或呼叫保持按键把当前话路置于保持状态，就可以拨号呼叫另外分机，把2个话路桥接起来，然后挂机。缺省值：no
transfer=yes
cancallforward: If enabled, you may activate “call
forwarding immediate” by dialing *72 (whereupon you get a dialrecall
tone) followed by the extension number you wish to forward your calls
to. If someone dials your extension, the call will be redirected to the
forwarding number. You may disable the call forwarding by dialing *73.
Default: no.transfer=yes
cancallforward：如果呼叫前转启用，可以通过拨号*72+想要转向的Extension，立刻激活呼叫前转（同时会有重拨提示音），这时如果有来话，那么呼叫会被重定向到设置的转移号码上，可以通过拨打*73取消呼叫前转。缺省值：no
cancallforward=yes
callreturn: If enabled, you may dial *69 to have Asterisk read to you the caller ID of the last person to call. You will hear the dialrecall tone if there is no record of a last caller. Default: no.
callreturn：如果开启此设置，可以通过拨打*69让Asterisk读出最后呼入的主叫号码，如果没有记录最后呼叫主叫号码，将听到重拨提示音，缺省值：no
callreturn=yes
callgroup: A channel may belong to zero or more callgroups. Callgroups specify who may answer this phone when it is ringing. If this channel is ringing, then any other channel whose pickupgroups include one of this channel’s callgroups may answer the call by dialing *8#. This feature is supported by Zap, SIP, Skinny and MGCP channels. Group numbers can range from 0 to 31. The default value is an empty string, i.e. no groups.
callgroup:通道可以不属于或者属于多个呼叫群组。呼叫群组定义了当电话振铃时，谁可以接听。当一个通道振铃时，其它那些pickupgroups中包含该通道callgroups其中之一的通道可以通过按*8#来接听电话。这个特性支持在ZAP，SIP。skinny和MGCP通道类型上使用，群组数字范围为0-31，，缺省值是空字符串，即没有组。
group=1
callgroup=1,2,3
pickupgroup: A channel may belong to zero or more pickupgroups. Pickupgroups specify whose phones you may answer. If another channel is ringing, and this channel’s pickupgroups include one of the ringing channel’s callgroups, then this channel may answer the call by dialing *8#. Group numbers can range from 0 to 31. The default value is an empty string, i.e. no groups.
pickupgroup：通道可以不属于或者属于多个摘机群组，摘机群组定义了可以应答那些电话。如果其他通道振铃，而本通道pickupgroup是振铃通道callgroups群组其中之一，那么本通道可以通过按*8#来接听振铃通道。群组范围为0-31，缺省值为空字符串，即没有群组。
group=1
See more about Channels and Groups
If you dial *8# when there is more than one channel whose calls you are eligible to answer, then it just answers the “first ringing channel”, i.e. you have no control which one you pick up.
如果同时不止一路通道振铃符合条件可以通过按键*8#接听，只能接听第一条振铃通道，即不能控制选择接听哪一条。
pickupgroup=3,4
useincomingcalleridonzaptransfer: If you set this option (Use Incoming Caller ID On Zap Transfer) to ‘yes’, then when you transfer a call to another phone, the original caller’s Caller ID will get forwarded on too. Default: no.
useincomingcalleridonzaptransfer：如果设置了这个选项（在ZAP通道上启用来电转接），可以转接来电到另外一个电话，外部呼叫的主叫号码同时跟随。
useincomingcalleridonzaptransfer=yes
Caller ID Options
callerid: Sets the Caller ID string to forward to the recipient when calls come in from this channel. You normally use this to set the Caller ID for handsets. Specify the Caller ID name in double quotation marks, followed by the Caller ID number in <> symbols. For trunk lines, set to “asreceived” to pass the received Caller ID forward.callerid：当来电时设置主叫ID字符串，传送给接收者，通常为手持设备设置callerID。定义callerid：双引号名字后紧跟角括号数字，对中继线路，设置asreceived来传送主叫ID。
callerid=”Mark Spencer” <256 428-6000>
callerid=
callerid=asreceived
Important Note: Caller ID can only be transmitted to the public phone network with supported hardware, such as a PRI. It is not possible to set external caller ID on analog lines.
重要事项：CallerID只能在硬件支持的公共电话交换网上被传输，例如PRI。在模拟线路上设置外部CallerID是不可能的。
usecallerid: For handsets, this option will cause Asterisk to send Caller ID data to the handset when ringing it. For trunk lines, this option causes Asterisk to look for Caller ID on incoming calls. Default: yes.
usecallerid:对于手持设备，这个选项可以在振铃时让Asterisk发送CallerID数据到到手持设备，对于中继线路，该选项致使Asterisk查找来电主叫ID，缺省值：yes
usecallerid=no
hidecallerid: (Not for FXO trunk lines) For PRI channels, this will stop the sending of Caller ID on outgoing calls. For FXS handsets, this will stop Asterisk from sending this channel’s Caller ID information to the called party when you make a call using this handset. FXS handset users may enable or disable sending of their Caller ID for the current call only by lifting the handset and dialing *82 (enable) or *67 (disable); you will then get a “dialrecall” tone whereupon you can dial the number of the extension you wish to contact. Default: no.
hidecallerid：主叫ID隐藏（不能应用于FXO中继线路），对于PRI通道，在外呼时停止传送主叫ID。对于FXS端外呼时，会停止发送主叫ID信息到被叫方。FXS端可以在话机上按*82（启用）*67（关闭）可以控制是否传送主叫ID传送。
hidecallerid=yes
restrictcid: (PRI channels only) This option has
effect only when hidecallerid=no. If hidecallerid=no and
restrictcid=yes, Asterisk will prevent the sending of the Caller ID data
as a presentation number when making outgoing calls (ANI data is still
sent). Default: no.hidecallerid=yes
restrictcid:（仅用于PRI通道），该选项在hidecallerid=no时可以有效设置，如果hidecallerd=no并且restrictcid=yes，外呼时，asterisk会阻止以显示号码方式发送主叫id，但ANI消息数据仍然发送），缺省为no
restrictcid=yes
usecallingpres: (PRI channels only) Whether or not to use the Caller ID presentation for the outgoing call that the calling switch is sending. See also the CallingPres command. Read more in this discussion from 2003.
usecallingpres:（仅PRI通道有效）不管是否把callerid作为外呼的显示号码，交换机都会传送。
usecallingpres=no
Audio Quality Tuning Options （音频质量调整选项）
These options adjust certain parameters of Asterisk that affect the audio quality of Zapata channels. See also:- Echo Cancellation on the Wildcard X100P
- Echo Cancellation in Asterisk
- Asterisk Echo Cancellation: FXO and FXS lines
- Asterisk Echo Avoidance
- Echo Analysis for Voice over IP
relaxdtmf:如果DTMF检测有问题，可以放宽DTMF检测的参数。
relaxdtmf=yes
echocancel: Disable or enable echo cancellation (default is ‘yes’). It is recommended that you do not turn this off. You may specify echocancel as ‘yes’ (128 taps), ‘no’ (0 taps, disabled), or a preset number of taps which are one of 16, 32, 64, 128, or 256. Each tap is one sample from the data stream, so on a T1 this will be 1/8000 of a second. Accordingly the number of taps equate to a 2ms, 4ms, 8ms, 16ms or 32ms tail length. Beware that if you set echocancel to a different value, Asterisk will fall back to the default of 128 taps without warning.
echocancel:开启或关闭回音消除（缺省值：是），建议不要关闭该设置，可以定义回音消除yes（128滤波参数）或者no（0滤波），或者定义参数为16,32,64,128,256中一个，每种滤波参数都是一种数据流样本，在T1线路上会是每秒1/8000，因此滤波参数值等于2ms,4ms,8ms,16ms,32ms尾长，如果设置的回音消除为不同的值，Asterisk将直接使用128而不会警告。
echocancel=no
echocancelwhenbridged: Enables or disables echo cancellation during a bridged TDM call. In principle, TDM bridged calls should not require echo cancellation, but often times audio performance is improved with this option enabled. Default: no.
echocancelwhenbridged:开启或关闭在桥接的TDM呼叫中的回音消除，原则：TDM桥接呼叫不需要回音消除，但开启这个选项通常可以提高语音效果。
echocancelwhenbridged=yes
echotraining: In some cases, the echo canceller doesn’t train quickly enough and there is echo at the beginning of the call which then quickly fades out. Enabling echo training will cause Asterisk to briefly mute the channel, send an impulse, and use the impulse response to pre-train the echo canceller so it can start out with a much closer idea of the actual echo. However, the characteristics of some trunks may change as the endpoints become connected and, if there is a considerable delay between the circuit being ‘up’ and the endpoints being finalised, the training impulse may measure the characteristics of the open trunk rather than the completed circuit. Accordingly you may either specify a value between 10ms and 4000ms to delay before starting the impulse response process or ‘yes’, which equates to 400ms. Default: undefined.
echotraining：有时回音消除不能够很快的自学习，通话开始时会有回音，然后很快消除。开启回音训练可以让Asterisk使通道暂时无声而发送一个刺激信号，并根据响应效果预训练回音消除，从而能够更接近真实的回音。然而如果在电路up和终端响应定位之间有相当的延时，某些典型中继被会作为终端进行连接，训练刺激信号会检测open中继的特性而不是实际电路。因此，在开始响应刺激信号处理之前，可以在10ms和4000ms延时之间定义一个值，或者定义yes，缺省就是400ms。默认值没有定义。
echotraining=no
rxgain: Adjusts receive gain. This is the audio recieved by Asterisk from the device. E.g: in a phone connected to a FXS channel, this would control the audio that is sent from the phone to Asterisk. This can be used to raise or lower the incoming volume to compensate for hardware differences. You specify gain as a decimal number from -100 to 100 representing dB. 10 is significantly high. Change these options by only a few dB at a time. Default value: 0.0
rxgain：调整接收获取强度值，这是指Asterisk从例如连接到FXS通道上的电话设备上接收到的音频，该选项能控制由电话发送给Asterisk的音频，可以用于提高或降低进入的声音音量，从而补偿硬件损耗。可以定义获得值从-100db到100db，10db就意味着很高了。修改时应进行微调。
rxgain=4.2
txgain: Adjusts transmit gain. This is the audio transmitted by Asterisk to the device. E.g: in a phone connected to a FXS device this would control the audio that is heard in the handset. This can be used to raise or lower the outgoing volume to compensate for hardware differences. Takes the same type of argument as rxgain. Default: 0.0
txgain:调整传出强度值，这是指由Asterisk发送给连接到FXS上的电话等设备的音频，Asterisk可以控制音频音量传送给手持设备端收听。这用于提高或降低外呼音量从而降低设备损耗。使用方法参数雷同fxgain，缺省值为0.0
txgain=-10.2
See: Asterisk zapata gain adjustment
Call Logging Options
Asterisk normally generates Call Detail Records (CDR), being a log or database of the calls made through Asterisk. This data can be used for Automated Machine Accounting (AMA). See Asterisk Billing.Asterisk通常会产生详单记录，记录是由Asterisk呼叫产生的，以日志或数据库存储。通话详单记录可以用作自动记账AMA。
accountcode: Sets the data for the “account code” field in the CDR for calls placed from this channel. The account code may be any alphanumeric string. It may be overridden at call time with the Asterisk cmd SetAccount|SetAccount command.
accountcode：设置通话详单中account code字段的数据，用于通道呼叫处理。计费代码可以是数字和文字字符串，可能在呼叫时被Asterisk命令setaccount重置。
accountcode=spencer145
amaflags: Sets the AMA flags, affecting the categorization of entries in the call detail records. Possible values are:
amaflags：设置AMA自动记账标记，影响通话详单中的分类条目。
- default: Let the CDR system use its default value. （CDR采用缺省值）
- omit: Do not record calls. （不记录）
- billing: Mark the entry for billing （产生记账条目）
- documentation: Mark the entry for documentation. （标记条目文档）
Timing Parameters （定时参数）
These keywords are used only with (non-PRI) T1 lines. All values are in milliseconds. These do not need to be set in most configurations, as the defaults work with most hardware. It has been noted that the common Adtran Atlas uses long winks of about 300 milliseconds, and channels from them should be configured accordingly.这个关键字仅用于T1线路，不包含pri。
prewink: Sets the pre-wink timing.
preflash: Sets the pre-flash timing.
wink: Sets the wink timing.
rxwink: Sets the receive wink timing.
rxflash: Sets the receive flash timing.
flash: Sets the flash timing.
start: Sets the start timing.
debounce: Sets the debounce timing. “The debounce settings in the Asterisk configuration affects how Asterisk
handles hookswitch transitions on its FXO/FXS interfaces.” — Derek Bruce
rxwink=300
prewink=20~~
Other Features
mailbox: If this option is defined for a channel, then when the handset is lifted, Asterisk will check the voicemail mailbox(es) specified here for new (unheard) messages. If there are any unheard messages in any of the mailboxes, Asterisk will use a stutter dialtone rather than the ordinary dialtone. On supported hardware, the message waiting light will also be activated — this probably requires that you also set adsi=yes. Update: This option does NOT require ADSI. It will send a standard FSK tone down the line that lights up the MWI on any capable analog phone.mailbox：这个选项为通道定义的。当摘机时，Asterisk会检测语音邮箱中未读的邮件。如果有未读邮件，Asterisk会有摘机警告音而不是通常的拨号音。在支持的硬件上，等待消息同样激活，这需要设置adsi=yes。这个选项不需要ADSI支持，它会发送一个 标准的频移键控提示音（也称为移频调制和移频信号）来挂掉支持WMI（消息等待支持）的模拟线路。
The parameters to this option are one or more comma-separated mailbox numbers, as defined in voicemail.conf.
mailbox = 1234
mailbox = 1,2
For each mailbox, if the mailbox is in a context other than “default”, place the context after the mailbox number
separated by an at sign (@).
如果语音邮件不是在default而是在context，按照mailbox@context的格式
mailbox = 1234@office
mailbox = 12@office,34@home
group: Allows you to group together a number of channels so that the Dial command will treat the group as a single channel. When Dial tries to make a call on a Zap group, the Zap channel module will use the first available (i.e. non-busy) channel in the group for the call. Multiple group memberships may be specified with commas, and to signify no group membership, the portion after the equals sign may be omitted. Group numbers can range from 0 to 31. The default value is an empty string, i.e. no groups.
group：允许把多个通道组成一组，Dial命令拨号的时候把群组视为一个单一通道。当Dial试图在ZAP组上拨号时，Zap通道模块使用组中第一个可用通道。多群组关系可以通过逗号来定义，等号后面省略表示没有群组。群组范围从0-31，缺省值时空字符串，即没有群组。
group=1
group=2,3
group=
See more about Channels and Groups
language: Each channel has a default language code that affects which language version of prerecorded sounds Asterisk uses for this channel. See Setting up a Multi-Language Asterisk Installation. The default is an empty string.
language：每个通道有一个缺省的语言编码，这是由预先录制声音的语言版本来定义的
language=en
progzone: This defines the timing and frequencies for call progress detection, which are buried in the sources in asterisk/dsp.c. This is DIFFERENT than the call progress timing defined in zaptel/zonedata.c and in /etc/asterisk/indications.conf, and so far only options you can use (defined in dsp.c) are us, ca, br, cr and uk. (This was added sometime between 1.0.9 stable and 1.2 beta). Default is: us
progzone：该选项为呼叫处理检测（在asterisk/dsp.c源代码中）定义了时间和频率，这与在zaptel/zonedata.c和/etc/asterisk/indications.conf中的定时呼叫处理不同。到目前为止该参数只能是：us,ca,br,uk，缺省是us
Important Stuff
context: This specifies which context a call will start in. The context controls how Asterisk will handle the call. Contexts are defined in the Dialplan. Default: “default”.context：定义了呼叫开始的context，context控制Asterisk如何处理呼叫。Context在dialplan中定义，缺省为”default”
context=internal
channel: This keyword is unlike all the other keywords in this configuration file, because where all the other keywords merely specify settings to use, this keyword causes Asterisk to actually allocate a channel with the settings that have been specified earlier in the file.
channel:这个关键字与配置文件中的其他关键字不同。原因是其他关键字仅仅定义设置来使用，这个关键字可以使Asterisk把前面定义的设置分配到通道中。
The channel keyword defines one or more channels. Each channel definition will inherit all options stated ahead of it in this file. Channels maybe specified individually, separated by commas, or as a range separated by a hyphen. Allocating a channel will not “clear” the settings, so any channels defined later on in this file will inherit the options for this channel unless you override settings.
通道关键字定义一个或多个通道，每行通道定义都会继承前面所有的选项配置状态。通道可以通过逗号分离单独定义，或者用连接符连接一组，分配通道不会清空设置，所以任何在后面定义的通道都会继承前面的选项除非覆盖设置。
channel => 16
channel => 2,3
channel => 1-8
安装好 dahdi之后，make config就会生成默认的配置文件
/etc/dahdi/init.conf
init.d自动启动脚本的配置，一般不需要改，以前是放在/etc/sysconfig下面的
/etc/dahdi/modules
需要加载的modules，看你的板卡型号，把不需要的注释掉
我的是TE410P，使用 wct4xxp
# Digium TE205P/TE207P/TE210P/TE212P: PCI dual-port T1/E1/J1
# Digium TE405P/TE407P/TE410P/TE412P: PCI quad-port T1/E1/J1
# Digium TE220: PCI-Express dual-port T1/E1/J1
# Digium TE420: PCI-Express quad-port T1/E1/J1
wct4xxp
/etc/dahdi/system.conf
最重要的配置文件，里面参数很多，基本都有注释，很清楚的了。
下面是我的情况，一条E1连接到该卡#1端口上面，配置如下
# Span 1: TE4/0/1 “T4XXP (PCI) Card 0 Span 1” (MASTER) HDB3/CCS
span=1,1,0,ccs,hdb3
# termtype: te
bchan=1-15,17-31
dchan=16
echocanceller=mg2,1-15,17-31
# Global data
loadzone = no
defaultzone = no
请参照你的E1，看看有没有用crc4，我配置了半天发现我的E1上面是不能加crc4的，郁闷死，嘿嘿。defaultzone 据说中国设置为no。
另外echocanceller也很重要，这个是硬件的回音消除，TE410P就支持。
/etc/modprobe.d/blacklist
这个是modules 不加载的列表，默认是要先取消所有模块，然后再加载。不用改
/etc/modprobe.d/dahdi
这个是每个模块的特别设置，比如T1到E1的软跳线，4个端口，一个个跳
The driver accepts parameter t1e1override and decimal value between 0 and 15 wich corespond to binary from 0000 to 1111 where each bit corespond to a span. 0 is T1 and 1 is E1.
Decimal | Binary
0 | 0000
1 | 0001
2 | 0010
3 | 0011
4 | 0100
5 | 0101
6 | 0110
7 | 0111
8 | 1000
9 | 1001
10 | 1010
11 | 1011
12 | 1100
13 | 1101
14 | 1110
15 | 1111
But how to know which port is configured for T1 or E1? Take a look at the picture below.Binary 0 0 0 0 Spans 4 3 2 1
noburst 就是是否开启突发传输，默认noburst=1就是关闭burst
options wct4xxp t1e1override=15 debug=1 noburst=0
options dahdi debug=1
启动脚本：
/etc/init.d/dahdi
Asterisk 1.6的配置文件：chan_dahdi.conf
[channels]
language=en
context=default
switchtype=euroisdn
pridialplan=national
internationalprefix = +
nationalprefix = +86
localprefix = +8620
privateprefix = +8620xxxxxxxx
unknownprefix =
signalling=pri_cpe
usecallerid=yes
hidecallerid=no
callwaiting=yes
callwaitingcallerid=yes
threewaycalling=yes
transfer=yes
cancallforward=yes
echocancel=yes
rxgain=0.0
txgain=0.0
group=1
callgroup=1
pickupgroup=1
immediate=no
callprogress=no
callerid=asreceived
group=1
context=default
signalling=pri_cpe
channel => 1-15,17-31
Asterisk 安装笔记（4）- 从程序发起呼出的几种方法
如果没有sip或者iax客户端的话，只是写简单的程序来实现呼出，就没有固定的dialplan了。。。方法有如下几种：
Automated dial out
The Asterisk dial plan extensions.conf responds to someone calling an extension on a channel. If you want to initiate a call from an external application, there are several ways to do this.There are basically four ways to initiate outgoing calls in Asterisk
- Use .call files. A call file is a text file that when placed in the correct directory makes Asterisk make an outgoing call.
- Use the manager API to activate a call. See Asterisk manager dialout
- Use the Asterisk CLI originate command
- FollowMe command of Asterisk 1.4: Since this has the abitility to fork (create multiple calls) it could be ‘misused’ to initiate outgoing calls.
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