Mar 11 2009

Asterisk 安装笔记(2)- Zaptel 和 Dahdi 的配置

Category: 技术ssmax @ 13:30:14

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:

   channel => 1

   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.

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.

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.
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.

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.


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.
Default: no.

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。

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。

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


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.

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 = 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.

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

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.

callwaitingcallerid: Sets whether Asterisk will send Caller ID data to the handset during call waiting indication. Requires also setting callwaiting=yes. Default: no.

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.

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.


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.

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.

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. 

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.

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. 

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.

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=”Mark Spencer” <256 428-6000>

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.
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.

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.


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.

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.

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:

relaxdtmf: If you are having trouble with DTMF detection, you can relax the DTMF detection parameters. Relaxing them may make the DTMF detector more likely to have “talkoff” where DTMF is detected when it shouldn’t be. Default: no.

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.

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.

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.

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

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

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.

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。

amaflags: Sets the AMA flags, affecting the categorization of entries in the call detail records. Possible values are:

  • 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.
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


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 (@).

   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.


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.

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

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”.

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.

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.

   channel => 16
   channel => 2,3
   channel => 1-8

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