dahdi

# lspci –vvvv
Network controller

# yum install kernel-devel
reboot

# cd /usr/src/dahdi-linux-complete-XX
# cd linux
# make
# make install
# cd tools
# ./configure
# make
# make install
# make config

最后會(huì)提示
DAHDI has been configured.
List of detected DAHDI devices:
pci:0000:02:09.0     wctdm-       e159:0001 Wildcard TDM400P REV I
run ‘dahdi_genconf modules’ to load support for only
the DAHDI hardware installed in this system.  By
default support for all DAHDI hardware is loaded at
DAHDI start.
make[1]: Leaving directory `/root/install/dahdi-linux-complete-2.4.1.2+2.4.1/tools’
注意這里的wctdm-表示驅(qū)動(dòng)尚未生效。


# cd ../asterisk-1.8.0 # ./configure # make # make install # make config

# modprobe dahdi
# modprobe wctdm opermode=CHINA  (模擬卡)
# modprobe wct4xxp  (4e1)
# modprobe wcte11xp (D110P/D110E)
# modprobe opvxd115 (DE110P_DE110E)



讀opermode確認(rèn)參量已經(jīng)被加載了
cat /sys/module/wctdm/parameters/opermode
如果用opvxa1200, 請(qǐng)按下面的命令執(zhí)行:
cat /sys/module/opvxa1200/parameters/opermode
加載了驅(qū)動(dòng)之后,運(yùn)行dmesg命令去檢查這個(gè)mode.

opermode修改
具體步驟如下：
運(yùn)行：
1）首先停止asterisk
asterisk -r
> stop now

2）然后退出asterisk的CLI命令行進(jìn)入Linux Shell命令行， 依次運(yùn)行下列命令
service dahdi stop
modprobe dahdi
modprobe wctdm opermode=CHINA
dahdi_cfg -vvvv

3）運(yùn)行下列命令： 確定opermode已經(jīng)修改成功
cat /sys/module/wctdm/parameters/opermode
如果結(jié)果顯示CHINA表示成功

4）重啟asterisk
asterisk -vvvvvvvvgc

以A1200P和中國模式為例
在文件/etc/modprobe.d/dahdi.conf中添加一行：
options opvxa1200 opermode=CHINA
然后重啟系統(tǒng)就可以了。

# dahdi_genconf
/usr/sbin/dahdi_genconf: Failed to open /etc/asterisk/dahdi-channels.conf: No such file or directory
如果有提示有個(gè)asterisk的一個(gè)文件沒有找到?jīng)]有關(guān)系的。

# echo "#include dahdi-channels.conf" >> /etc/asterisk/chan_dahdi.conf

system.conf
loadzone = cn
defaultzone = cn
/etc/asterisk/indications.conf
country=cn

# dahdi_cfg –vvvvvv
localhost*CLI> dahdi show channels
dahdi_hardware
pci:0000:02:09.0     wctdm+       e159:0001 Wildcard TDM400P REV I
可以看到+號(hào)，表示驅(qū)動(dòng)已經(jīng)生效。
這是時(shí)候可以看到卡上的4個(gè)綠燈長亮。

 

ls /dev/dahdi/
1  2  3  4  channel  ctl  pseudo  timer  transcode
可以看到設(shè)備文件已經(jīng)創(chuàng)建

cat /proc/dahdi/1
Span 1: WCTDM/4 “Wildcard TDM400P REV I Board 5″ (MASTER)
1 WCTDM/4/0
2 WCTDM/4/1 FXSKS RED
3 WCTDM/4/2 FXSKS RED
4 WCTDM/4/3 FXSKS RED
這里可以看到卡的插線狀態(tài) S口因?yàn)闆]有接計(jì)算機(jī)電源因此沒有任何狀態(tài)。
3個(gè)O口 的RED表示入局電話線沒有信號(hào)（沒有插電話線）。

lspci
02:09.0 Network controller: Tiger Jet Network Inc. Tiger3XX Modem/ISDN interface
可以看到TDM400P卡的中斷號(hào)碼

PRI

prilocaldialplan
pridialplan

1. ; unknown: Unknown
2. ; private: Private ISDN
3. ; local: Local ISDN
4. ; national: National ISDN
5. ; international: International ISDN
6. ; dynamic: Dynamically selects the appropriate dialplan
7. ; redundant: Same as dynamic, except that the underlying number is not
8. ; changed (not common)

pridialplan: Sets an option required for some (rare) switches that require a dialplan parameter to be passed. This option is ignored by most PRI switches. It may be necessary on a few pieces of hardware. Valid options are: unknown, local, private, national, and international. This option can almost always be left unchanged from the default. Default: national.
pridialplan=local 
pridialplan設(shè)置為某些需預(yù)先傳入?yún)?shù)的交換設(shè)備設(shè)定預(yù)先傳入?yún)?shù)選項(xiàng)。大多數(shù)pri交換機(jī)會(huì)忽略這個(gè)選項(xiàng)，很少數(shù)交換機(jī)是需要的。正確的選項(xiàng)值為，unknown,local,private,national以及international。

ISDN PRI Switch Configuration
If you have a PRI line, specify your type of switch here. (PRI is a type of ISDN typically used to connect a PBX to a telephone exchange. They have multiple channels on the one connection.)
如果有Pri線路，需要定義交換類型。（pri是ISDN的一種類型，用于連接交換設(shè)備，有多種通道類型）

switchtype: Sets the type of PRI switch being used. Default: national. Acceptable values are:

national: National ISDN type2 (common in the US)
ni1: National ISDN type 1
dms100: Nortel DMS100
4ess: AT&T 4ESS
5ess: Lucent 5ESS
euroisdn: EuroISDN
qsig: Minimalistic protocol to build a "network" with two or more PBX of different vendors!

lsmod | grep dahdi

driver should be 'wctdm' but is actually 'netjet'

echo "blacklist netjet" >> /etc/modprobe.d/dahdi.blacklist.conf
reboot
modprobe
wcb4xxp

Zaptel DAHDI 說明
ztcfg -vv  dahdi_cfg -vv  分析system.conf，配置語音卡參數(shù)
zttool  dahdi_tool  語音卡檢測(cè)、測(cè)試
genzaptelconf  dahdi_genconf  生成Asterisk配置文件/etc/dahdi/system.conf 
ztmonitor   dahdi_monitor 監(jiān)控錄音某個(gè)通道
ztscan  dahdi_scan 掃描通道狀態(tài)
ztspeed  dahdi_speed 測(cè)試CPU占用率
zttest  dahdi_test 中斷穩(wěn)定性測(cè)試
ztloop  dahdi_loop 自環(huán)測(cè)試E1數(shù)字中繼語音卡
dahdi_hardware，顯示檢測(cè)到的DAHDI 硬件列表。

Zaptel 文件名 DAHDI 文件名 說明
/etc/zaptel.conf  /etc/dahdi/system.conf  主要的配置文件 
/etc/sysconfig/zaptel  /etc/dahdi/modules, /etc/dahdi/init.conf  開機(jī)啟動(dòng)參數(shù)設(shè)置 
/etc/init.d/zaptel  /etc/init.d/dahdi  開機(jī)自動(dòng)載入 
/etc/asterisk/zapata.conf  /etc/asterisk/chan_dahdi.conf  Asterisk 配置文件 

檢查模擬語音卡的IRQ中斷
cat /proc/interrupts

85,如何用fxstest測(cè)試DAHDI FXS模塊

硬件環(huán)境：A800P（2FXS+1FXO），A400P（2FXS+1FXO），步步高6082G來電顯示有繩電話

軟件環(huán)境：Centos5.5，dahdi-linux-complete-2.3.0.1+2.3.0

編譯安裝：

1.輸入以下命令：

#cd /usr/src/dahdi-linux-complete-2.3.0.1+2.3.0/tools 
#make menuselect

選擇“fxstest”，然后選擇“Save&Exit”保存退出。

2. 開始編譯dahdi。輸入以下命令：

#cd /usr/src/dahdi-linux-complete-2.3.0.1+2.3.0
#make
#make install
#make config

 3. 重新加載驅(qū)動(dòng)。

#service dahdi stop
#modprobe –r wctdm
#modprobe –r opvxa1200
#modprobe –r dahdi
#modprobe dahdi
#modprobe opvxa1200 opermode=CHINA
#modprobe wctdm opermode=CHINA
#dahdi_genconf
#dahdi_cfg –vvv （如果沒有報(bào)錯(cuò)，則表示已經(jīng)OK）
#dahdi_scan （此命令可以看到語音卡的基本信息及通道信息）

開始測(cè)試：

fxstest命令有兩個(gè)固定參數(shù)。第一個(gè)需要指定操作的設(shè)備文件，如/dev/dahdi/1。第二個(gè)參數(shù)則是需要操作的命令。測(cè)試時(shí)，F(xiàn)XS口需要接上電話機(jī)，不能啟動(dòng)asterisk。

1. 查看電壓伏特?cái)?shù)

# fxstest /dev/dahdi/1 stats //A800P第一個(gè)通道的電壓
TIP: -5.6400 Volts
RING: -54.1440 Volts
VBAT: -62.7920 Volts

#fxstest /dev/dahdi/11 stats //A400P第三個(gè)通道的電壓
TIP: -5.6400 Volts
RING: -54.8960 Volts

VBAT: -63.9200 Volts

2. 查看內(nèi)核寄存器

# fxstest /dev/dahdi/1 regdump //A800P第一個(gè)通道的寄存器
Direct registers: 
0. 05 1. 28 2. 18 3. 00 4. 18 5. 00 6. 00 7. 00 
8. 00 9. 00 10. 08 11. 33 12. 80 13. 10 14. 00 15. 00 
…

Indirect registers: 
0. 55c2 1. 51e6 2. 4b85 3. 4937 4. 3333 5. 0202 
6. 0202 7. 0198 8. 0198 9. 0611 10. 0202 11. 00e5 
…

3. 播放一串音調(diào)。

# fxstest /dev/dahdi/1 tones 

這時(shí)拿起電話，可以聽到一串音調(diào)。按CTRL+C中止。

4. 傳送dtmfcid。

#fxstest /dev/dahdi/1 dtmfcid 
Going to send a set of DTMF tones >A5551212C<
Phone is ringing
Ringing Done

電話振鈴時(shí)，可以看到來電顯示傳送的cid：5551212

5. 測(cè)試極性反轉(zhuǎn)。（此項(xiàng)測(cè)試必須要在傳送dtmfcid之后，才能正常顯示反轉(zhuǎn)信息）

# fxstest /dev/dahdi/1 polarity
 Twiddling polarity...
 Polarity is forward...
 Polarity is reversed...
 Polarity is forward...

6. 發(fā)送一串dtmf信號(hào)（“-”表示沒有發(fā)送信號(hào)）。

# fxstest /dev/dahdi/1 dtmf "12324" 70
Going to send a set of DTMF tones >12324<
Using a duration of 70 mS per tone

這時(shí)可以在電話來電顯示上看到發(fā)送的字符串“12324”，注意：時(shí)間要在70ms或以上才能顯示完整的字符串。

7. 觸發(fā)語音信箱等待指示燈、HV、NEON。（A800P不支持此項(xiàng)測(cè)試）

# fxstest /dev/dahdi/11 vmwi
Twiddling vmwi ...
Set 1 Voice Message...
Set 2 Voice Messages...
Set No Voice messages...

# fxstest /dev/dahdi/11 hvdc
Twiddling hvdc ...
Set 1 Voice Message...
Set 2 Voice Messages...
Set No Voice messages...

# fxstest /dev/dahdi/11 neon
Twiddling neon ...
Set 1 Voice Message...
Set 2 Voice Messages...
Set No Voice messages...

 -------------------------------------------------------------------

Zap Channel Module ConfigurationThe 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模塊允許Asterisk與zaptel驅(qū)動(dòng)程序之間通訊。通過配置zapata.conf文件實(shí)現(xiàn)

You will need the Zaptel kernel module device driver installed. See:

• Zaptel Installation

Although TDMoE is not directly related to Zapata hardware, it uses a pseudo-TDM engine, and gets configured here.

Using MySQL For Zap Channel ConfigurationIt 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而配置存儲(chǔ)在mysql表中，而不是zapatap.conf中，這需要版本支持

• Configuring Asterisk from a Database

The rest of this page assumes you are editing the zapata.conf file by hand.

Creating ChannelsThe 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，沒有看上去那么復(fù)雜，大多數(shù)關(guān)鍵詞自己不做什么，僅僅是定義通道參數(shù)，channel關(guān)鍵詞才是真正的創(chuàng)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義echocancel=no，會(huì)使影響所有通道，直到定義echocancel=yes，他會(huì)往下執(zhí)行，因此，定義必須在channel=>前面進(jìn)行定義

Available Settings
Signalling TypeThe 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.
信令類型是唯一強(qiáng)制設(shè)置，在分配一個(gè)通道之前，必須定義信令類型。如果連接的模擬電話設(shè)備，注意模擬信令是導(dǎo)致混亂的來源。FXS通道采用FXO信令，反之，Asterisk通知內(nèi)部設(shè)備采用相反方式。FXO接口卡采用FXS信令，同樣須定義。

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:設(shè)置通道信令類型，這些參數(shù)須與zaptel驅(qū)動(dòng)配置匹配。設(shè)置基于采用什么樣的板卡，如果通道信令設(shè)置錯(cuò)誤，如果配置描述與zaptel驅(qū)動(dòng)配置不匹配，或者如果卡不存在而配置正確，Asterisk不會(huì)工作。正確的設(shè)置通常包含下面4中信令中一種，fxs_ks, fxo_ks, pri_cpe or pri_net。該設(shè)置沒有缺省值，必須在分配通道前設(shè)置信令值，下面是Asterisk支持的信令類型。

• 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通道存在一個(gè)空閑Extension和一個(gè)微小閑置數(shù)字，只要閑置通道是空閑的，ZAP通道模塊就會(huì)嘗試在該通道上進(jìn)行空閑撥號(hào)，然后Asterisk就會(huì)執(zhí)行定義為idelext的Context和Extension中的命令。當(dāng)通道需要進(jìn)行語音呼叫時(shí)，'空閑'呼叫會(huì)斷開并讓多數(shù)通道有效。（當(dāng)然盡管有微小閑置呼叫正在運(yùn)行）。主要的用途是創(chuàng)建動(dòng)態(tài)的服務(wù)，當(dāng)閑置通道綁定了multilnk ppp協(xié)議后，將比傳統(tǒng)的多重映射提供更有效率的提供綜合的語音/數(shù)據(jù)服務(wù)。

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:最小可用閑置通道的數(shù)量。如果有很少的通道可用，Asterisk不會(huì)嘗試?yán)壢魏瓮ǖ肋M(jìn)行數(shù)據(jù)連接。該參數(shù)需要一個(gè)整數(shù)。
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:最小綁定進(jìn)行數(shù)據(jù)連接的通道數(shù)量，Asterisk會(huì)為數(shù)據(jù)開啟這個(gè)通道數(shù)量，而不是在需要的時(shí)候返回到語音通道的使用上。該參數(shù)需要一個(gè)整數(shù)。

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: 空閑撥號(hào)的數(shù)量，這是用于撥叫遠(yuǎn)程訪問服務(wù)器最基本的一個(gè)數(shù)字，為數(shù)據(jù)預(yù)留的閑置通道被這個(gè)分機(jī)。該參數(shù)需要一個(gè)整數(shù)，與在撥號(hào)方案中定義了idleext的分機(jī)不會(huì)產(chǎn)生沖突。

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：用于空閑分機(jī)的extension，以[url=]exten@context[/url]的用法使用，典型的用法是被作為分機(jī)運(yùn)行ZapRAS命令。
   minunused=2
   minidle=1
   idledial=6999
   idleext=6999@idle

Analog Trunk Features （模擬中繼特征）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:是否嘗試識(shí)別來電特殊鈴音，這不需要音頻分析，因?yàn)殁徱粼谀M線路上是非常簡(jiǎn)單轉(zhuǎn)換，只需要匹配轉(zhuǎn)換樣本。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：如果設(shè)置了usedistinctiveringdetection=yes，就需要定義三種不同特點(diǎn)的鈴音風(fēng)格，以便于Asterisk能夠嘗試識(shí)別。每種風(fēng)格使用逗號(hào)分割三個(gè)整數(shù)來定義。沒有文檔說明三個(gè)數(shù)字的含義，因此需要自己測(cè)試鑒別在不同國家不同公司中，不同數(shù)字代表的風(fēng)格。Asterisk控制臺(tái)上也會(huì)顯示識(shí)別的風(fēng)格數(shù)字，具體風(fēng)格可能會(huì)是以下一些情況。
    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:根據(jù)三種不同的鈴音風(fēng)格設(shè)置不同的context進(jìn)行來電跳轉(zhuǎn)，如果來電沒有定義的風(fēng)格匹配，就會(huì)進(jìn)入該通道缺省的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：忙音檢測(cè)，如果開啟，Asterisk會(huì)撥號(hào)嘗試或通話中分析在線的音頻，從而嘗試識(shí)別忙音信號(hào)。這非常在模擬中繼接口上外呼時(shí)檢測(cè)忙音信號(hào)非常有用，可以檢測(cè)何時(shí)掛機(jī)。確認(rèn)不能在例如QuadBri等卡上使用該參數(shù)，否則出現(xiàn)中斷通話的問題，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:這個(gè)選項(xiàng)需要busydetect=yes，可以定義等待掛機(jī)的忙音信號(hào)數(shù)量，缺省值是3，但能達(dá)到的最好效果可能是設(shè)置6或者8，數(shù)字越高，檢測(cè)掛機(jī)通道所需要的時(shí)間就越長，但小的數(shù)字可能會(huì)導(dǎo)致把其他聲音錯(cuò)誤的識(shí)別為忙音信號(hào)。

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 可以通過嘗試監(jiān)控呼叫狀態(tài)來偵聽振鈴音，忙音，擁塞音以及線路已經(jīng)應(yīng)答聲音特征。這個(gè)特征不受busydetect特征影響，兩者可以并行處理，獨(dú)自嘗試識(shí)別忙音信號(hào)。callprogress的特征是高實(shí)驗(yàn)證明更容易檢測(cè)錯(cuò)誤應(yīng)答，所以不要指望它非常準(zhǔn)確。因此，目前僅僅在標(biāo)準(zhǔn)美國電話鈴音中配置，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標(biāo)準(zhǔn)安裝中，沒有允許定義Zaptel卡使用脈沖撥號(hào)，盡管Zaptel驅(qū)動(dòng)支持脈沖撥號(hào)，但可以更新補(bǔ)丁文件，使用pulse關(guān)鍵字去開啟脈沖撥號(hào)。
    pulse=yes


Analog Handset Features 模擬電話特征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:如果手持設(shè)備支持ADSI（模擬顯示服務(wù)接口），設(shè)置set adsi=yes，ADSI類似來電顯示功能，傳遞編碼信息到手持設(shè)備。它可以在多行顯示的手持設(shè)備上創(chuàng)建交互式可視化菜單，通過文本接口提供類似語音郵件的訪問服務(wù)。

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設(shè)置為no，缺省值），當(dāng)FXS話機(jī)掛機(jī)時(shí)，Zaptel驅(qū)動(dòng)會(huì)馬上提供撥號(hào)音，等待撥號(hào)并傳遞給Asterisk。 Asterisk等到接收到extension匹配號(hào)碼時(shí)，就會(huì)開始執(zhí)行相應(yīng)的命令，如果設(shè)置 immediate=yes，Asterisk會(huì)命令 Zaptel驅(qū)動(dòng)不要在FXS掛機(jī)時(shí)產(chǎn)生撥號(hào)音，而是把控制權(quán)交還給Asterisk，Asterisk會(huì)開始執(zhí)行這個(gè)通道的s extension。這通常應(yīng)用于batphone 模式（蝙蝠電話？），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：如果開啟，在通話過程中如果有來電時(shí)，Asterisk就會(huì)產(chǎn)生呼叫等待提示音。如果通道缺省有呼叫等待，可以臨時(shí)摘機(jī)按鍵*70取消，這種情況下，會(huì)收到重播提示音去撥打希望撥打的號(hào)碼。沒有合適的方法臨時(shí)開啟缺省設(shè)置為關(guān)閉的通道的呼叫等待。缺省為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:設(shè)置在呼叫等待過程中是否傳送主叫號(hào)碼等數(shù)據(jù)，需要設(shè)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：（三方通話）如果設(shè)置開啟，可以在按保持鍵切換話路，讓原通話處于保持狀態(tài)，這時(shí)會(huì)收到重?fù)芴崾疽簦㈤_啟另外一方通話。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:（呼叫轉(zhuǎn)接）這個(gè)選項(xiàng)僅當(dāng)三方通話=yes時(shí)有效，當(dāng)設(shè)置了三方通話和呼叫轉(zhuǎn)接，一旦通過或呼叫保持按鍵把當(dāng)前話路置于保持狀態(tài)，就可以撥號(hào)呼叫另外分機(jī)，把2個(gè)話路橋接起來，然后掛機(jī)。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.
cancallforward：如果呼叫前轉(zhuǎn)啟用，可以通過撥號(hào)*72+想要轉(zhuǎn)向的Extension，立刻激活呼叫前轉(zhuǎn)（同時(shí)會(huì)有重?fù)芴崾疽簦@時(shí)如果有來話，那么呼叫會(huì)被重定向到設(shè)置的轉(zhuǎn)移號(hào)碼上，可以通過撥打*73取消呼叫前轉(zhuǎn)。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：如果開啟此設(shè)置，可以通過撥打*69讓Asterisk讀出最后呼入的主叫號(hào)碼，如果沒有記錄最后呼叫主叫號(hào)碼，將聽到重?fù)芴崾疽簦笔≈担簄o
    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:通道可以不屬于或者屬于多個(gè)呼叫群組。呼叫群組定義了當(dāng)電話振鈴時(shí)，誰可以接聽。當(dāng)一個(gè)通道振鈴時(shí)，其它那些pickupgroups中包含該通道 callgroups其中之一的通道可以通過按*8#來接聽電話。這個(gè)特性支持在ZAP，SIP。skinny和MGCP通道類型上使用，群組數(shù)字范圍為 0-31，，缺省值是空字符串，即沒有組。
   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：通道可以不屬于或者屬于多個(gè)摘機(jī)群組，摘機(jī)群組定義了可以應(yīng)答那些電話。如果其他通道振鈴，而本通道pickupgroup是振鈴?fù)ǖ纁allgroups群組其中之一，那么本通道可以通過按*8#來接聽振鈴?fù)ǖ馈Ｈ航M范圍為0-31，缺省值為空字符串，即沒有群組。
   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.
如果同時(shí)不止一路通道振鈴符合條件可以通過按鍵*8#接聽，只能接聽第一條振鈴?fù)ǖ溃床荒芸刂七x擇接聽哪一條。
   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：如果設(shè)置了這個(gè)選項(xiàng)（在ZAP通道上啟用來電轉(zhuǎn)接），可以轉(zhuǎn)接來電到另外一個(gè)電話，外部呼叫的主叫號(hào)碼同時(shí)跟隨。
    useincomingcalleridonzaptransfer=yes

Caller ID Optionscallerid: 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：當(dāng)來電時(shí)設(shè)置主叫ID字符串，傳送給接收者，通常為手持設(shè)備設(shè)置callerID。定義callerid：雙引號(hào)名字后緊跟角括號(hào)數(shù)字，對(duì)中繼線路，設(shè)置asreceived來傳送主叫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.
重要事項(xiàng)：CallerID只能在硬件支持的公共電話交換網(wǎng)上被傳輸，例如PRI。在模擬線路上設(shè)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:對(duì)于手持設(shè)備，這個(gè)選項(xiàng)可以在振鈴時(shí)讓Asterisk發(fā)送CallerID數(shù)據(jù)到到手持設(shè)備，對(duì)于中繼線路，該選項(xiàng)致使Asterisk查找來電主叫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隱藏（不能應(yīng)用于FXO中繼線路），對(duì)于PRI通道，在外呼時(shí)停止傳送主叫ID。對(duì)于FXS端外呼時(shí)，會(huì)停止發(fā)送主叫ID信息到被叫方。FXS端可以在話機(jī)上按*82（啟用）*67（關(guān)閉）可以控制是否傳送主叫ID傳送。
    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.
restrictcid: （僅用于PRI通道），該選項(xiàng)在hidecallerid=no時(shí)可以有效設(shè)置，如果hidecallerd=no并且restrictcid=yes，外呼時(shí)，asterisk會(huì)阻止以顯示號(hào)碼方式發(fā)送主叫id，但ANI消息數(shù)據(jù)仍然發(fā)送），缺省為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:（僅PRI通道有效）不管是否把callerid作為外呼的顯示號(hào)碼，交換機(jī)都會(huì)傳送。
    usecallingpres=no

Audio Quality Tuning Options （音頻質(zhì)量調(diào)整選項(xiàng)）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: 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.
relaxdtmf:如果DTMF檢測(cè)有問題，可以放寬DTMF檢測(cè)的參數(shù)。
    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: 開啟或關(guān)閉回音消除（缺省值：是），建議不要關(guān)閉該設(shè)置，可以定義回音消除yes（128濾波參數(shù)）或者no（0濾波），或者定義參數(shù)為 16,32,64,128,256中一個(gè)，每種濾波參數(shù)都是一種數(shù)據(jù)流樣本，在T1線路上會(huì)是每秒1/8000，因此濾波參數(shù)值等于2ms,4ms, 8ms,16ms,32ms尾長，如果設(shè)置的回音消除為不同的值，Asterisk將直接使用128而不會(huì)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:開啟或關(guān)閉在橋接的TDM呼叫中的回音消除，原則：TDM橋接呼叫不需要回音消除，但開啟這個(gè)選項(xiàng)通常可以提高語音效果。
    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：有時(shí)回音消除不能夠很快的自學(xué)習(xí)，通話開始時(shí)會(huì)有回音，然后很快消除。開啟回音訓(xùn)練可以讓Asterisk使通道暫時(shí)無聲而發(fā)送一個(gè)刺激信號(hào)，并根據(jù)響應(yīng)效果預(yù)訓(xùn)練回音消除，從而能夠更接近真實(shí)的回音。然而如果在電路up和終端響應(yīng)定位之間有相當(dāng)?shù)难訒r(shí)，某些典型中繼被會(huì)作為終端進(jìn)行連接，訓(xùn)練刺激信號(hào)會(huì)檢測(cè)open中繼的特性而不是實(shí)際電路。因此，在開始響應(yīng)刺激信號(hào)處理之前，可以在10ms和4000ms延時(shí)之間定義一個(gè)值，或者定義yes，缺省就是 400ms。默認(rèn)值沒有定義。
    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：調(diào)整接收獲取強(qiáng)度值，這是指Asterisk從例如連接到FXS通道上的電話設(shè)備上接收到的音頻，該選項(xiàng)能控制由電話發(fā)送給Asterisk的音頻，可以用于提高或降低進(jìn)入的聲音音量，從而補(bǔ)償硬件損耗。可以定義獲得值從-100db到100db，10db就意味著很高了。修改時(shí)應(yīng)進(jìn)行微調(diào)。
    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:調(diào)整傳出強(qiáng)度值，這是指由Asterisk發(fā)送給連接到FXS上的電話等設(shè)備的音頻，Asterisk可以控制音頻音量傳送給手持設(shè)備端收聽。這用于提高或降低外呼音量從而降低設(shè)備損耗。使用方法參數(shù)雷同fxgain，缺省值為0.0
    txgain=-10.2

See: Asterisk zapata gain adjustment

Call Logging OptionsAsterisk 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通常會(huì)產(chǎn)生詳單記錄，記錄是由Asterisk呼叫產(chǎn)生的，以日志或數(shù)據(jù)庫存儲(chǔ)。通話詳單記錄可以用作自動(dòng)記賬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：設(shè)置通話詳單中account code字段的數(shù)據(jù)，用于通道呼叫處理。計(jì)費(fèi)代碼可以是數(shù)字和文字字符串，可能在呼叫時(shí)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：設(shè)置AMA自動(dòng)記賬標(biāo)記，影響通話詳單中的分類條目。

• default: Let the CDR system use its default value. （CDR采用缺省值）
• omit: Do not record calls. （不記錄）
• billing: Mark the entry for billing （產(chǎn)生記賬條目）
• documentation: Mark the entry for documentation. （標(biāo)記條目文檔）

   amaflags=billing

Timing Parameters （定時(shí)參數(shù)）
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.
這個(gè)關(guān)鍵字僅用于T1線路，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 Featuresmailbox: 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：這個(gè)選項(xiàng)為通道定義的。當(dāng)摘機(jī)時(shí)， Asterisk會(huì)檢測(cè)語音郵箱中未讀的郵件。如果有未讀郵件，Asterisk會(huì)有摘機(jī)警告音而不是通常的撥號(hào)音。在支持的硬件上，等待消息同樣激活，這需要設(shè)置adsi=yes。這個(gè)選項(xiàng)不需要ADSI支持，它會(huì)發(fā)送一個(gè) 標(biāo)準(zhǔn)的頻移鍵控提示音（也稱為移頻調(diào)制和移頻信號(hào)）來掛掉支持WMI（消息等待支持）的模擬線路。

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 (@).
如果語音郵件不是在default而是在context，按照[url=]mailbox@context[/url]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：允許把多個(gè)通道組成一組，Dial命令撥號(hào)的時(shí)候把群組視為一個(gè)單一通道。當(dāng)Dial試圖在ZAP組上撥號(hào)時(shí)，Zap通道模塊使用組中第一個(gè)可用通道。多群組關(guān)系可以通過逗號(hào)來定義，等號(hào)后面省略表示沒有群組。群組范圍從0-31，缺省值時(shí)空字符串，即沒有群組。

    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：每個(gè)通道有一個(gè)缺省的語言編碼，這是由預(yù)先錄制聲音的語言版本來定義的
    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：該選項(xiàng)為呼叫處理檢測(cè)（在asterisk/dsp.c源代碼中）定義了時(shí)間和頻率，這與在zaptel/zonedata.c和 /etc/asterisk/indications.conf中的定時(shí)呼叫處理不同。到目前為止該參數(shù)只能是：us,ca,br,uk，缺省是us

Important Stuffcontext: 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：定義了呼叫開始的context，context控制Asterisk如何處理呼叫。Context在dialplan中定義，缺省為"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:這個(gè)關(guān)鍵字與配置文件中的其他關(guān)鍵字不同。原因是其他關(guān)鍵字僅僅定義設(shè)置來使用，這個(gè)關(guān)鍵字可以使Asterisk把前面定義的設(shè)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.
通道關(guān)鍵字定義一個(gè)或多個(gè)通道，每行通道定義都會(huì)繼承前面所有的選項(xiàng)配置狀態(tài)。通道可以通過逗號(hào)分離單獨(dú)定義，或者用連接符連接一組，分配通道不會(huì)清空設(shè)置，所以任何在后面定義的通道都會(huì)繼承前面的選項(xiàng)除非覆蓋設(shè)置。

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

Obsolete Settingsstripmsd: (Obsolete) Strip the 'Most Significant Digit,' the first digit or digits from all calls outbound on the given trunk channels. Takes as an argument the number of digits to strip. Use ${EXTEN:x} for this functionality.

posted on 2011-11-24 11:37 八葉草 閱讀(7322) 評(píng)論(2)  編輯 收藏 引用 所屬分類: asterisk