如何使星号服务器自动响应SIP呼叫? [英] How to make asterisk server automatically response to SIP call?

问题描述

我的目标:我想在星号服务器上使用softphone(3CX电话)注册，并调用服务器并进行星号操作

作为服务器来自动响应某些内容，例如播放歌曲.

我的工作方式:我使用virtualbox安装了asteriskNow，并通过为我的SIP设备设置扩展名来注册了软件电话

(扩展名333).我在 etc/asterisk/extensions.conf 中编写了一个拨号计划.拨号计划是:

[incoming]
exten =>s,1,Answer()
exten =>s,n,Playback(dir-intro-oper)
exten =>s,n,Hangup()

我想要服务器的任何来电，服务器将自动应答并播放预定义的语音(dir-intro-oper.gsm)

然后举手.

但是我遇到的问题是:

我使用网络电话，但我不知道应该拨哪个号码到星号服务器.我应该为

设置分机号吗

星号服务器本身?如果是这样，该怎么做?通过设置SIP卡车?将拨号计划写在sip.conf中吗?还是其他?

另一个问题: 我读过与星号相关的书星号，未来的电话"，该书告诉我们在extensions.conf

中编写Dialplan.

直接，但是我发现服务器中的extensions.conf提醒我们不要直接修改文件，必须使用web-gui

进行修改.那么我应该遵循哪种方式?

在这种情况下，我不使用任何其他硬件电话. 我是星号的新手，请给我一些提示和详细过程.

解决方案

您使用的"s"扩展名是特殊"的，当Asterisk不知道要做什么时，它将尝试使用该扩展名. >

如果您真的希望通过VoIP电话或ITSP拨打电话，以相同的方式进行处理，请尝试以下操作:

[incoming]
exten =>_X.,1,Answer()
same => n,Playback(dir-intro-oper)
same => n,Hangup()

...，并确保在您设置的SIP电话和SIP中继定义中:

context=incoming

这实际上会强制所有呼叫进入您的上下文，然后无论您拨打什么电话，您始终会匹配分机号码.

请访问 https://wiki.asterisk.org/wiki/display了解更多信息/AST/模式+匹配

My objective: I want to use softphone(3CX phone) register with asterisk server, and make call to the server and asterisk act

as a server to automatically response something, like play a song.

How i did: I installed asteriskNow using virtualbox, and registered the softphone by setting exntension for my SIP device

(extension 333). And i write a dialplan in etc/asterisk/extensions.conf. The dialplan is :

[incoming]
exten =>s,1,Answer()
exten =>s,n,Playback(dir-intro-oper)
exten =>s,n,Hangup()

I want any incoming call to server, the server will automatically answer, and play a pre-defined voice (dir-intro-oper.gsm )

then handup.

But I met the problem is:

I use softphone, and i dont know which number i should dial to the asterisk server. Should i set up a extension number for

asterisk server itself? If so, how to do that? By setting up SIP truck? Write the dialplan in sip.conf? or anything else?

Another questions: I read the asterisk related book"asterisk, the future telephony" which tells us to write dialplan in the extensions.conf

directly, but i found the extensions.conf in the server which alerts us do not modified the file directly, must use web-gui

to modify.So which way i should follow?

In this case, i do not use any other hardware phone. I am a novice on asterisk, please give me some hints and detail procedure.

解决方案

The "s" extension that you are using is a "special" that when Asterisk doesn't know what to do, it tries to use that.

If you really want any call to the box, either from a VoIP phone or an ITSP to get handled the same way, try this:

[incoming]
exten =>_X.,1,Answer()
same => n,Playback(dir-intro-oper)
same => n,Hangup()

... and make sure that in your SIP phone and SIP trunk definitions that you set:

context=incoming

That literally forces all calls into your context and then no matter what you dial, you always match the extension number.

More reading at https://wiki.asterisk.org/wiki/display/AST/Pattern+Matching

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