How to make asterisk system automically response SIP call?

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How to make asterisk system automically response SIP call?

Postby laiyuanwei » Sun Sep 16, 2012 1:57 am

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 :

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 tell 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.
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Joined: Sun Sep 16, 2012 1:55 am

Re: How to make asterisk system automically response SIP call?

Postby david55 » Sun Sep 16, 2012 4:37 am

This is not a general comment. It should be in a Support forum.

If you want to use AsteriskNow, you must use the GUI. If you want to follow the book, and use Asterisk directly, you should be on the Asterisk Support forum. In that case, you start with extensions.conf.sample, which has no warning about not modifying it (or from a completely empty extensions.conf).

It is very unusual to start with an extension s when using SIP phones. The main use of s is for analogue phones, which seize the line as soon as they go off hook.

(AsteriskNow's extensions.conf does include files that can be user modified, but I don't know enough about the GUIs to know what can and cannot be modified. I would have thought that the GUIs used with AsteriskNow would have supported some sort of simple IVR.)
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