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Asterisk DoorEntry project need help with extensions.conf.

 
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davro
planetFreshman


Joined: 15 Mar 2005
Posts: 1
Location: England, London

PostPosted: Tue Mar 15, 2005 3:42 pm    Post subject: Asterisk DoorEntry project need help with extensions.conf. Reply with quote

Hi all,
Currently playing with asterisk, on a home door entry project.
Found this forum from a www.voip-info.org, the project so far ...

VIA MiniITX EPIA 5000 fanless
2.5 20gig Hardisk
512 MB Memory
55 watt power supply fanless

OS: Linux Ubuntu Debian based http://www.ubuntulinux.org/
Asterisk: Downloaded and Compiled VIA would aspire to be a i686, but it is actually a i586 from a compiler standpoint.

Asterisk *CLI seems to be working on the command line.
Code:

$ sudo asterisk
$ sudo asterisk -r

Asterisk CVS-HEAD-02/24/05-00:56:47, Copyright (C) 1999 - 2005 Digium.
Written by Mark Spencer <markster@digium.com>
=========================================================================
Connected to Asterisk CVS-HEAD-02/24/05-00:56:47 currently running on generis (pid = 3900)
generis*CLI>


Currently im reading up on howto configure extensions.conf sip.conf

Project requirements for this door entry project are quite simple at the moment.
1 DoorPhone can call any LocalSIPPhone.
2 LocalSIPPhone can call any DoorPhones and command a lock release with peers.

All Phones should talk to the asterisk server, this is where i am a little lost, on howto it all up. has anyone else done or doing anything simular.

Davro
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planetWayne
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Joined: 30 Jan 2003
Posts: 280

PostPosted: Wed Mar 16, 2005 10:06 pm    Post subject: Reply with quote

You'll have to bare with me on this...

Theres two files you need to look at with * in your case. Theres SIP.CONF which holds all your phone details and, as you suggested the EXTENTIONS.CONF that kinda holds it all together.

You'll then need to tell your sip phones to register with * (theres going to be some config on them I guess that tells it what its sip server and proxy are)

There are quite a few examples in Sip.Conf to have a look at - If you tweak that and have installed the demo config with * you should be able to get calls to the * demo (dial 500) - as long as you have an internet connection for your * box.

Once you have your phones registered in Sip.Conf you can start playing with the Extentions.Conf to get things happening when you dial numbers.

Take a look - if you get stuck - drop another line Smile

Wayne.
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