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Re: [LUG] Own Asterisk Server

 

Hello can I setup FreePBX in a virtual machine and use it to setup a group
of ciso phones for both internal calls and external via sip to other
people??? I presume pstn cards are needed to connect to phone network or can
I configure the phones to connect using sip to the phone network? Very new
to this pbx thing. Thanks ed.

-----Original Message-----
From: list-bounces@xxxxxxxxxxxxx [mailto:list-bounces@xxxxxxxxxxxxx] On
Behalf Of Neil Stone
Sent: 02 August 2009 09:18
To: list@xxxxxxxxxxxxx
Subject: Re: [LUG] Own Asterisk Server

Gordon Henderson wrote:
> On Sat, 1 Aug 2009, Edwin Rhodes wrote:
>
>   
>> Hello I am not sure if you are the right person, I am interested in
making
>> my own asterix server please advise, thanks ed.
>>     
>
> Edwin,
>
> I'm guessing this is aimed at me, so I've started a new thread for this to

> separate it from the existing ones. Always a good thing to do with a new 
> subject - i.e. start a new email rather than reply.
>
> (Although there are others there who use asterisk too - Dave Walker's also

> a guru ;-)
>
> Disclaimer: I build and sell asterisk boxes and provide ITSP services.
>
> You need something to run it on - and remember that you might need/want to

> run it 24/7 to pickup voicemail, etc. when you're not about.
>
> You also need some phones, although you can use soft-phones on a PC 
> (Asterisk itself has a soft-phone built in, but I don't recommend using
it)
>
> And something to connect to to make/take calls - A PSTN card to use your 
> BT connections, or a SIP (or IAX) connection to an ITSP of some sorts.
>
> Then there's the asterisk itself - many Linux distributions come with a 
> bundled version, but personally I favour compiling it from scratch myself.
>
> Asterisk itself comes in 3 basic versions - the 1.2 version, 1.4 and 1.6, 
> and to further complicate matters, the 1.6 is currently split into 2 
> branches. 1.2 is considered obsolete, but there are literally thousands 
> upon thousands of 1.2 installations out there. My own systems are based on

> 1.2, but I'm working on 1.4. I consider 1.6 too buggy for production use, 
> and even 1.4 has bugs that 1.2 doesn't have, but right now it has features

> I want that 1.2 doesn't have, so I'm migrating to it.
>
> I'd suggest going for 1.4 if compiling from scratch.
>
> Another way might be to dedicate an old PC and use one of the pre-canned 
> packages. pbxinaflash, trixbox or astlinux - there are others. These are 
> basically whole Linux distributions (centOs I think) which come with a pre

> built asterisk, web front-end, etc.
>
> I looked at these, then started from scratch.
>
> If you want to play on your workstation, then get the sources for the 
> latest version of 1.4 and compile it up yourself. Use the default settings

> and off you go. You won't need anything else at this stage.
>
> The tricky part isn't compiling and installing, it's editing the dozens of

> config files and the magical thing that controls it all; The Dialplan. 
> (Also known as extensions.conf)
>
> The dialplan is actually a programming language. You write programs in it 
> which are executed by various triggers - usually by a phone dialling a 
> number which matches something in the dialplan.
>
> What I suggest you do now though is go and get a book - it's free and in 
> PDF format. It's Asterisk: The Future of Telephony.
>
> Google around for it, but try:
>
>    http://www.asterisk-voip.nl/wiki/images/7/7b/AsteriskTFOT.zip
>
> The book is quite old now and will talk about Zap or Zaptel devices. These

> are 1.2 specific stuff - for a pure SIP/VoIP system you don't need any of 
> it, and Zap has been renamed DAHDI in the latest version of asterisk too.
>
> Enjoy!
>
> Gordon
>
>   
Well put, however I can recommend the use of either AsteriskNOW or 
FreePBX, the latter being the GUI that a number of PBXs are using. 
Personally I use Debian Lenny with FreePBX ontop.. I even wrote a script 
to deploy the whole thing if you're interested..!

TTFN

Neil

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