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Re: [LUG] Own Asterisk Server

 

A short howto would be great!!! Thanks ed.

-----Original Message-----
From: list-bounces@xxxxxxxxxxxxx [mailto:list-bounces@xxxxxxxxxxxxx] On
Behalf Of Neil Tancock
Sent: 03 August 2009 21:55
To: list@xxxxxxxxxxxxx
Subject: Re: [LUG] Own Asterisk Server

Hi guys, with reference to the Cisco phones, I had to re-flash a few of
these for a client who wanted to move from their Cisco PBX to asterisk.
You?re supposed to purchase the SIP firmware but there are a few places
where you can still get it for free.  Setting up the phones involves
re-flashing them, compiling a boot image for each type of phone and setting
up your DHCP server so it tells the phone the right place to get the image.
(It's called option 66, helpfully).  It sounds complex but once you've done
it a couple of times it gets easier.

I made some notes on it as I went through it, contact me if you need some
help.

Many thanks,

-Neil
Neil Tancock
Safeharbour Technology
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-----Original Message-----
From: list-bounces@xxxxxxxxxxxxx [mailto:list-bounces@xxxxxxxxxxxxx] On
Behalf Of Gordon Henderson
Sent: 02 August 2009 18:15
To: list@xxxxxxxxxxxxx
Subject: Re: [LUG] Own Asterisk Server

On Sun, 2 Aug 2009, Edwin Rhodes wrote:

> 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.

Well... The easy answer is yes... The long answer invovles sucking of air 
through teeth and ...

Cisco phones support Ciscos own protocol, which isn't SIP. Asterisk 
supports this protocol too, but from what I've heard, not very well.

Some Cisco phones can be re-flashed with SIP firmware, and I'm told that 
this is the prefered way forward.

Personally, I'd not run asterisk in a virtual machine, but others have 
done so.

You can make external SIP calls to other people if you register with them 
(or if they accept anonymous SIP calls) If you want to call "normal" 
numbers via SIP, then you need an account with an ITSP (Internet Telephony 
Service Provider) - basically a company that will do the "plumbing" to 
connect VoIP to the PSTN, and bill you for it ... :)

There are a great many of these ITSPs - Some even provide you with free 
calls to various destinations. Some provide personal support, others none. 
You get what you pay for.

If you need to talk to existing PSTN connections at your own location, 
then you need an interface card of some sorts - POTS (i.e. ordinary 
analogue phone lines), ISDN2 or ISDN30 are all avalable - for a price. 
Again, not sure I'd ever recomend it in a virtual machine - it depends on 
how good the VM is at letting an instance access underlying hardware and 
take interrupts from it.

I'd also suggest that if this is a PBX for a business, to do it properly - 
dedicate a server to it, because if the VM server goes down, so does your 
phones. You also need a business quality ISP with a good rotuer, and I'd 
strongly suggest an ITSP with a business level of support.

But if this is just for playing at home, then anything will do. No 
guarantees though :)

Gordon

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FAQ: http://www.dcglug.org.uk/linux_adm/list-faq.html