[Asterisk-Users] Hardware for PBX with 4 incoming/outgoing linesand 20 phones

[Andrew Thompson]

Andrew Elchuk wrote:
> Hi
> 
> I am interested in setting up an Asterisk PBX in my office with digium
> hardware, and I just have a few questions in regards to what I would
> need.  It is my understanding that an FXO card is used to interface
> with an incoming/outgoing phone line, and an FXS card is used for
> interfacing with a phone within the system.  Currently we have 4
> incoming/outgoing phone lines and would like to have 20 phones in the
> system.  In order to accomodate this, would you either reccommend
> having 1 TDM04B (4 FXO modules on it) for the 4 incoming/outgoing
> lines, and 5 TDM40B (4 FXS modules on each) for the 20 phones we
> would have in the system.  Or would you reccommend 1 TDM04B for the 4
> incoming/outgoing lines, and a T100P connected to a channel bank of
> some sort to connect to the internal phones?  If you reccommend the
> T100P and channel bank, where do you suggest I get an FXS channel
> bank?  Please let me know if I got any of this mixed up (like if I
> got the FXO and FXS cards mixed up) and thank you in advance for your
> help in us deciding the hardware we need for a new PBX.

Will your users be happy with single line analog phones at their desks?

If you go the channel bank route, you'll need a channel for each phone.

If you use the T-1 card, you'll have 24 analog channels total. You could set
up 4 to plug into analog lines from your carrier, and the rest to analog
phones in your office. The catch here is, you've then maxed out your (T1)
channel bank.

(I have not used any digium hardware, so I can't speak to the
effectiveness/ease-of-setup of a particular configuration.)

> 
> Andrew Elchuk
> 
> P.S.  We are currently using an X-Like software phone with a free
> world dialup account for communication with our other office in a
> different city.  My question is if I can configure the
> extensions.conf to connect to a free world dialup number when
> executing a dial command, or would I need to edit sip.conf as well or
> some other configs or is that even possible?  Thank you again.

If you have a static IP connection to the net, you can get by without using
FWD.

Using FWD as an intermediary or not, you would be able to configure standard
extensions or phone numbers inside your office that when called pipe out
over the net to your other office. If you spend enough time on it, you could
map extensions(as extensions or whole phone numbers) into asterisk on both
sides so that when anyone dialed a number for the other office asterisk
would transparently route it directly to the intended person on the other
side.

-----
Andrew Thompson
http://aktzero.com/