What should my phone system look like?

I’m the IT specialist for our small business and I’ve been charged with upgrading our old phone system to newer technology. I know very little about phone systems. Basically we have one phone number and four lines and four phones. I’m almost certain that this is an analog setup. From what I can see, from the phone box where the phone lines are coming in, there are 4 Cat5 cables running into our patch bay, then in each office, there is a Cat5 cable running from the wall into an analog phone, split into two RJ11 ends, each going into the phone (one into lines 1 and 2, the other going into lines 3 and 4).

We’ve recently purchased 12 IP based phones (Linksys SPA942). What we want is 8 lines and these 12 phones set up (although I think the Linksys phones only support 4 lines).

I’m not quite sure what my initial set up with Asterisk needs to be. Do we need to get 4 more lines from our phone company to have the 8 lines we want, or do we just go down to 1 line from our phone company and let Asterisk split this into 8 lines?

I understand that for my Asterisk system, I’ll need to get an interface card. If we just do the one line and let Asterisk handle splitting this up into 8 lines, is it just a matter of getting the phone cord from the phone box and plugging it into a modem on the Asterisk box?

I’ve been looking at the free Asterisk: The Future of Telephony book, but the chapter about preparing really only talks about the actual Asterisk system, and not the way my phone system is set up.

All in all, I’m really just looking for the best and most economical way to get what we’re looking for. If we need to switch from our analog phone lines to a VOIP solution and let Asterisk handle it, then that’s fine. I’m not afraid of getting my hands dirty with setting up this Asterisk system, but I’m just not sure what needs to happen in order for me to begin.

One more thing. I’m looking at this card selector:


For the type I chose analog, just because that’s how our lines are set up right now. Then for FXS ports it says: FXS is an interface that connects to a station, such as an analog telephone.

My new phones are IP phones, not analog phones, so I wouldn’t need FXS ports right?

So I chose digital just to see what was different. I’ve noticed the T1 option.

We just got a T1 in last week, with no phone service paired with it. Would it be easier to cancel our current analog phone service and get our phone service through our T1 provider? Again, would I just need to get 1 line or 8 lines from our provider?

dear sir,

it seems that you would like to setup a pbx with 8 incoming lines and 12 internal extensions (each one being a ip phone).

you can choose digital (T1) or analog lines from your telco provider. digium has a wide range of T1 cards and FXO cards for digital and analog interface respectively.

T1 line provides a future growth path because it has 23 channels and you don’t need to upgrade your card if you later expand your incoming traffic volume. on the other hand, if you stick to 8 analog lines, you need at least a 8-port FXO card.

in your case, i don’t think you need any FXS ports. FXS are used usually in gateways connecting analog phones. your ip phones are connected by RJ45, not RJ11.

assuming that you go for the T1 option, you need an asterisk box with a T1 card installed and connected to your telco. then you need to define 12 sip accounts for extensions (ie the ip phone) and design the dial plan logics. having the same set of sip accounts setup, your ip phones should register automatically to asterisk as sip peers and share the same T1. controlled by your dial plan logics, incoming calls from the T1 would be routed to the corresponding ip phones. conversely, each ip phone can place outgoing calls via the T1 lines to your telco provider.


That sounds great. Will I only need one line from my T1 provider and let Asterisk handle splitting this up into 8 different lines, or will I need to get 8 lines from my T1 provider?

If you go with a T1 you will only have one connection that carries up to 23 channels (calls). If you go with analog you will need a FXO port for each line. You will still be limited to one inbound or outbound call on an analog line so you will need 8 lines if you want to handle 8 external calls. Internal calls are kept within the asterisk system.

Another thing you have not mentioned is fax machines. If you have them you will need FXS ports to connect them.

So let’s say that I have my T1 line. I’m assuming that I get my current phone number transferred to my T1 provider. Right now, there are two wires running from the phone box to another smaller box on the wall which turns it into a regular ethernet cable with an RJ45 end. This is plugged into one of the four available T1/E1 ports (labeled as P0 - P3) on the back of the T1 box. Then one of the two available ports labeled Ethernet on the back is plugged into my switch.

Would the T1 provider run another set of wires from the phone box into another smaller box, which would run into another one of the T1/E1 ports on the T1 box, then from the second Ethernet port on the back of the T1 box into one of these cards: store.digium.com/productview.php … e=1TE122BF

Then would I just run a cable from a regular ethernet port on my Asterisk box to my switch and plug my IP phones into the switch?

Let’s say that I have a fax machine. Would I need to get a digital card from Digium to interface with my T1 as well as an analog card for the fax machine?

I’m just completely lost on how this topology should look like.


the asterisk box, your ip phones and the fax machine are networked in your LAN.

to connect fax machine to asterisk, you need an ATA device (which has a LAN port and a FXS port). the FXS port looks like a RJ11 socket for you to plug in the fax machine.

in short, your scenario might be like this:

TELCO->T1 line->Asterisk box (with digium T1 card)–>LAN–>IP PHONEs

Fax machine --> ATA device --> LAN


I don’t know why you’d want a T1 interface. It just adds more cost for trunking. Since you already have sip phones, it would make more sense to me to go with a pure voip solution. For that you wouldn’t need any interface cards in the server. Seems to me that would be cheaper, and one less thing to screw with. Eight sip trunks should be cheaper than a T1 anyway. Depending on the codecs used, you only need a 512K internet connection and might get by with 256k. Eight sip trunks can be had for as little as about $80 per month I think. I’m pretty sure an 8 channel T1 would be more than that. Any time you can get enough internet bandwidth I think going any other route than pure voip is a waste.