Using T1 for dial tone for FreePBX


I’ve been on and off looking at ways to break away from our internal phone system, but really don’t understand what I would need to do so.

Here is what we have/how it works and what I would like to do.

Right now our small school has a T1 Circuit providing dial tone for 6 phone numbers broken up as such:

1 main phone number for the building and a roll over number
1 main phone number for another department in the building and a roll over number
1 fax line for the school
1 fax line for the other department.

The “other” department is a cyber school division inside the same building.

I thought the best way to enter VOIP using some open source solution would be to take a small subset such as the cyber school department and setup FreePBX to handle their 5 staff members.

I’m having trouble understanding how I would be able to utilize their 2 phone numbers and 1 fax to “port” them over using FreePBX since we have a T1.

I was thinking about getting a FX0/FXS card and simply taking the twisted pairs for those lines and plugging them directly into the FXO/FXS card. Somehow in my mind that seems like it would be work, however even if it did I still question if that is the “right” way to do it.

Can someone offer some advice on how I would go about what I want to accomplish?

Eventually I would love to replace our phone system with a system that has more flexibility on management for Moves/Adds/Changes. I possibly would love the idea of tying in our other two schools to have a main number with a switchboard to route calls, but that is really a long term dream. I have often though of a cloud based solution since our other two buildings aren’t connected via fiber. Right now I do have a subnet setup with a site-to-site vpn between all 3 buildings. That subnet is used for our Domain Controllers to sync between each other.

Thank you!

I think it’s possible to achieve everything you want.

You can terminate your T1 on your FreePBX server via PCI card or using a PRI gateway (Sangoma, Digium, Adtran, etc…).
You can connect your faxes to FXO/FXS cards in your server or use a gateway for that too.

If you got sufficient bandwidth, you certainly can run calls between your buildings over the site-to-site VPN.

How is your T1 currently connected to your PBX?

What he said, but I think you need to start smaller and less “business critical”. Start out by setting up a phone system at the “remote” school. It sounds like they’re tech-savvy over there, so they would be good people to have on your side.

There are lots of questions left unanswered in your post. For example, do you know how the T1 is terminated? Choices are a dedicated PBX system (maybe an old Panasonic or Avaya) or a “channel bank”. The first is a full-up “forklift” upgrade - everything including the wiring needs to be replaced. If it’s a channel bank, you can connect an Asterisk server through a T1 card and terminate all of the calls in the server.

The amount of traffic over the network for most “phone” applications is pretty lightweight, so the existing network will probably be fine.

One fine-print wrinkle is that your T1 may be providing your phone service AND your Internet connectivity. If that’s the case, you will need to coordinate with your upstream, provider.

There are a thousand ways to set this up so it works well. Here’s one:

  1. Set up an Asterisk server and extensions that talk to the Asterisk server. You can use almost any scheme you’d like. I like 4 digit numbers that start with ‘4’. Do this is a room somewhere so you can demonstrate the system before you shut down your old phones.

  2. Convert your phone numbers to a VOIP provider. Most of the big carriers support that now. If you school’s Internet service is coming in with your phone service, you can move the phone channels to support your Internet connection and increase your Internet Access by 25%. Your incoming phone service will now be delivered over the T1 as Internet Traffic.

  3. Set up your Asterisk server to route incoming calls to your front office/attendant/whatever. They can then route the calls through either attended or unattended transfers.

As you get further along, you can implement other routes.

The only part of this that I’ve ever found tricky is the FAX part. I usually end up going to an “off campus” FAX-To-Email gateway for my incoming FAX services. If you choose well, you can also get outbound EMail-to-FAX services. My VOIP provider provides a service like this. There are plenty of places around that can help you make this happen.


I have to look again, but I believe there is an AdTran box that has the 6 voice lines terminated on a 66 block. I think the 6 lines go into the Phone system. The current phone system is a Samsung Office Serv 100 and there are two Amphenol connections inside that come out and are terminated on 66 blocks for the extensions. There is another set of 66 blocks that are punched down to the phones in the building via Cat5e. The two sets of 66 blocks are cross connected using typical twisted pairs, making the extension connections.

We have separate Internet for all of our data. It is provided by our local Internet company and is a cable modem connection.

Does this help? I would love to run the current phone system for the main school and take the applicable lines from the T1 to the Asterisk or FreePBX (whatever I decide) and use that as my experiment. The cyber school department is less critical than the main school for voice.

Sorry, I’m trying to get a better understanding of how this works. Do you recommend eventially ditching the T1 for something else that is more easily adapted to VOIP while still being able to satisfy our current digital phone system (Samsung Office Serv 100)?

Thank you!

The AdTran is acting as a channel bank - your T1 is split from 24 Channels into 6 discrete phone lines through that. Disconnecting that from the AdTran and plugging it into an Asterisk Server with a T1 termination card is doable.

If you really mean a 66-block and aren’t using the term as a generic punchdown block, then all of that stuff is useless in a new IP-Based phone system. It sounds like your Samsung is set up as a basic KSU with 6 incoming ports configured and a few digital phones attached. is the documentation for your phone system.

After that, the flexibility of the Samsung system generates a series of challenges. From the documentation, it sounds like the system automatically identifies what kinds of phones are connected to each outgoing channel and they are managed internally. This means (from the “I’m not there, so I have no way of ever knowing” perspective) that you could be running any number of possible types of phones.

To duplicate your current system, you can use an Asterisk server set up with an 8-port (they usually come in multiples of 4) FXO card to take your six incoming lines and convert them to VOIP. If you’d rather, you can use a T1 terminus card and connect it in front of the AdTran - makes no difference EXCEPT for the FAX part (which, as you may recall, I said can be tricky with VOIP).

So, now you have to figure out what kinds of handsets you need. You have lots of choices, but it basically boils down to what you have and matching it to something that will work for you.

In this area, there are three distinct types of phones:

  1. Digital phones that connect ONLY to your PBX - if I’m reading the docs right, the IDCS phones from the PDF are probably designed to only work with your PBX.

  2. Single- or double-line POTS phones - the system appears to be able to do these, and there are some cases where they make sense. From your description, it wouldn’t surprise me if you had a few of these. This includes things like FAX lines, door phones, and other single-use, single-user applications.

  3. VOIP phones - the system also appears to be able to communicate with VOIP phones. The upside is that you may be able to reuse the VOIP Phones in a new Asterisk based system. The downside is that they may have implemented a proprietary VOIP protocol. Won’t know until you get more information about the existing phones.

The fact that you’re using 66-blocks (and I assume you really mean 66-blocks) tells me that you probably are not using any of the VOIP capabilities, even though your wiring guy installed Cat-5 cable (same price as Cat-3 and he probably had it on hand, left over from another job).

If each instrument terminates on those punchdown blocks, it’s reasonable to assume that they are either digital phones (using two pairs each) or single lines (using a single pair). Knowing what you have connected to each pair (or set of pairs) will help you understand what your system is doing now.

In a perfect setup, you would identify each of the phones in the system and figure out what the suitable sub is for that phone. Of course, everyone has their preferences for specific instruments. I, for example, usually go for Cisco phones in SCCP mode. I can usually match up the functions needed from the old phones to a new phone and get you where you want to go. Others may suggest Sangoma’s new phone line (since FreePBX is owned by Sangoma now…).

Remember that you can also find and employ a 24-port FSX card and plug in up to 24 pots phones (or door phones…) in your Asterisk server (using a technology called DAHDI) and really duplicate what is there now (as well as reusing some of that 66-block wiring).

Your current system is complicated and will be hard to replicate, but only because this is your first foray into the technology. Since you are working between three campuses, you have to also take into account what equipment is installed there as well (we haven’t even scratched the surface yet). The advantage you have over me is that, based on your experience, you only have a hammer (SIP phones and trunks) so you can treat everything like a nail. :smile:

Something that I say about once a week that sometimes helps is that installing an Asterisk system changes the way the phones logically connect to the world. There are no more “dedicated” lines in or out - the Asterisk system handles everything from the incoming calls to the intercom. Asterisk becomes the phone and what used to be the phones are now just really cool handsets. I started telling people this after I installed a phone system for a doctor’s office. One of the nurses thought she needed a “dedicated line” just for her. She couldn’t understand that, with VOIP, there’s no such thing. The only way a number is “dedicated” is because we tell Asterisk where to send the call.

I hope that helps.

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I would add to @cynjut excellent post that DAHDI can also perform DAC’sing from one/some input DAHDI channel(s) to (an)other output DAHDI channel(s) (without FreePBX/asterisk being involved at all) .

If your final box needs to preserve faxing without worrying about VOIP from one channel to another extant FXS port with the fax machine attached. So a transparent drop in solution could be a FreePBX with two T1 interfaces and a couple of FXS ports between the Telco and the Adtrans, any channel could be either passed through to the legacy PBX or filtered through an extension to add FreePBX voicemail conferencing etc. Eventually the legacy box could disappear and if you do it baby step by baby step, no one would ever know until their desk phone changed one day.

Thanks Dicko.

Now that we’ve completely overwhelmed you, it’s important for you to understand that what you want to do is not just a really good idea, but appears to be the direction that all telephony is moving.

A couple of years ago, I bid on a project to replace the overhead speakers with a new VOIP-based system. There were no good ways to do overhead paging from a VOIP system, so I had to big in POTS (DAHDI) lines to make it work. Of course, it also meant that I was going to end up installing a phone server for them when they were sure all they wanted was a PA.

I got a catalog in the mail last week, and every piece of tech that I needed to bid that job as a VOIP system is ready-to-go. The stuff that I was going to have to invent is now ready and waiting to be bought. I don’t see that trend stopping or even slowing down.

We moved away from bus-bar systems to digital, and now we’re moving from dedicated phone systems to multiuse networking. It’s a trend I don’t see stopping.

WIth that said, I really do hope you start moving forward on your project. Little steps at first, then bigger one as you move to improve your schools communication infrastructure.


In my market, AT&T will no longer provision T1’s over copper or any OC hardware, they push you into IP FLEX (unfortunately a disappointing sub-set of both the old style T1 super-trunks/PRI’s and barely functional SIP provisioning). Not a great example of “multiuse networking” but they are AT&T so they get away with it :wink:

(IMHO overhead paging onsite is best/cheapest done through the asterisk console driver on “audio out” adding a 10k:120ohm balun if necessary into your current “PA/Tannoy” )

Yes, now I’m really overwhelmed! Network stuff is 2nd nature to me. Vlans, routing, vpns etc… no problem. Something as simple as phones make me bite my nails! I’m determined to move forward on this little project as it has been on my mind for a while. I’m going to take a picture of the dmarc where the termination of the T1 and phone system along with the punch down blocks (Yes I was referring to the punch downs as 66 blocks). The handsets are ds-5021 (or similar minus the number of buttons depending on their location).

I’m sure I’ll be posting more questions this week when I go back to work. Thanks for all the input so far. Your experience and knowledge is much appreciated!!

It is networking, just a different sort, take the two pairs of wire (blue/orange) from the telco that goes to the Adtrans, this is your T1, it has certain parameters, on the line, the line coding and the framing. The device it connects to perhaps does some local transforms about gain either way.

A good description here:-

So, to terminate alternatively on your DIGIUM compliant card you will need to know at the T1 level the framing, line-coding and gain needed which you can get (with some effort) from your Telco , then you go to the DS1 level this might be as described in many ways , but in your case probably a “fractional T1” carrying DS0’s in slots from 1 to 6 as FXO’s (dial tone) and using RBS signalling on the link . Also the Telco should be able to get you all the USOC’s

When you get that right , you have networked the DS0’s as audio channels that given the right signalling , can connect with any phone line anywhere in the world.

Next step , connect your legacy PBX to the other T1 similarly provisioned but slaveing to the master span on 1 for clock timing and cross connect channels 1-6 on span 1 to 1-6 on span 2. You probably need a cross-over for that to keep RX/TX on the correct pair.

In dahdi system.conf add to your span definition defining timing, framing, line coding and attenuation


at this point and checking your TX/RX pairs for correctness your box should be totally transparent to your Samsung and you haven’t yet had to worry about FreePBX/Asterisk.

Baby step two, peel off a fax channel DS1/channel 5 or 6 and dacs it to a free FXS your fax machine is plugged into

Still no FreePBX/Asterisk needed, the fax machine should be able to send and receive faxes using the DID of the channel diverted.

. . Then try replacing the DAC for DS1-3 (channel 3) as a local FXO channel which would use confusingly FXS signaling probably loopstart but possibly groundstart !! and voip it through FreePBX/asterisk . . . .

Now you are using FreePBX/Asterisk and the telephony world is your oyster :slight_smile:

The handsets are proprietary, you can sell them with the switch when you want to but you won’t get much for them :wink:

LOL - Yeah, simple as that.

Actually, just go slow and you’ll be fine.

A “translation” of what he said was "find the wires that go into your PBX that are used by the FAX machines and dead-head them straight to the AdTran. Everything else gets plugged into the Asterisk server through the individual plugs on the DAHDI card. By the way, I use Synway FXM3200(E or P). They are a four-port card that ‘cascades’ through a daughter board. Each card has 4 plugs (where you plug in your phone lines from the AdTran), so get as many cards as you need and as many modules as you need. Sangoma make similar cards, but I like the SynWay cards.

Since you are networking guy, it might help to think of the new project as a client-server set. The Asterisk server is the (not surprisingly) the server. The phones are clients. You can set up other Asterisk servers at other locations (through IP) and use them as ‘gateways’ - or not. You can set up all of the phones in the system just like you would desktops.

The only place that gets strange is the fax machines, and that’s because they are aren’t network devices.

Actually Dave that’s not what I am suggesting, I am suggesting that telco goes to your new box T1(in) and T1(out) goes to the Adtrans, map with DAHDI T1(in) to T1(out) on a channel to channel basis. That is the transparent bit of the inline T1 path.

Eventually he can eliminate the Adtrans and connect the Fax machines to the FXS ports on the new box after cross connecting channels 5 and 6 to the FXS’s instaead of the Adtrans. Then at leisure he can remap one by one the DAHDI voice channels to either the Adtrans, but ultimately the Asterisk/FreePBX service. The old Samsung is still available on the T1(out) channels 1-4 if he wants to get clever or preserve the old hardware.

Hi Dave,

Here is a drawing of what I have going on. The lines labeled 3 and 4 would be the lines that I would like to take out of the punch block and plug into the asterisk telephony card. So would I be ok on getting one of those dahdi cards? As you can see I would leave the fax machines exactly the way they are connected.

Since it’s only two lines, I would only need a two port FXO card correct?

Thank you again for all the information. I do understand the client/server analogy. I never really thought of it that way, but makes sense. I guess I’m just having trouble understanding what hardware is used to get out to the PSTN with what we have in place.

Thanks again for posting great info!


I like the picture.

It’s reasonable to assume that the AdTran is acting as a channel bank. A “Channel Bank” is an old-old-old term that basically takes the incoming T1 (all 24 channels) and breaks it out into 24 individual T0 channels. A “T0 Channel” is the theoretical version of a standard 2-wire telephone circuit. A typical physical representation of a “T0” channel is a beige plastic box stuck to a wall with red and green telephone wire inside that you plug a Plain Old Telephone Service (POTS) phone.

One more bit of “technological political correctness”. When I talk to people about the technology, it’s important to me that they understand that “Asterisk” is the PBX software that actually handles the phones and incoming calls. There are other free PBX programs - some of them are off-shoots of Asterisk, others aren’t. Managing ASTERISK is a huge PITA without some kind of software management package. FreePBX is where that management happens. Since we’re here in the FreePBX discussion forum, let’s assume you are using FreePBX as your management interface to an Asterisk server.

if that’s true, life is good so far.

Here’s a good article about what KIND of ports will be talking about:

Your AdTran is probably breaking down your incoming T1 into six phone lines. Four of them are phone lines, two of them are FAX lines.

For your Asterisk server, a four-port DAHDI interface card would be plenty. I don’t know of any “two port” cards - there are single port cards (which are almost all uniformly crap) or cards that take “personality modules” that turn the four jacks on the card stave into a pair of dead ports and a pair of FXS ports (or FXO ports) or two pairs of plugs (2 FXS and 2 FXO, or 4 FXS or FXO ports). So, my recommendation is get a good four port card and get a single FXO module - that will give you expansion capability into four ports later by adding another module. When you are ready to convert everything, you can pick up another FXO module.

TECHNOLOGY RUNNING AMOCK - DOGS AND CATS LIVING TOGETHER WARNING! Of course, you can go completely NUTS if you want. You can get an “eight-port card” (which will probably be two four-port card hooked together) and then get two FXO modules and plug the four AdTran lines into that, then get two FXS modules and run these four lines to your existing PBX. Adding your Asterisk PBX in front of your existing Samsung would be crazy stupid, but you could do it.

While it’s perfectly reasonable to assume that the four “phone” lines coming from your AdTran are just individual phone lines, it’s not guaranteed. Get a butt-set and plug into those lines at the first punch-down block. If you get dial-tone, you have four individual phone lines. If you don’t get dial-tone, there’s more going on and we need more information (the T1 is split and you’re getting the rest of the channels delivered to the Samsung, for example). It’s also possible that what your phone company is calling a “T1” is actually an ISDN PRI, which is delivered on a “T1” circuit, but is a different kind of beast. The butt-set test will give you more information.

That last paragraph is a big part of the reason why it’s so hard to get straight answers. There are no less than three ways for the phone company to deliver phone calls to you. Most of them terminate in a “T1-like” connection (actual T1, PRI, and Frame-Relay). An Asterisk “T1” card can handle most of these. One of the funnest things about this is that they all connect to your server with an RJ-45 connector.

Having said all of that, I’m going to make a few assumptions from which you can use or not use whatever helps.

Let’s assume you have four lines with dial-tone. Since you are connecting to lines that have dial-tone, you need however many FXO ports on your PBX to plug into the FXS ports on the punch-down block. To answer one of your questions - you can plug any phone-line with dial tone into an FXO port on an Asterisk PBX.

Not confusing at all, is it?

When the incoming phone line “rings”, the PBX (assuming it’s properly configured) will answer it and route it to “somewhere”. depending on how you configure the server it could be a phone, an application (like an IVR) or just hang up. If you want to place a call, you call the PBX and it picks up the phone line to place the call.

Rich, we were in a similar situation.

Like cynjut suggested, it’s good to go slow.

We have an older Avaya CM system running and are slowly migrating our phones to an Asterisk PBX. We still have both systems running simultaneously and will for some time.
We didn’t want to forklift replace the Avaya system (i.e. throw everything out at once), so we ran internal trunks between the two PBXs and can make calls from a phone on Avaya to a phone on Asterisk.
Our Avaya can’t do SIP, but had a couple H.323 licenses that we could use. We also installed a DS1 board on the Avaya and connected it to a PRI gateway, giving us additional ISDN/SIP trunks between systems.

If your Samsung can do SIP trunking or ISDN, I recommend you go that route also and simply run a trunk between the two PBXs. This will have the advantage, that you can go as slowly as you want implementing the new system. Fire up the FreePBX server, try a couple of IP phones and you won’t have a lot of disruption migrating. You could keep your T1 for now and worry about replacing it when you have installed the new PBX and tested it for a while as well as keep doing your faxing on it. Just route calls to the PSTN made from new IP phones through your Samsung or get a few SIP trunks.

We are also running plenty of faxes and still have them on our old system. This way we don’t need to worry about issues with using faxing over IP.

Hi Dave,

Thanks so much for explaining what the AdTran is really representing in terms of service. I never understood it’s functionality. Having a poor understanding, I just thought it was our T1. I did know that a T1 circuit is capable of 23 or 24 voice lines, but wasn’t sure of the inner workings of the AdTran. I’m very curious if the 4 lines to the punch block give dial tone. I don’t have a butt set, but probably should get one for future use and troubleshooting. In the mean time, I was going to punch down a modular end and plug a standard analog phone into the jack to see if I get dial tone.

I jumped the gun a bit and purchased a 4 port (2 FX0 / 2 FXS) Asterisk TDM410 Card and a $60 business class Intel dual core hp midsize tower to run FreePBX.

As long as those 4 phone lines get dial tone, I’m going to pull two lines from the punch block that are used for the one department and plug them into the Asterisk card.

From there I’ll experiment with getting the trunks setup and configure Asterisk using FreePBX.

If it doesn’t go well or don’t have time to complete the setup, then I can just re-punch the two lines into the block that reconnects to the Samsung PBX and all communications would remain as they have in the past.

Thank you for all of your help. I’ve learned a great deal just from your replies! I’ll keep my progress posted.



I was able to check for dial tone on our T1 for two of our numbers that I originally planned to use for the voip phone system with success. I patched the two numbers from the adtran to the analog card in the system I have setup with freepbx.

So far, I was able to get two hard phones registered (Polycom soundpoint 550). I still am learning on how to configure other features, but managed to get inbound and outbound calls.

I’m ready to move the small department off of the digital samsung office serv 100 system to the voip system for a trial run.

I have the polycom soundpoint 550’s setup, but I’m not sure the proper way to configure the line buttons on the left hand side. There are four physical line buttons and I’m not sure if they should be programmed for internal extensions or if they should be the outside lines. If they are typically used for outside lines, then I’m not sure how to configure them because the configuration asks for User ID and Password. This makes me believe it is used for extensions. If anyone is familiar with those, I’d be happy to hear how they should be programmed.

Thanks again for all your help and suggestions.

While I have almost no experience with the Polycom phones, I’m pretty sure those buttons are for extensions. The manual implies that you set those up with extension numbers. I’m sure there are more than a couple of people here that can give you a “best practice” for how to set up the phones.

Remember - there are no such things as “lines” any more. The extension connects to the Asterisk server and the Asterisk server answers your DAHDI lines.

If you can dial “*65” and get the current extension number, your basic phone is set up correctly. Your incoming lines appear to be working, so it sounds like you just need to experiment with the configuration of the server to make it do “the cool stuff”.

*65 replied with the correct extension. I’m now poking around with ring groups, IVR’s, voicemail and some other functions.

I did notice that the inbound caller id doesn’t seem to work for the analog lines. I tried a couple global configs, but they didn’t seem to work. I’m thinking the CID isn’t possible due to the Adtran. Do you know of a work around?

Thanks again!