Call landline to get voip dialtone

first off: I’m a techie who’s heavy into Linux but is a telecom newbie. I’m not familiar with the terms, hence, I don’t know which keywords to search with. I won’t be asking for a walkthrough. A point in the right direction will be most appreciated.

I have two Linksys ATA phones/routers from vonage with analog phones attached to them. Let’s call them phones A and B. The linksys phones are wired to a linksys wireless router which gives them their internet connection.

I have two ordinary phone lines at home. Let’s call them C and D.

What I want to do is, away from home, call either C or D (my ordinary phone lines), punch a few numbers and get a dial tone form A or B (my voip phones), and make long distance calls with it.

Is there a term for this setup? Is this possible to implement using Asterisk?


Are trying to achieve toll-bypass using Vonage? If so then you will have a couple of problems.

  1. Although Vonage uses SIP it currently does not work with Asterisk unless you purchase the softphone option.

  2. I have never used the Linksys ATA devices but I think their firmware is designed to only work with Vonage.

If you want to achieve this it would be best to get service through another provider (broadvoice, simple telecom) and purchase a ATA device that you can work with (IE: Handy Tone 286).

Yes it is.

I don’t use Vonage, and I am outside the US, so all I know about Vonage is what I read from other replies, but as I have read (confirm this yourself) you are unable to do any kind of configuration on the Vonage equipment, so the answer I give here assumes that you cannot play with Vonage itself. If you could, the answer would be different.

Well, first step, to be able to call C or D (using a regular PSTN phone line) you have to setup a FXO board at home (i.e. Digium X100P - see and eBay) . In this way you will have an Asterisk box (lets call it S1) connected to the Internet through the regular LAN card to your Internet router and to your PSTN phone line (C or D) through the FXO board.

In your office, your ATA can be connected to another Asterisk box (S2) that has another (just like the first one) FXO board. It will be connected to the A or B VoIP Vonage “phone lines” (plugged into your ATA). This box is also (I assume) connected to the Internet through a LAN card.

You can setup the S1 box to answer the C or D phone lines when they ring and pass you to a menu where you can chose to connect to the other asterisk (S2) and from there pickup the A or B lines (Vonage) and make the call.

Everything would be MUCH EASIER if you had a PSTN phone line close to your asterisk S1, so you could have the two FXO connected there and no need for two boxes and link load.

Well, if I understood you right, that’s it.


much thanks to the quick reply. i have read your post a number of times and am beginning to understand how it works.

for clarifications: all the phones (A, B, C and D) are at one place (my house). i do not plan on hacking the linksys phones. i only aim to get a dialtone from them through analog means. for all intent: let’s look at the vonage phones as ordinary analog phone lines as i plan on connecting them using their r11 jacks.

here’s a scenario: i’m at a mall and suddenly remembered that i have to make a long distance call. i have no phone cards but have a few coins to make a local call. what i do is phone home (call either C or D), get punch a few numbers and get a dialtone from either linksys phone (A or B), then dial the long distance number. I was thinking it’s possible to make asterisk as a bridge of sorts between the ordinary phone lines and the voip phones


with all the phones in one place, it is really easier. :smile:

Connect two FXO boards (lets call them F1 and F2) to an asterisk box (call it S) and connect F1 to A (Vonage ATA adaptor-through a regular RJ11, as if it was a regular phone set) and F2 do C.

Make asterisk answer C when it ring and if you punch the right password (for example) it will take you to a menu where it will wait for a long distance phone number you will dial. After dialling, asterisk will use F1 to dial through A your long distance number.

I am in São Paulo, Brazil and so I can’t call any 1-800 US numbers from my regular PSTN… so I do something similar (without the Vonage BOX) to dial US 1-800 numbers through FWD. It works just fine. :smile:

Will give you some trouble on installation and configuration, but works… you can configure this same S to accept VoIP calls (if you are using a Softphone from some other place) to dial using Vonage through your Asterisk box from a remote location…

again, my most gracious thanks! will order the fxo unit statim. i will document the whole installation and post it here.


I was just wondering… you can do that dialing in from the other Vonage line (just connect each of the FXO board to one of the Vonage) so you can eventually benefit from a local Vonage number if you are in anyther county or so…

But that’s another story… :smile:

Oh! You will need not one but 2 FXO connections… If you plan to use the X100P, each board has one of them, so you would need 2 boards.

I am having some small problems with the X100P I am using, but nothing serious. I would advise you to read more about them before making a decision on which kind of boards to buy. Digium offers a new product that can have until 4 FXO or 4 FXS or any combination of FXS and FXO that has called my attention, but I haven’t read much about it.

Oh… Try eBay and the X100P… VERY HARD TO BEAT the prices, specially for testing purposes…

Be advised that I have never actually done this bridging between two X100P FXO interfaces, and as they use the same kernel module, there could be some kind of problem. I have never read anything about possible problems, though.

reading a bit, i saw that modems with the sm56 chip can be used. will try that first. awfully cheap pre-owned here (manila, philippines), ~USD 2.

the digium card is pretty pricey after shipping. but will try that once the x100p rig proves to be successful

Once you’ve got the FXO cards working, you want to look at the application known as DISA. With about 5 minutes work, you’ll have the whole thing working. If you want to play around with it while you’re waiting for the FXO card(s) to come in, you can setup DISA with any IAX or SIP account. (IE, when the call comes to you via IAX or SIP, you send it into the DISA app. DISA will then dial out via IAX or SIP. You’ll then just setup the FXO to send the call into the same extension).




is this the DISA we are talking about? Does it come with Asterisk?

Thanks for the heads-up. without it I might have been stuck before I even started

That’s the one. IIRC, it works exactly as advertised. No real quirks or bugs. :smile:

(That said, if you have troubles, I’d suggest you look first to make sure your DTMF [the touch tone] is getting passed correctly. That’s a common place where mistakes occur.) (As long as you’re working with an FXO or IAX it should “just work”. If you’re trying to make it work with SIP, then make sure your dtmfmode setting matches your providers.)


Binfone and slide,

DISA is not necessary for this kind of bridging.

I rather go with a normal extension setup where I get the long distance number to be dialed from the calling party and then use a simple Dial() on the Vonage ATA. I think it is even easier to configure.

That what I do with my 1-800 configuration, no need to DISA.


Would he not require 2 of each type of port?

2 FXO in incoming PTSN line
2 FXS for outgoing/incoming from the linksys ATA box?

the ports on the ATA box are not FXO, they are expecting a regular telephone…


Well, lets review the concepts…

A FXO device has to be connected to a FXS device and vice-versa, right? You can’t connect one FXO to another FXO and one FXS to another FXS.

Your telephone set is a FXO device, so it must connect to a FXS port. As your telephone is a FXO, the ATA port must be a FXS, right?

Well, once your telephone set is a FXO and it connects to the PSTN telephone line, so, the telephne line must be also a FXS, correct?

If the ATA box is a FXS, how can I connect it to another FXS port, in the PC? It must be connected to a FXO, as if the PC was the telephone set itself. The ATA box behaves like a PSTN telephone line.

As for the 2 sets, I guess what he wants is to do that do only to A and C, and not to both PSTN lines and both Vonage lines. If he wanted to do that, he would need 4 FXO ports, not 2 FXO and 2 FXS.

An FXS does not typically connect to an FXO or vice-versa. An FXO (office) connects to the PSTN (incoming phone line) and an FXS (station) connects to a phone. The big difference between an FXO and FXS is that an FXS provides ring voltage. ATAs usually have FXS ports.


well, seems quite logical that if you connect your telephone (which is a FXO, as you said) to the ATA FXS port and to the PSTN phone line, then both the PSTN phone line and the ATA FXS port provides the same protocol (analog in this case, but still a protocol). As so, you can use the same telephone set with both of them.

In other words, the PSTN line and the ATA port behaves in the same way to a telephone set that is connected to them. So, they must provide the same signals. I can, without loss of content, say that the PSTN phone line that reaches you delivers a FXS signal.

I thought it was this way before your question, but I just found some supporting documentation at

This documentation suggests clearly that I can call the PSTN line a FXS port. And that the port on your phone set is a FXO port.

I ask you that if you can point me to some other (better) documentation that states the opposite, please do, as I am still researching this content. Thanks.

I will quote the definition found there:

FXS - Foreign eXchange Subscriber interface (the plug on the wall) delivers POTS service from
the local phone company’s Central Office (CO) and must be connected to subscriber equipment
(telephones, modems, and fax machines). In other words an FXS interface points to the
subscriber. An FXS interface provides the following primary services to a subscriber device:

Dial Tone
Battery Current
Ring Voltage

You may also see the FXS acronym rendered as Foreign eXchange System.

FXO - Foreign eXchange Office interface (the plug on the phone) receives POTS service,
typically from a Central Office of the Public Switched Telephone Network (PSTN). In other words
an FXO interface points to the Telco office. An FXO interface provides the following primary
service to the Telco network device:

on-hook/off-hook indication (loop closure)

How it Works

Because of the characteristics described above, a telecommunications line from an FXO portmust connect to an FXS port in order for the connection to work. Similarly, a line from an FXS

port must connect to an FXO port in order for the connection to work. When the FXO port on your
analog telephone is connected to the FXS port in the wall, you receive (FXS) service from the
telephone company and you hear a dial tone when you pick up the phone.

You are correct, traditionally an FXO connects to an FXS. I typically do not think of them in that context, I had my VOIP solution hat on. If a client has an analog line coming into their office I think “they will need an FXO”, if they have a Fax machine I think “they will need an FXS”.


I conduct some more research on the terms, it seems that the FXO/FXS terminology is much older than VoIP itself.

This paper from Texas Instruments describes the origin of the terms:

I guess they can be used to identify telephone sets and PSTN lines just as VoIP interfaces, for what I have found…