I just started playing with A@H for use in my home and I have a couple of newbie questions that I hope some of the ‘experts’ out there are not tired of hearing.
Essentially, I would like to get asterisk to work with my home phone so that I can:
Plug-in my pots line into my asterisk box
Plug in traditional phones to asterisk so that they call through (but treat them as one extension).
Based on what I think I know I would need to plug pots into FXO and the home phone(s) into FXS is this true?
Can I plug multiple POTS phones into one FXS if they will all have the same extension?
I am also looking for reccomendations for FXO/FXS or plain FXO hardware that is inexpensive.
I read up a little since my last post (I downloaded the book as a pdf). And I now know the difference between FXO/FXS, which leads me to my next question:
I have a cisco ATA-186, and I know it has two ports. I also know that it supports FXS, because my analog phone gets dial tone. But does anyone know if I can configure the other port to be FXO? I essentially want to have Port0 connect to my POTS line at home and then connect my phone to Port1, so that my phone can dial out over POTS or over VOIP.
it is not possible. the Cisco ATA has 2 FXS ports and that is.
You can not attach a regular phone line to any of the ports.
Well, there are FXS/FXO convertors but their price is not that cheap, so you better go for a combined FXS/FXO gateway. Maybe Linksys or Sipura ?
If you only have one incoming phone line, one of the Digium clone cards (or a real Digium card) that provides FXO should work for you–
That way, your incoming phone line from your local telco could connect to that, and the ATA could support the house phone wiring.
Keep in mind that the REN on an ATA-186 is not very high-- they are not meant to support 4-5 phones-- really only one (maybe two or three)
You’ll get ring issues (delayed, no ring) issues if you connect too many phones.
I suppose you could do a round robin to two separate extensions tied to both ATA ports-- some phones on one, some on the other…
that’d help out with the REN problem perhaps-- but I don’t know how you’d do the roundrobin thing in the Asterisk server…
My situation is very similar and here is what I did, which works well so far.
I had a POTS line coming in from my telephone company and with the help of a neighbour who works for them, I was able to get into the telco side of the demark on the outside of my house.
From there, we simply clipped the lines between the demark and the homeruns which go to each RJ11 jack in my house and wired the incoming line into one port of a dual RJ11 jack inside my house. On the other port, we wired all the homeruns, and simply connected a small RJ11 cable from one port to the other to complete the line and ensure that our wiring job was successful.
Once we verified that it was, we introduced the SPA-3000 into the mix, connecting two cables. One going from the POTS line port into the FXO port of the ATA and another going from the FXS port of the ATA back into the homeruns port.
POTS --> Wall Jack #1 --> FXO --> FXS --> Wall Jack 2 --> Analog Phones
Now when someone calls, the SPA-3000 rings all of my analog phones which are connected through the FXS port for 15 seconds and then passes it off via the RJ45 cable to the Asterisk server for voice mail.
I’ve only had the SPA-3000 for a short time, so I can’t recommend it yet, but I can tell you that the setup options are vast and it can take a fair bit of time to get it setup properly. Thus far however, things seem to be working well.
The Sipura SPA3000 as mentioned will do what you want. Although it can be very confusing to configure, there are a few articles (Nerd Vittles, etc.) that have very good instructions on how to configure them to do what you want. Once you’ve had it - you can choose if you want to configure it to further customize it. (You can do one of the suggestions which is to have it act as the gateway to the phones and only pass to Asterisk if not answered, or you can go straight to Asterisk to manage the whole call (which is what I did and prefer).
One advantage of the Sipura 3000 approach over a Digium card is that it provides some good failover mechanisms. On power loss, it will mechanically connect the fxs-fxo port so if you have an extension on the fxs port and power is down, it will bridge the connection. (Just make sure that phone is not dependent on power like most cordless, or battery backed - or you still won’t have a phone:)) Also - if power is up and Asterisk isn’t responsive, you can have it fail-over to the FXS port as well.
For a home setup - this is a very good choice. (You can even use it to provide VOIP service if you decide that you don’t want to use Asterisk any longer).
