Help! Trying to connect PBX in a Flash to PSTN

I am trying to set up a home PBX. I just converted my old Dell Dimension 2350 to a PBX in a Flash Asterisk server. So far I am very happy with everything I have set up by myself. I have AT&T as my home phone service. I am attempting to connect this service to my system. My Dell Dimension has a phone (RJ11 I think) port on it and I am wondering if I can use this instead of having to buy another device.

nope - sorry.

Darn… Okay, well do you know what device I need to do this? I am thinking a Cisco SPA-3102?


I personally prefer DAHDI cards. Both Sangoma and Digium make excellent cards.

Alternatively, you could move your phone service to a SIP provider (such a SIPSTATION) though you would want to test and make sure you are able and acceptable Quality of Service with your broadband connection and current infrastructure before committing.

I thought about that but I have an ADT security system and need to have a phone connection 100% of the time (or close to). I don’t trust my internet connection to be on 100% of the time, so I prefer keeping my standard AT&T landline. Would a TDM400P be a good choice for me? Could I just leave my alarm system connected directly to the landline?

ADT most likely REQUIRES a POTS line and will NOT allow VoIP, so you will need to keep that for the security system and it is a very wise move to do so.

Beyond that, I’m bias wrt to SIPSTATION, I think it is the right way to go if it meets your needs but of course I am not objective on the matter :slight_smile:

Thanks for your quick reply! I am correct, though, about configuration: just install the module and that’s about it?

Also, I can port my existing home phone number to SipStation, right? From looking at the website, I like Sipstation. I think it exceeds my needs (a good thing)!


The SIPStation module is step 1.

Then you need to purchase at least one trunk and then you are presented with a KEY to use in the SIPStation module to set up your trunk(s) and DID(s).

Okay… Thanks to everyone that has helped me here! :slight_smile:

Just to clarify in case I don’t switch to Sipstation…
I am leaving all my AT&T wiring alone. Do you think my ADT system would still work if I just left it connected the way it is to the line and also connected a TDM400P to the line. I am planning on replacing all phones with Cisco 79XXs so my existing wiring would still be “hot” but not in use by any phones (basically the only things connected to the actual AT&T line would by my ADT system and my Asterisk box using a TDM400P). Do you think my ADT system would still function correctly with an Asterisk server connected to the same line (ADT not connected through Asterisk)?

Do you all get what I’m saying? Do you think that would work?

A TDM400P would work great for you. I would prefer to get something that has hardware echo cancellation though…I had an analog line for a while in my home asterisk system with a tdm400p, but got annoyed by always having to use OSLEC to minimize the echo, and also I had issues with AT&T not providing me disconnect supervision. So when someone was calling me and they hung up, my system kept ringing till voice mail picked up and then waited like 7 seconds through the silence. Just a few things to think of. I wouldn’t think there would be any issue with a FXO SIP ATA because that will have an echo canceler on it.

Thanks Riddlebox for the advice.
I have been thinking more about the SIPSTATION option for simplicity. From what I can tell, it looks really easy to set up; all you have to do is install their module and do a few other things, right? With me being new at all this, I am looking for simplicity and function. Has anyone had any problems with Sipstation? I was thinking about keeping my AT&T landline and removing all the features except basic calling b/c the only thing that would need the line is the ADT security system. That would only be about $16.00 or less a month, and then Sipstation would only be another $25.00, and that combined is still cheaper than what I’m paying now. I would like to get rid of AT&T completely, but with the ADT system, I’m afraid to.

The your current telco demark should be an RJ-32x jack.


As long as you connect the VoIP server to the location where the existing telephone instruments are connected, you’ll be ok.

The bottom line is that the telephone line and alarm equipment will not know the difference whether there’s a conventional analog telephone or VoIP adapter placed downstream.


Assuming your alarm system is wired properly, you should be able to connect the internal phone lines to any TDM card as it is “just another fancy phone.”

As far as SIPSTATION, your best thing to do is go purchase a trunk and DID in the portal and get yourself configured.

The module does most of it, you still need to forward the appropriate ports at your firewall for optimal operations.

Now you can test the service with both inbound and outbound. If you like it you can port your number, if not, you can always cancel. If you don’t need the DID you purchased after you have decided to keep the service, you can remove the DID.

As far as phones, if you have not already purchase the Cisco, I would look around at alternatives. Aastra, for example, has tremendous support with great XML scripts available from Aastra that tie very well into FreePBX.

The Cisco phones are one of the most complex to work with (and you did indicate you were looking for ease) and often run into limitations since Cisco has no interest in supporting anything but their own systems… They are good phones, but there are a lot of other good ones to consider as well.