SIP --> POTS? Possible and how?

Hello everyone,
This is my first post here.
I’ve been following the progress of this great software for awhile but have not jumped in and used it yet.

My name is Michael and I am a software engineer. I run a small startup company called Melenion Development Studios Inc. However I have a small problem.

Here is my situation.
I will soon be purchasing a number of VOIP Phones for the office (wired and wireless) and what I would like to do is setup a server that will accept calls via POTS (only one line though) and allow people in the office to call each other.

As far as I read, (and using some common sense)
I assume that I will need asterisk and a server with a voice modem right?
Also, how would I got about setting this up? I have two server available that run openSuSe 11 although neither have a voice modem.

Is this possible?
Can someone recommend some hardware that I could use?


Yes you can do what you want, except you cant use a “voicemodem” if you expect anything resembling even half decent call quality.

Have a read of the forum and also down load a copy of “The future of Telephony”

and you will find all the basics you need


Alright Thanks,
I have done a little research and I have found that I will need a FXO correct?
I have looked at a few, and am considering the GrandStream GS-488, SPA3000, SPA3102 or the X100P SE from, what would you recommend for a single line FXO?


Although I personally have never used one I was told that the SPA3102 is good. You can also try to get a X100P off of eBay. Again never used a X100P but it should do (I only play with the “bigger boys” :wink: )

I too have not used the SPA3102 but it looks like a good unit, and the one I plan to pick up in the future. I do have a couple Grandstream HT-503’s, one of which I’m using the FXO port which forwards the call over to an IP phone (Grandstream gxp2000 - I love that phone!) and it seems to work ok.

Why I’m not recommending the Grandstream ht503 is that I cannot get the FXO port to appear as a trunk, allowing my dial plan to make routing decisions based on the dialed number (ie. go out my local phone line for local numbers and use my VOIP provider for all other long distance). Right now I have set it up as an extension (999) that I have to dial into, get a second dial tone, and then dial my local number.

As far as a server goes, I’d highly recommend getting a dedicated box for Asterisk. I’m running on an old PIII 800 mhz laptop and it runs great so far! The nice thing about this is you could then jump in with a preconfig’d distro like Elastix. Toss the iso in your drive, go through a short install, and boom you’re up and running. That’s what I did and it seems to do all I need it too.

Good luck!