Newbie question

Hi all,

I am an asterisk newbie… just started browsing documentation etc… a couple of hours ago. There is a lot to learn, but I have some fundamental questions that I cant seem to find a clear answer to:

I’d like to install asterisk on a linux box at home (as a hobbyist more than anything). I want to be able to receive and send calls over the PSTN from 2 or 3 soft-phones on the home network.
I do not need to connect to a VOIP provider, at least not for now, but I do want asterisk to answer calls, after which callers can dial an “extension number” and the call gets forwarded to the living room, kitchen etc…

So basically just an interface between PSTN and my intranet.
Can this be done with a normal modem or do I need to buy ATA hardware etc?
Any help very much appreciated.


9 times out of 10 your normal modem won’t work. You’ll need an ATA with as many FXO ports as PSTN lines you have. Or you can get a PCI card from Digium/Songoma/

Check to see if your modem might work. This wiki is EXTREMELY useful.

Good luck.


Asterisk can do what you want, but it cannot use a ‘normal modem’ to access the line. If you want to play around on the cheap you can get an X100 clone card, there are a number of sites that sell them. The X100p was a digium product, a one-line FXO interface that was really a rebadged voice modem. However, other voice modems are not supported.
X100 cards are often to blame for a wide variety of issues including but not limited to, heavy unsolvable echo, dtmf detection, problems detecting call progress, etc etc. So it might be good to get started with, but I wouldn’t use it for anything important.

You’d want at least one FXO port. I’d recommend the Digium TDM400 card with an FXO module, however this will be more expensive.

You could also use an ATA with an FXO port (has to be real FXO not just pstn passthru port) but I generally recommend getting a real card due to easier setup. ATAs tho can be easier to install if you don’t want the Asterisk server to be where the phone line enters the building, put an ATA or two there and * anywhere and you have much of your problem solved, just run ethernet to it.

I’d also recommend NOT softphones. They are decent if youe ither have a call center or just want to play around, but once you start using the thing, you will really miss having a hardphone. You can buy cheap IP phones (grandstream BT1xx) for around $40, i’d recommend the $60 BT200 as it supports intercom (paging) and remote disconnect (on the BT100, if you are on speaker and the other side hangs up, it will play a busy tone until you push the speakerphone button to turn it off.). There’s also the GXP2000 for just under $100 which makes a decent desk or wallmount phone, plus it supports power over ethernet. Netgear or Dlink (forget which) make a little 8 port powered switch for like $150 which goes great with this. Both of them now make a few low end powered switches so i’m sure you can find something.
If you want better phones use SNOM or AAstra. You will find a LOT of asterisk geeks with Snom 360’s on their desks… including myself.

as for programming asterisk, yeah you can do all that no problem. You can also assign your family members their own voicemail box, but have other phones ring too-
exten => 3,1,Set(CALLERID(name)=Jen-${CALLERIDNAME}) ; caller pushed 3 for Jen, rewrite the caller id name with Jen- at the front so people know calls are for her
exten => 3,2,SipAddHeader(Alert-Info: Bellcore-DR3) ; specifies a distinctive ring. Many SIP phones support this.
exten => 3,3,Dial(SIP/jen,10) ; ring Jen for 10sec
exten => 3,4,Dial(SIP/jen&SIP/bill&SIP/familyrm&SIP/bedrm&SIP/kitchen,20) ; if nobody picked up on Jen’s phone in 10 secs above, add all the other phones for 20sec
exten => 3,5,VoiceMail(jen@default) ; nobody answered, go to jen’s voicemail

The result of this is that 1. when someone calls for Jen, it is made obvious both by appearing on the phone’s callerid screen and also the distinctive ring. It first rings her phone only for 10 sec in case she is in her office, then rings both her phone and all the others (sip headers and callerid still apply to the second Dial) for 20sec in case she is somewhere else in the house. Then it goes to her voicemail.

Hope that helps!

thanks for the reply!

Found one at for little under $30… this should do right?

yes that will do. However keep in mind that as your system becomes useful you may want to upgrade to a real tdm card… x100’s often suck

thanks ironhelix

yep i saw the tdm400 card, but if i’m not mistaken the price is way above that of the x100p, so for starts (and since im still in the playing around phase) i could live with a bit of echo until i learn the ropes.

Yep I used grandstream phones at a previous place of work… will get to buying Snom or AAstra once the system is up and running.

thanks also for the programming hints… looking forward to getting my hands dirty…



I understand I just need a phone line and then plug and call? Does TDM or X100P cards make then the use of a PSTN router obsolete?
Will my telephone provider charge me for using the line to route my SIP packet?

Don’t know if my question makes sense…

Another newbie.


once you have the TDM or x100 card you plug it into the phone line. it behaves like a normal phone, picks up the line, dials with dtmf tones, etc. as such the telephone provider has no idea you are even using voip and they dont care, to them its just a phone call and * is a phone. So no they dont charge more for using *.

not sure what you mean by pstn router.