I’ve been considering for the last months an Asterisk, Trixbox or an AsteriskWin32 solution for my business, but found myself confused with the hardware part. I wanted something free all the way. I could handle the dedication and discipline of making the system work, if there was a guaranteed free solution. I’ve been reading some of the free manuals, ebooks and VoIP websites about the technology to answer some of my questions. It seems to me though that to make this work even in the simplest form (1 PSTN line & 1-4 internal extensions), I’m going to need an FXO PCI card for my PSTN line, which as you can see on the digium site can cost from $200-400. There are other cards for more complex setups, which I don’t need. Now, considering that I’m right about the details above and thinking from a business perspective, isn’t it better of buy a PBX-VoIP appliance that can do this and more (through a GUI) for $100-200, or do I buy the FXO PCI card an go through the pain and suffering of configuring the system? It seems to me that even though the Asterisk software can be very flexible and powerful, the hardware prices and variety of cards keep new potential asterisk users off and still confused about the convenience of it all from a business mind-set. Thank you.
I would use 1 fxo to sip gateway for the pstn line and 4 sip phones, all registered and connected together through *.
Regards.
Thanks for your answer mbruni.
Are you well versed with Asterisk?
If there are other options out there (like the one you recommended) cheaper and more convenient to setup than Asterisk, then when & why should anybody consider Asterisk’s use from a business perspective?
Actually, I can’t think of a setup that would be easier to set up and configure than the 1 fxo to sip and 4 sip phones.
- no IAX to fiddle with
- no SIP trunking to configure shudder
- just direct extension to extension calling, and total dialplan control on calls coming in over the pstn line. cake.
.d
to be fair to asterisk, not everyone experiences ‘pain and suffering’ when setting it up. creating SIP trunks do not make everyone ‘shudder’.
considering that given a bog standard linux box and tdm400 you can download and configure a fully functional pbx with voicemail and connectivity to the pstn in about an hour you can see the difference between asterisk and a voip gateway.
Asterisk is going to be far more powerful and flexible and expandable than any $100 - $200 voip gateway device.
seabro
Just for the sake of testing without spending $$$ on any FXO card. Is it possible to make Asterisk work under my circumstances, maybe through a VM, without the need of any special hardware?
Yes it is, use a sip “trunk” provided by an itsp and you’ll have the pstn termination without additional hardware and use a free sip softphone (x-lite, sjphone, openwengo, ekiga) instead of a sip hardphone.
Regards.
[quote=“mbruni”]… use a sip “trunk” provided by an itsp and you’ll have the pstn termination without additional hardware…
Regards.[/quote]
Forgive my ignorance in the subject, but I’ve searched the asterisk manuals for the term sip “trunk” or itsp, but found no info. I know what SIP is, but if you can point in the right direction where I could find more about making a “SIP trunk” or what a itsp is… I would appreciate it. Also, I asume this setup will allow me to connect a lab Asterisk server to a sip softphone installed on another computer (both on a contained internal environment) without access to an external PSTN line. Is my assumption correct? Thanks again.
ITSP:http://en.wikipedia.org/wiki/ITSP.
SIP trunk:http://www.siptrunk.org/.
Asterisk defines a sip trunk as a sip object (in sip.conf) of type “friend” (it could be splitted in it two but don’t want to add confusion 
).
If you just want to have a sip softphone and call an asterisk box, in a lab for some testing, you don’t need a sip trunk nor an itsp.
Hope this helps.
Regards.
i was in your position a year ago. I used an old pIII box that was sitting around, downloaded and installed the software, set up an account at Voicepulse (connect.voicepulse.com) for $11 / month for a DID and was up and running with 7 extensions and access to the world in less than 4 hours. Out of pocket cost was the $25 deposit I had to pay voicepulse.
Later I added the digium card to do some direct PSTN stuff but that was not necessary to get going.
You may be overthinking!
[quote=“mudslide567”]i was in your position a year ago. I used an old pIII box that was sitting around, downloaded and installed the software, set up an account at Voicepulse (connect.voicepulse.com) for $11 / month for a DID and was up and running with 7 extensions and access to the world in less than 4 hours. Out of pocket cost was the $25 deposit I had to pay voicepulse.
Later I added the digium card to do some direct PSTN stuff but that was not necessary to get going.
You may be overthinking![/quote]
I guess, what I’m trying to visualize from a bird’s-eye point of view is the following: If I’m trying to connect the outside telephony world (PSTN) with my Asterisk server, where on the back of the server do I plug that phone line to receive calls in if at this point I have no digium cards at all?
He meant that for testing you won’t need it.
You may just test one user calling other users, use a sip phone as would-be-PTSN to test out-to-pstn dialplans…
Also consider using a VOIP provider instead of a PSTN, for instance if you live in north america, you can get a 300/300kbps internet connection for a low cost and a VOIP line with any area code.
to clarify, with my initial setup, i had a voicepulse account … a trunk using the IAX2 protocols that takes care of the connection to PSTN world. There is no need for a telephone cable plugged into the box to accomplish that.
As I also said, I later added a Digium card so I could plug into the PSTN network directly. The telephone cables plug into the card but you do not need that to get experience and talk to the world.
Old machine, a few dollars for an account with a ITSP and a decent internet connection and you are off and running.
Thanks a lot for the input guys. I get the idea. I guess, I was trying to complicate the situation a bit. I’m currently located in Europe and would like to make the setup work from here. Do you know if voicepulse has any restrictions on certain areas? Will it only service US customers?
If you want to make your setup work for europe i guess you want a DID from your country ?
voip-info.org/wiki/view/VOIP … ess+Europe
By the way www.voip-info.org was a very good friend (and still are) for me when i was starting with asterisk.
Very interesting conversation. I am totally new to * and I am amazed at its possibilities.
I am a computer programmer but the terminology here is like another language in itself.
I would like to set up asterisk to handle conference calls over the phone and also to be able to let users call, put in their id number ( login number ) and record a message.
I would than like to hook up that voice recording to their profile on the website.
How difficult would this be and what are the steps?