Forgive me if this has been answered, but I’m wading through the forum and various links at the same time (almost)
My situtation.
Two UK locations (site A and B, B being the main site). Each site has POTS and ADSL. At the moment, if a call comes to site A, and the person required is at Site B, the caller has to hang up and call other site. I would like to be able to foward the call, ideally using VoIP.
I would like to be able to front end the system with an IVR, e.g. press 1 for site A, 2 for Site B, 3 for… etc.
does Asterisk fit the bill? I suspect it does
Can it forward a POTS/PSTN call to a VOIP call (suspect it can)
Assume the call scenario above, and a caller is now talking to someone at Site B. Will a second caller, calling site A, recieve an engaged number tone? ie is Site A pstn line engaged until the first caller disconnects?
What equipment, other than asterisk, and a linux PC and network card, do I need to attach the box to the phone line
Assuming I need >1 line to address Q3, what should I order (second POTS line? ISDN? remembering I need to retain dsl) from telco.
5b. What additional hardware would I need for the asterisk box?
Many thanks, and apologies if these are answered elsewhere. I’m a data person not a voice bod, but learning.
Of course. But someone else can still call the site B pstn line, this call can be answered from site B of course but also from site A !
You need an FXO card to connect to your PSTN line and an FXS card to connect your classic PSTN phone. The easiest i think is to use a Digium TDM400P with one FXO and one FXS module… you can find cheapest solutions but the TDM400P is for me the best compromise, compatibility, expandability, quality,…
Now, you can also have a look at full voip solutions : forget your PSTN line and take some VOIP provider wich will give you one phone number directed to your server via internet that many people can call at the same time…
With this possibility, you just need one asterisk server, one voip provider, and two VOIP phones (one for site A, one for site B).
Well, as I was typing my response TWO people beat me to it. All I can add is this:
An ISDN line will require a different type of card than the Digium TDM mentioned. Something like: junghanns.net/en/produkte.html. I think you should get 1 or 2 more POTS lines and just add FXO modules.
Check out this site for the best starting place as far as documentation.
Any recommendations? I had a quick look at a couple of providers, one thing that is not clear is whether their services have a limit on the number of inbound connections/calls per number. They implied one call per voip number allocated…
I don’t know in UK… most of belgian and french provider allow at least two concurent incoming call…
Just for info : i use sipdiscount.com wich is a uk voip provider, but only for outgoing call, and i can handle at least 3 simultaneous calls, dont know for their incoming call services.
Have found some cheap (for testing purposes) genuine X100P FXO cards. These connect the server to the PSTN line.
What connects the handset to the server? Some of the “clone” X100/X101 cards seem to have two interfaces - is the 2nd the FXS interface? If so, does the Digium X100P card have this functionality?
If not, what cheap FXS card should I look for? This is for testing, so I’m not fussed it it is cheap n cheerful; I know ideally a TDM400 series with FXO and FXS would be better, but I have a plan for the testkit afterwards.
2nd question, looking further ahead. How does one attach a VOIP phone to the system? Assuming there is an FXO (to retain PSTN connectivity), and VOIP calls via LAN interface. Assumption that both can route to a ‘legacy’ phone/dect handset/whatever. OK, how do I attach a VOIP phone?
I’m asking so I can buy the right gear for testing, then look at the total costs for deployment, and avoid mistakes!
have you read the wiki yet ? your questions are answered there.
a single port FXO card will not let you connect your “handset”. you can either use an FXS card (rare these days, unless you use a TDM or equiv card), or an ATA. i use the linksys PAP2 and find it OK, but nowhere as useful as a proper IP phone … which connects to the network.
a cheap card for any deployment, including home use, will ultimately disappoint. people notice quality. use the cheap card to learn Asterisk, but invest a bit more when you want others to use it.
Have found some cheap (for testing purposes) genuine X100P FXO cards. These connect the server to the PSTN line.
What connects the handset to the server? Some of the “clone” X100/X101 cards seem to have two interfaces - is the 2nd the FXS interface? If so, does the Digium X100P card have this functionality?
If not, what cheap FXS card should I look for? This is for testing, so I’m not fussed it it is cheap n cheerful; I know ideally a TDM400 series with FXO and FXS would be better, but I have a plan for the testkit afterwards.
2nd question, looking further ahead. How does one attach a VOIP phone to the system? Assuming there is an FXO (to retain PSTN connectivity), and VOIP calls via LAN interface. Assumption that both can route to a ‘legacy’ phone/dect handset/whatever. OK, how do I attach a VOIP phone?
I’m asking so I can buy the right gear for testing, then look at the total costs for deployment, and avoid mistakes!
And thanks for the very useful replies to date![/quote]
I’m probably a couple of weeks in front of where you are now.
i hadn’t even heard of asterisk till a couple of weeks ago.
for initial testing i downloaded a vm copy of asterisk at home and used that. Now i’ve got it set up in my office.
there are various devices that connect the handset to the server. sipura do them, now bought by linksys. SPA-3000
connecting the voip phone, i’m assumin you mean something like the sipura (linksys) SPA-941 this is done via your ethernet network.
once you get into it there are a ton of possibilities… hence why i’m still in ask the daft question mode
@baconbuttie. Yep, taken on board. I wanted to test using kit available, and with minimal outlay. A TDM card with FXO/FXS is likely to cost around £150+ hence the hope that /something/ out there would allow me to attach an existing ‘legacy’ phone to the box for not a lot, just to proof-of-concept test.
SIP equipment is going to cost, and doesn’t prove what I need to prove right now, which is basically call->asterisk->handset (existing) and vice-versa.
I have a few wireless handsets. I need to leverage existing equipment, not replace it as this destroys the cost savings. And, taking into account the time, I would be just as well off using the telco provided service which would provide much of the functionality. Having been tied into a provided service before (on a massive scale, e.g. national to the tune of £xx million), I’m wary of vendor tie-ins!
On a competely different note, its interesting that VoIP has finally driven down traditional PSTN dial costs… my provider is cheaper than most VoIP callin plans I have seen!
While I agree that the POTS can be cheaper. With a sip or IAX2 provider you can split the channels so if they allow 3 incoming lines (Which my provider does) then you can actually connect 3 handset to 3 incoming calls with a PSTN provider you need to have 3 incoming lines. Also with a VOIP provider you can have an outgoing call active from hadnset 1 and when someone dials your number handset 2 and 3 can ring.
