I would like to set up an asterisk system on a slackware box that I have at home. This needs to pass the “wife test” and be seamless integration for her i.e same analog phones and same analog line coming in from the telco. Now my question is what do I need hardware wise to get this working. I know i need a FXO to connect to telco and FXS i believe behind to connect to my existing POTS lines in the house. I would like to know if I can use one FXS module to run to my main connection for the internal lines (5 phones)? My main goal is IVR, voice mail, extensions and blacklist for telemarketers etc. any help is appreciated.Thanks
yes you need one FXO (i recommend not X100 clone- get a real card) and at least one FXS. Keep in mind that each FXS is one addressable extension- you can plug all your house phones into one FXS but then they are all the same extension and you can pick up on each other etc.
If you want the extensions to be individual they each need their own FXS ports.
You might also consider IP phones- having real xfer/conf buttons are much nicer than having to flash and enter a star code…
As far as the “wife test” is concerned, I suggest reading and fully understanding the documentation comments in the sample extensions.conf as well as reading the O’Reilly book “Asterisk: The Future of Telephony” which is available for free online (creative commons licence).
Once you fully understand it, you will be able to do things like letting people dial phone numbers like normal and have them automagically go out through your analog line, but then internet calls are routed correctly as well as internal extension-to-extension calls. This makes it totally transparent for those that can’t remember to “dial 9 first”.
(disclaimer: my example assumes you are in the USA) What I mean is if you normally dial 5551234 now, most people program PBXes for that to be 9-5551234… but in my scenario, just directly dialing 5551234 does the same thing – since we know that 555- is a local number. Also, if someone just dials 1+something, we know they are dialing long distance.
In my asterisk configuration, anything starting with 18 (1800, 1888, 1877, etc) routes over the internet through gnophone. Dialing 0120 goes through my PSTN line (toll free prefix in Japan). 080 or 090 are assumed to be cell phones. Any other 0- number that isn’t my area code (0176) is assumed to be long distance and depends upon whether that line has long distance privelages. Once I get my Skype gateway working, I plan to make it so that I can directly dial US long distance through it with 1+areacode+number.
I also have some funky emergency number setups, like 911 forwards to the english-speaking emergency number, while 119 still goes to the Japanese one. It also is setup so that in case there is a problem with the line, these emergency numbers keep trying automatically until the caller hangs up.
Anything is possible
Thanks for the help Now if I only use one fxs card (connected to the POTS lines) am i still able to to have distinctive rings for various people in the house? This would be nicer on the budget and help to pass the “wife test”. One more question what about these all in one boxes
Does this take care of all the config or do I still have the capability to use * for config and this box for connectivity? Once again thanks for all the help.
if you’re using a Zaptel card then you can use distinctive ring types to distinguish between calls. to be honest, i’ve not had the success with this that i wanted, and the need for simultaneous calls persuaded me to use VoIP trunks to a provider for additional lines.
i’ve no experience with the Zoom FXO/FXS, but i would hope the same way as the SPA/Linksys 3000 … regular phone connected to FXS port, and PSTN line to FXO. register the FXS as a SIP user, and the FXO as a SIP trunk. let Asterisk dialplan do the rest ! but if the Zoom doesn’t support distinctive ring then you’re back to the Zaptel card.
dring- depends on your question.
On your FXO (phone line) side, if you use zaptel cards of some kind Asterisk can detect distinctive ring and route accordingly. IE, if it gets one ring cadence it will ring phone 1 3 and 4 then go to bobs voicemail, if it gets another cadence it will ring phones 2 3 and 5 then go to marys voicemail.
You are still limited to one analog channel, so you can still only have one person on the phone @ a time.
Asterisk can generate distinctive ring cadences on the FXS (house extens) side but that will not help you- your phones will still all ring. Each individually addressable exten must have either its own FXS port or be an IP phone.
As far as costs go you might go the cheap way and buy ATAs- a 2-FXS port ATA can be had for about $60. It will take more setup though, because you must set up SIP on *, and SIP on the ATA, and the ata itself for your needs… which is why if you can afford it I recommend zaptel cards. They are much more $$/port though.
As for the wife test you might consider waiting for * 1.4 because it will have SLA- shared line appearance. That way if one exten is on the phone they can join a call in progress to another exten.
The device you linked is an ATA- it just has one FXO and one FXS port It is not a phone system by itself, it needs *.
Thanks for the help. AS of right now, my goal is not to have multiple lines, that will come down the road. What I am looking to do is have the distinctive rings ( on all phones is fine) voice mail for each user and the telemarketer blacklist. So from what I am understanding the Zoom device I linked will work for me to achieve these goals. Is there any other hardware possibilities in the same price range ($70 can) I should be looking at instead of the Zoom? Thanks
keep in mind- when i said distinctive ring I meant for Zaptel cards. Not sure if the Zoom device supports it, check the specs.
As spec’d the result will be that a call will come in on your one line/one number, get an IVR prompting the caller to choose their destination. All the phones will then ring, although they will ring in a different pattern depending on who the caller selected. Then anybody can answer, and if someone else picks up they will have a 3way call. If nobody answers, it will go to one voicemail box or another.
This sounds like a bad idea for the wife test, at the very least make sure it’s emailing out the voice mails. With one shared line for the whole house, all it really adds is an IVR and possibly blacklist, and I can see a voice mail sitting for weeks in somebodys box because they didnt know to check it (as there was no MWI).
Consider for the time being not installing the IVR, just using the blacklist. IVRs frustrate callers as they can’t ‘just call’. Maybe make it sort of an intelligent answering machine- when they call the phones ring, but if nobody answers then they can select who to leave a msg for?
Also if you got voip service like broadvoice or viatalk you could save some money on both sides, maybe get the service for a while, see if you like it, and if so port your phone # over. The ATAs sold by BV and Viatalk can be unlocked to work with * and the service can work with * too, so you have the service go straight into * via SIP, and then you have the two ATA ports…
hmm ok I will check in on the Zoom device specs. And your suggestions about the IVR etc make sense. The good thing is at home we don’t use the phone very much (I deal with phones too much at work as it is so the “wife test” is can i call out? can we receive calls? voice mail as long as it is there its all good Now as far as VOIP providers Any suggestions of who to use in Canada? Any boxes that can be unlocked if i don’t like their service?
ps just noticed this link
Why would you want to block NPA-555-xxxx calls? The 555 Exchange is not a pay-per-call exchange per se nor a designated high volume exchange (see www.nanpa.com).
You will not power your whole house wiring with one FXS port.
Two phones maybe three with REALLY SHORT wires.
And you would need to disconnect from the telco and then reroute the wiring.
[quote=“bubba”]You will not power your whole house wiring with one FXS port.
Two phones maybe three with REALLY SHORT wires.
And you would need to disconnect from the telco and then reroute the wiring.[/quote]
It really depends on how many phones you have, how big your house is, how well it’s wired, and what kind of ATA you have.
First you will of course have to disconnect your telco interface. Keep in mind many alarm systems and such things have ‘one way wiring’ through a device called an RJ31x jack, if you have this then you will need to rewire it so your alarm can use the ATA. Ideally you would put the ATA before the 31x jack so it will still work correctly.
However after that it will often work fine. You need to watch your REN (ringer equivalence number), a standard phone line generally has an REN of 5.0 or so. Most ATAs put out between 3.0 and 5.0. A standard analog telephone with a ringer bell uses up 1.0, electronic phones usually use less, and powered phones (such as cordless phones) often use up as little as 0.2 REN.
Of course if your house has old, noisy wiring that crackles then you will get noisy, crackly VoIP.