on our site we have our small business and our home.
The house has a phone line with just a few phones and the business has 1 phone line with about 5 phones on an internal pbx phone system. Last week the 10+ year old pbx decided to catch fire and that was the end of that.
I was looking for a replacement and thought we might as well move on into the future a bit. The handy thing is that we already have network cables run all round the site for file and internet access on all the computers.
The house and business share one DSL internet connection with 512k download and 256k upload (all we can get in our location)
Now ive been reading up a bit on this and this is what i have planned but correct me if im looking at anything wrong.
All the businees phones can be replaced with hardware IP phones.
The house phones can have converters to make them into IP phones
We can have a basic server running asterisk with 2 FXO’s plugged into our two telephone lines
Then i would set it up so when somebody calls on the business line all the business phones ring and the same with the house line. but people can still call internally.
Also in the future depending on the type of call made and the time calls can be routed differently (eg. through the internet)
The only thing im not sure about at the monent is how internal calls will be made… before the burning of the ancient pbx the internal phones had buttons along the top, one for each extention and one for the external line, and all you did was push the button corresponding to the location you would like to call… nice and easy. Now with ip phones how would we achieve the same thing or how could we have it differently?
has anybody got any thoughts on what im trying to achieve?
also does anybody have any links to internal IP phone systems becasue most of the guides i’ve read seem to be mainly for making cheaper internet based calls…
That would depend on what IP phones you chose. You’ll need to configure each extension with an extension number (in extensions.conf) and then you can call another extension by dialling its number. You’ll probably want a 2-digit number, although you can use any number of digits you like.
Some phones will allow you to configure speed-dial buttons - which you could set up similarly to how you had it before. From the analog phones, of course, you will have to dial the extension number.
Have a read of this book - and buy it, seeing as you’re using it for business
Be warned, though, that to run an Asterisk pbx in a business environment you will need to have a good knowlege of Asterisk and of Linux system administration.
Asterisk would serve your needs pretty nicely, especially as you have ethernet cables everywhere.
You have the setup pretty well down, the converters yous peak of are called ATAs (Analog Telephony Adapters). Depending on how many extensions there are, and where they wire to, you might look into using zaptel channels off a TDM400 card maybe instead of ATAs. One less thing to configure. FXS ports off a TDM400 cost more than ATAs though.
What you are talking about is a few different things.
Many VoIP phones have speed dial buttons, usually not the 10+ that are on a pbx or key system but there are usually 7 or 8. Some phones (with displays) let you scroll through a list and pick one, or have softkeys. This is very easy to do.
Remember, you can always dial by extension, even on the old system usually. The buttons are just a shortcut.
On your old system, each button probably had a light next to it that would come on when that person was on the phone or when the line was in use. This is called Busy Lamp Field (BLF). Asterisk supports it, some (but not all) phones do, and it’s relatively easy to setup.
Third, on your old system, if a user was on the phone and their light was lit up, you could probably push their lit button or the lit button of the line they were on, and be joined into the conversation as 3-way. This is called Shared Call Appearances (SCA) (sometimes Bridged Call Appearances), and asterisk DOES NOT CURRENTLY support it. IMHO, this is the one place where asterisk is seriously lacking.
If a line was ringing on your old system, the light next to the button would flash. Asterisk supports this as part of BLF. You could then push the button for the line (even if it wasn’t specifically ringing your phone) and andwer the call. Asterisk DOESN’T support this as part of SCA. However, it is POSSIBLE to do such a thing, ie pick up another ringing channel, but you have to set it up in the dialplan or use *8.
Lack of SCA is a good thing usually in a business environment (random people can’t barge into your calls), and a bad thing in a home environment (jane is on the phone to dad, bill can’t grab the line and remind dad to pick up his guitar at the music repair store; he has to call jane and get 3way’d in).
yeah thephones with busy indicators sound like what we need.
we have a few phones in busy workshops where its easier to just have a button labled “Office” rather than people having to remember speed dials or extensions.
because of where the standard analogue phones are located and to allow flexability over where the asterisk server is placed we would probably end up using ATA’s rather than the FXS ports.
due to my complete lack of knowledge of linux i was looking into Asterisk@Home… obviously it doesnt do all the things that asterisk can do, but in my setup what will it lack that i might need?
I think it is just worth pointing out that you don’t need to convert all of the house phones to SIP with ATAs. Standard analog phones can work quite nicely with Asterisk and do just about all the functions of a SIP phone. You need ZAP FXS ports in a TDM400 configuration (or similar gear).
I don’t know A@H personally, but your setup is quite simple and standard, so i doubt you’d have any problems using A@H for it.
I would be a good place to start, anyway, and if you find later that you want to get more sophisticated, you will have started to learn about Asterisk and, maybe, Linux and will be in a better position to switch to standard Asterisk.
However, you still need to bear in mind that what you’re dealing with is still computer hardware and software and therefore has all the inherent potential unreliability that comes with that. If you haven’t got a proficient Linux and Asterisk system administrator on hand, you could find yourself without a pbx - which i guess is roughly where you are at the moment though!
what will says is true- Asterisk is reliable, as is Linux, but especially if you don’t know Linux it IS still a computer
As for the speed dials, it would function much the same way (pick up the phone and push the button), I don’t mean like dial 51 instead of 1-xxx-xxx-xxxx type speed dial.
As for AAH, it can do everything that Asterisk can do. AAH is actually just a distribution of CentOS linux which autoinstalls itself and a bunch of Asterisk tools. Where you are limited is with its web interface (Asterisk Management Portal). AMP is good but you can’t do nearly as much with AMP as you can with some good dialplan scripts. For what you need tho, AAH will do nicely. Just remember to change your passwords
looks like thats the route that we’ll be taking then.
Any recommended phones we could use with speed dials? Busy indicator lights being a nice addition but not really needed. Cheap as possible really.
Sorry about the old PBX. But atleast now you are moving up out there.
I used the Snom 360 and it worked great. You can program the buttons with anything that you want. It is all real easy with thier web-app that comes with the phone.
if you have a bigger budget- Snom 360, aastra 9133 also looks pretty good. Aastra 480i if you can deal with the buttons being softkeys. If you need more buttons, the 360 supports a sidecar that gives you another ~20 buttons.
If you want cheap, try a Grandstream GXP-2000. It’s only about $100 per phone, supports PoE (power over ethernet 802.3af), and does most of what you’d want. Keep in mind tho with the 2000, the firmware is still a bit immature (not nearly as finished as Aastra’s 1.3 release or Snom 3.60). It doesn’t support BLF until the pre-beta verison 1.0.1.13 that was posted on voip-info.org. However they are working on things and they usually release a new firmware once every 1-2 months. 1.0.1.13 however is pretty workable, and a Wiki member is currently alpha-testing their next version.
You could also try a Sipura SPA-841 (a steal at only $80), but i think that has fewer softkeys and has no backlit display.
Also keep in mind that you can mix and match. Put cheap phones where they are rarely used (ie on the wall in the hall) and more expensive phones for people that use them alot.
Lastly, whatever you buy, buy 1-2 of them first and see if you like them / learn their quirks.
yeah either the 9133i or Grandstream GXP-2000 look like a good plan.
At the moment we are running everything from behind a NAT router. Would all external calls (ie to the internet etc) run through to asterisk machine? meaning ports would only need to be forwarded to 1 machine?
I wouldnt consider running any business calls through our DSL connection unless we had some sort of QoS control becasue with such a slow connection its so easy to saturate it.
If we were going to do this i would probably build a combined NAT/firewall/Asterisk box with some sort of QoS controls on it.
or maybe even have another DSL line especially for voip?
yes exactly, if you set canreinvite=no for your outgoing sip trunks (voip providers) then you only need to forward ports to *, which is the way it should be done anyway. You need to forward UDP/5060 and a bunch of UDP ports from rtp.conf (set it to whatever, it only needs a few hundred). You also need to set externip= and localnet= in sip.conf.
An easy cheap way to do QoS, is get a Linksys WRT54g (Version 4 or lower, NOT VERSION 5); WRT54gs, or WRT54gl. Load sveasoft alchemy firmware on it and you have a very very capable router. It can do QoS by port or by mac or by protocol, so you’re all set. Get a good Ethernet switch for the majority of the ports and you’re all set. Forward the ports on the router, setup QoS and you have a pretty capable solution. The Alchemy firmware also supports PPTP VPN, so you can link in over the Internet and make secure phone calls from a softphone.
[quote=“timfishy”]At the moment we are running everything from behind a NAT router. Would all external calls (ie to the internet etc) run through to asterisk machine? meaning ports would only need to be forwarded to 1 machine?
[/quote]
Yes - so long as it was properly configured.
[quote]If we were going to do this i would probably build a combined NAT/firewall/Asterisk box with some sort of QoS controls on it.
or maybe even have another DSL line especially for voip?
any ideas?[/quote]
That sounds like a reasonable plan. QoS isn’t really much of a worry under those circumstances - if you were only using the connnection for telephony. You would, of course, have to limit the number of concurrent calls to fit into the bandwidth.
Contention is more of a problem than QoS - i.e., as in contention ratios that your ISP works by. When the network in congested, there’s nothing you can do about it anyway. How much of an issue that is depends on things like what country you’re in and probably whether your in a rural area or not, etc… And, of course, how good your ISP is.
Today ive been looking through our phone bills and looking up call rates to see where we could save money (all our phone bills are itemised).
Most of our phone bill nowadays seems to be mobile phones (around 70%). Just on the house side of things we make £200GBP of mobile calls a year. And with the business most customers just give their mobile numbers.
Looking around at voip providers here in the UK, on mobile calls we would only actaully save at most 10%.
Then ive been looking at hooking up a GSM gateway to the asterisk box where we can get calls for as little £0.04GBP per minute. I can see this getting more and more fun
Cheers for everybodies guidance. i’ll probably come back once weve set it up and let you know how it goes.
