Hello,
Please forgive me if something like this has already been answered, but I just need some clarification.
Some background: We are a medium sized business with 50 (expanding to 100+) employees. We have two T1’s coming in to our office, one for data and one for voice. (We’re not locked in to this configuration. Our T1’s can be reconfigured as necessary to implement a VOIP solution) We are looking at upgrading our current Digital phone system to a VOIP solution and would like to use Asterisk.
For hardware configuration, would we configure as follows?
T1 plugs directly into Asterisk server via T1 Card.
Asterisk server connects into Ethernet 10/100 switch.
Phones connect to jacks that are wired back to connect into switch from above.
Am I missing anything there? Also, would a voice T1 be used for Asterisk or a data T1? I have started reading the documentation, but much of what I have found has been geared toward either the home user or setups with regular POTS lines. Any assistance/info is greatly appreciated.
Thanks in advance!
Michael
yes, that’s pretty much it. i’m assuming the phones you are planning on using are VOIP compatible - they’ll need to be if you intend to plug them into a network jack. if you want to use standard analog phones, you will need some sort of gateway between them and the asterisk box - something like a wildcard or channel bank.
otherwise, that’s pretty much exactly how we’re set up, for almost 400 users (spread across multiple servers, obviously).
you probably could get by with one fairly decent server if you only have one T1 for voice. you shouldn’t need any reconfig on the voice T1, you’ll basically plug it into your asterisk box. the signal will be converted by asterisk and the hardware to travel over your existing data network to the individual phones.
Hello,
Thanks so much for the reply. We will be purchasing a slew of new VOIP compatible phones. My main confusion was regarding when a channel bank was needed. So I can plug in my voice T1 to Asterisk directly and it can use the T1 as-is for incoming/outgoing calls? That would save me much headache if so. Then I’d just need to start shopping for the card.
Thanks again!
Michael
Hello again,
One other question I’ve got:
I’ll need to do a “test setup” where I’ll get a few IP phones, connect them to an Asterisk server via a switch and connect the Asterisk server to either a single POTS line, or a single line extension through my current phone system. Any recommendations on the card I should get to do this?
Thanks!
Michael
That’s correct, you should be able to plug your T1 directly into a Digium or Sangoma card in your asterisk server…unless you have some weird configuration, it should be plug and play pretty much.
As far as a test system, you could do a couple of things - you could buy a TDM400P with one FXO module to interface to the PSTN (i always get FXO/FXS mixed up, but i’m pretty sure you need an FXO port to go out to the PSTN…somebody confirm?). Regardless, you’d need this card and a couple of SIP compatible phones to get a small test lab up. I would recommend this as it’s a great way to learn some of the neat features of asterisk, as well as get good as setting up dial plans, etc.
My first build was a live server with 4 T1’s and 50 users - I wouldn’t wish that on anyone…
If you don’t have a copy, go buy Asterisk - the Future of Telephony and read it cover to cover - it’s a great primer and should give you a good base from which to start. Then learn to love voip-info (which i’m sure you already do) and you should be set.
I am in the middle of implementing asterisk with about 75 extensions and a single incoming T1 line. We are using a Dell PE2850 (2 3.2GHz procs and 2GB Ram). I would suggest using the Digium hardware as it seems to work very well and it obvoisly is well supported for use with asterisk. For my pilot test I just used an desktop and put in a TDM400P with one FXS and one FXO card so I could test both. For the real server I bought a TE210P. The TE210P has 2 T1 ports. We have one T1 and we are going to hook up a Rhino channel bank to the other for about 18 analog extensions that we have.
FXO for the PSTN One other way you could do it is with some thing like the Sipura 3000 wich has an fxo port but I have never used one so I cant say if they are anygood.
I am running the onboard GigE port (I disabled one of them) and I did see you post about the stability issues. Once every thing is biult I was going to see if I run into the same issues as you did. I am thinking now I might as well just get a card now and disable to second onboard NIC. How has your stability been since you swithed to the seperate card?
We’re up to 16 days without issue on one of the boxes that was having difficulty…that’s the only machine that I’ve put a third-party NIC in. It seems to have helped our dropped calls and overall sound quality as well, although part of that might be the weather warming up (we have a bad static issue in our building, and softphones apparently don’t like static discharge).
If we finish out this week without any problems, i’ll probably try to upgrade the other 7 servers and see where we’re at.
Do yourself a favor if you decide to put a card in - find one that is on the HAL for your flavor of linux. We’re using a non-supported Linksys card with Fedora and it was a PITA to get it to load properly, where the netgear card would have loaded automatically…
