Hey everyone. I’ve been reading through *@home handbook, but I can’t find the answer to my specific question. Right now I’m administering a small business with around 24 users. We’re in the process of growing, and I’m going to switch our current voice/data situation from a T-1 to a dedicated PRI for voice, and a 2Mb line for data. Part of this switch is going to require configuring a new phone server, because we’re currently running Televantage 3.5 and we can’t secure any more licenses for it. However, upgrading to Televantage 7 is going to cost me well over $10K, and now that I’ve discovered Asterisk, I can’t help but wonder if that can be avoided.
So here are my questions. First, can Asterisk@Home handle as many as 40 voice users and 23 voice lines, or should I be looking at another iteration of Asterisk? I am relatively unfamiliar with Linux, and while I could probably get Asterisk up and running on a Linux box from scratch, it would be less than ideal, hence my preference for *@home. Second, can it do this with analog telephony? Right now our office is wired with CAT3 and we have a bunch of analog phones that everyone here is used to–if I can avoid laying down extra CAT5 and purchasing new VoIP phones, it would be phenomenal. I already have some Intel Dialogic boards for our analog stations, and I’m prepping to buy a PRI card anyhow, so if I could get a PRI card that Asterisk supports (does it support any?), and still use my Dialogic boards for the phones, that would be ideal.
Am I making any sense? Ideally I want to bring in a PRI, hook it into a server that’s running Asterisk, and then run out analog lines to around 35 stations. Is this is a possibility? If not, how close to that can I get? Could I get by with internal VoIP, but still without resorting to a VoIP provider? (VoIP service is much more expensive than a PRI in my area.)
Yes you can bring in a PRI with any of the Digium T1 cards. If you have 35 analog phones you still want to use you can use a 2 channel banks. As far as using your Dialogic station cards, there isn’t really any support for them yet. Or you can just use voip phones internally.
I’m not going to offer any advice except to say “Go for it.” And just be careful that you test the setup thoroughly first. What I’ve seen thus far is that * will do most if not all of what you’re attempting but handling 40 users over one box means you’ll need some real speed in that box.
I use * 126.96.36.199 not @Home. Tried it but didn’t like it…just my opinion though. I’ll be watching to see how you progress with this so keep us posted.
BTW - I setup a small office (5 users) with * but we bought new IP phones for everyone. The internal stuff works good and their VOIP provider was working well. I know they want to expand this soon so I’m anxious to hear how your setup worksout.
we actually migrated FROM televantage to asterisk, starting about 9 months ago. we figure we have saved around $750,000 in licensing costs alone, and we have a system that is MUCH easier to configure and modify.
as far as load is concerned, we use dell 2850’s, and have 8 servers total. each server has 40-60 users on it, and our simultaneous call average is around 15-30 calls per server, more during peak times.
we’re not even CLOSE to hitting the limit of what the servers can handle. i would assume that, with no transcoding, we could probably max out a quad span T1 card without too much trouble. that’s 92 concurrent calls (sip-to-zap - we don’t have much internal traffic).
get a test box built, learn the dial plans, and see about doing some load testing with SIPp (search the wiki for it) - that will tell you how many concurrent calls you can handle (or at least give you an approximation) on any given hardware.
as far as PRI cards go, both digium and sangoma make interface cards - i’m partial to sangoma, but digium seems to work fairly well if you have a stable server…we’ve had a few issues with the dells being instable, but that seems to be related to the onboard NIC.
hope this helps.
good luck and have fun!
Wow, you guys are amazing! I wasn’t planning on getting any responses today, much less three informative replies!
It’s really heartening to hear that any of the Digium cards will bring in a PRI. I’m still confused on getting to analog stations, though–angler, what did you mean by 2 channel banks?
Also, assuming I decide to go with a VoIP (internal) solution, how much would the associated hardware cost me? How do I actually connect the phones to my server–is there a separate card I need to buy, or is it as simple as a second NIC and a managed switch? How much would a typical VoIP phone cost me?
Thanks again for all of your help.
internal voip - here are a few options.
switch to voip hardphones, which would use your existing network and replace your existing phones. cost is between $100 and $500 per set, with the average around $250.
use a channel bank/ATAs to convert existing analog station lines to VOIP. less overall cost, probably the best of both worlds, but you may not be able to use all features in asterisk. a channel bank is simply a bank of analog ports that connects to your asterisk box (correct me if i’m wrong here, somebody - that was always my understanding of a channel bank)
softphones - SOFTware PHONES - cheap (xlite is free), easy to roll out if you already have a network and computers, but you’re tethered to the computer via a headset. however, we have a 400 seat call center, and 90% of the users here are on softphones and we’ve had very good luck with them.
as far as connecting the above to your server, options 1 and 3 would use the existing NIC in the server - no additional hardware required. option 2 depends - i know there are USB based channel banks, which wouldn’t require any special hardware, but others might. there are also individual ATAs (analog telephone adapters) that are about the size of a cigarette pack that convert the analog signal to digital (VOIP) and have between 1 and 4 lines…so you’d need multiples of this, but they plug into your existing network as well.
my vote would be VOIP hardphones, but if you’re on a budget, the ATAs are nice - they’re more work to configure, but they allow you quite a bit of flexibility.
for a proof of concept, use softphones - they’re perfect for testing and the like.
hope this helps.
another option- a high density card like a digium 2400 series. Puts up to 24 pots channels from a full length PCI card with an amphenol connector that will (if you are wired right) plug straight into a terminal block. for a fully loaded 24 port fxs card, that puts you at about $68/port ($79/port for the one with echo cancel onboard)
ironhelix is absolutely right, although we ended up having to put our wildcard (the digium tdm2400) in it’s own box, as it would not load properly with the TE410P we were running as well. these are especially nice if you already have patch panels installed and can just connect asterisk via an amphenol cable.
yet another option, there is a company, xorcom i think, that makes a series of usb2.0 channel banks… i haven’t heard much about them but the few things I have heard are pretty good.
Not sure about pricing tho.
AAH 2.8 is OK but if you only want the PBX
(not the ARI and the FreePBX GUI) just do Raw Asterisk.
The ARI from Little John is very cool… webased Voice mail, call monitored recordings, if you record all inbound / outbound calls you can login and listen form anywhere, any time.
freepbx.org/trac (Rob Thomas the FreePBX guy does a $ 250.00 install, worth the money to get it right)
stay away from TrixBox very much a BETA (I want everything installed from source for easy updating, TrixBox does some funky stuff with rpm)
Just a couple of comments/suggestions/experiences
Do not use asterisk at home… it’s far to complicated in my opinion. Just managing the static config files is easy for up to a hundred extensions. You will also learn a lot more and solve problems more quickly.
I just migrated a system of 40 users/stations from a two-server asterisk at home with a Cisco MC3800 setting to a one-server static config. Hardware is a P4@2.6 Ghz W/256 Meg of ram, 1 Sangoma A104D card w/ hardware echo cancellation, 1 Digium 2port T1/E1 card. We often have 30 simultaneous calls running and load average rarely gets above 0.35. I have a local Argentina PRI E1 coming in on the Sangoma and two junky old channel banks on the digium right now- one Adtran and one CAC2. The system runs flawlessly. We have about 80 US DIDs, 40 from Argentina, 1 from New Zealand, and three carriers we use for inbound and outbound. Channel banks are definitely the cheapest per-seat solution, except for perhaps softphones. The system is integrated on the lan to do CallID winpopups. The reps are very pleased with the improvements, and I smell like a rose with management. It is legal her to do call monitoring and so our QC loves the zapscan applicaiton.
Total cost in hardware for the above system was less than US$4500. My time in building and configuring it was about 80 hours.
One last thing to mention- I’m partial to Centos 4.3.
Just adding my experience.
I started building Asterisk system on Fedora 5.0 (one of the forum friend helped in the configuration). I am also planning to move this setup to 40 to 50 users. Below is my configuration
Server: Dell PE 2300/600MHz/Dual Processor/1GB RAM/32GB HDD
Operating System: Fedora 5.0
Add-on Cards: Digium TE205P (my PRI Lines hooked up to this card)
Phones: Cisco 7940/Cisco 7960 (SIP Converted. Working Fine)
I still need to work on how to integrate my existing VG248 module to Asterisk server for Analog Lines. Also, looking at TDM2400P model to get 24 fxs channel for my fax systems.
Hope this information helps.