Fri 7 Jul 2006
Jumping Onto the VoIP Bandwagon
Posted by Andrew Mitry under VoIP/Telecom
[10] Comments
I spent several years working in telecom, at Winstar and for small vendors installing traditional hybrid PBX phone systems. I installed mainly Nortel systems, but we did service Avaya (Lucent), Toshiba, Panasonic and NEC on occasion. I originally installed a donated Nortel Norstar 824 with Startalk B Voicemail for the church back around 2000. Over the years we kept expanding it the system, but were limited by the two ports on the Startalk. I looked into upgrading the voicemail and it would be several thousand dollars and we would still be limited on features. The other issue with the Nortel is that programming was complex and it was limited and what it could do.
I began to look into newer hybrid PBX systems, but they were very pricey. At that time we had 8 analog lines and 20 phones, we were looking at $8,000 plus. Not only, was the initial cost high, but we would have to buy proprietary phones, and add expensive cards as we grow. It was around the same time that Asterisk started to gain traction. I played with it for over a year before I felt it was ready for production, my main concern was finding a good VoIP phone at a reasonable price. Late last year Linksys began to roll out good quality VoIP phones at a very reasonable price. Also, the release of Asterisk@Home, now called TrixBox, made installation and administration of Asterisk much easier.
We purchased 20 Linksys SPA-941 Two-Line VoIP Phones, a Polycom SoundStation IP 4000 VoIP Speaker Phone, and a Digium Wildcard TE205P Dual T1 card ( I would recommend getting the newer 207P because it comes with hardware echo cancellation) for less than $5,000 from VoIP Supply. I had an old Adit 600 channel bank sitting in my basement, remnants from my Winstar days, which turned out to be perfect for setting up an additional 24 analog extensions (hence the need for a dual T1 card). We had an extra Sony Vaio P4 2.8 HT 1 GB RAM and 120 GB Hard Disk that we installed the T1 card and setup Asterisk@Home 2.7 on. Ward Mundy over at Nerd Vittles has a great walk through: Newbie’s Guide to Asterisk@Home 2.7: Unabridged Installation and Upgrade Guide. The only thing we had to do different was the configuration for the T1 card and channel bank. We configured the VoIP phones one by one through their web interface, we are researching how to get the automatic provisioning via tftp working. Because our ISDN PRI wasn’t installed yet, we ran on VoIP trunks from Teliax until Cox cut us over.
To ensure good voice quality on the internal lan, we put all the VoIP phones on a separate VLAN, for those without managed switches, this could be accomplished by using a separate switch, isolated from your data network. Note, for those who have heard about sound quality issues with VoIP, this becomes a moot point when you can guarantee quality of service, such as on an internal LAN. The only potential for sound quality issues comes in when you are running calls over the Internet. For business quality service, I recommend routing service over the PSTN for now, with VoIP trunked over the Internet as a backup or secondary option (dial a prefix to route long distance calls over VoIP).
We have now been running the system for over two months on the ISDN PRI without any problems, thank God. We are enjoying the benefits of an enterprise class phone system, voicemail/fax to email, remote extensions and soft clients. We plan on adding at least 20 more phones this summer and setup paging so that school principal can call all the naughty kids to the office.
Recently, we also setup a smaller system for Coptic Orphans, they are running 10 Linksys SPA-941 VoIP phones and 4 analog lines using the Sangoma A200 PCI Card 4 FXO Ports + Echo Cancellation, costing just under $2000, not including the server (another Sony Vaio). This time we kept the fax outside the system to keep things simpler. Installation was fairly straightforward, make sure to follow the driver instructions on the Sangoma card closely, as that we got hung up when we missed one seemingly minor step.
Generally, ISDN PRI is used when you exceed 8 analog lines of service, it works out to be the same price or cheaper and you get the benefits of direct inward dialing (DID), caller ID, faster call setup, and 23 channels available for inbound/outbound calling. Upfront cost is a little higher due to the more expensive cards.
In the past couple months, Linksys had released more VoIP phone models which increase flexibility in planning out a system. If you are in the market for a new phone system, I would seriously consider a similar configuration.
VoIP PBX Links:
Trixbox – Latest version of Asterisk@Home.
Trixbox Asterisk-based PBX virtual machine
Asterisk Open Source PBX - Asterisk is the core PBX powering Trixbox.
Digium – Original Developers of Asterisk, they sell/support PSTN cards and an enterprise grade version of Asterisk.
Linksys – VoIP Phones and analog terminal adapters.
Nerd Vittles – Great guides on installing, configuring and tweaking Trixbox.
Voip Supply – We have been working with Brian Dooley, so far they have been great with us but I did see this post go up yesterday VoIPSupply.com disappoints, again.
Atacomm – Another VoIP equipment vendor.
VoIP Wiki – The best resource for all things VoIP, including driver configs for cards and sample setups.
Fonality – Preconfigured Asterisk VoIP servers, for those who don’t want to get their hands dirty.
Teliax – VoIP Trunking, they allow multiple simultaneous inbound and outbound calls on the same VoIP lines, but they do charge 2 cents a minute for incoming and outgoing traffic. Useful for testing and as a backup service provider.
CounterPath – X-Lite Free is a good soft client but you have to upgrade to transfer calls.
SJ Labs – SJphone is a full featured free soft phone for Windows
10 Responses to “ Jumping Onto the VoIP Bandwagon ”
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March 19th, 2007 at 1:40 pm[...] For those of you who are interested in an open source VoIP PBX but don’t have the time or desire to build your own Asterisk box, Fonality is releasing the Trixbox Appliance. Trixbox is the new name for Asterisk@Home, the platform we currently run at church. Asterisk@Home 2.7 is working well for us, although we do look forward to upgrading soon so that we can take advantage of the new features in trixbox. [...]
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March 28th, 2007 at 2:02 pm[...] Per Matt Singley’s recommendation, I decided to give SpinVox a try. I don’t use voicemail on my cell phone so I decided to run it on my office line. Setting it up with our Asterisk PBX was pretty straightforward. [...]


July 10th, 2006 at 8:05 pm
What, if I may ask, caused you to go with the Linksys 941’s over the 942’s? Was it just the $1k you saved on the 20 stations? Did you just stick with the 2 lines, or did you go for the 4-line upgrade? Not interested in PoE, I guess, as well..
Our church is considering a move to VoIP, and I’m the geek in charge of the investigation. Thus far, I’ve installed in my home, a Trixbox (running in a vm under vmware server on my home server), connected to a handful of SIP/IAX trunks (Gizmo, FWD, Voipdiscount, Voipjet, and a SPA-3000 for PSTN connectivity). My desk phone is a Linksys SPA-942, I’ve got a cordless phone connected to the FXS port of the SPA-3000 and I’ve got my Nokia E61 talking to the PBX over WiFi. Go ahead, call me a geek.
If you’re doing multiple lines for the users, how are you handling extension assignments (200/1200 or 200/201, etc?).
Further, as I am not skilled in the ways of the PRI, am I correct in thinking that you’ve got 23 channels per PRI, which are not directly mapped to any particular DID? In other words, a call to DID1 may come in on channel0, but the next call to DID1 may come in on channel7? If so, does it typically cost extra to get additional DIDs assigned to a PRI, i.e. you get a single # associated with the PRI, then add extra DID #’s as an add-on? Am I making sense?
Ok, that’s a lot of questions to throw your way.. Thanks and God bless!
July 10th, 2006 at 8:31 pm
We actually purchased the Linksys 941’s just as they began shipping, the 942’s were not even announced at the time (there were some rumors floating around, but nothing confirmed). As we add I do plan on purchasing the 922’s or 942’s, I am especially interested in the integrated switch as that it supposedly supports VLan tagging. I stuck with the 2-lines, no need for the 4-line upgrade in our configuration, each user can enable call waiting on their extension if they so choose and can dial out a second call on the same extension. The only time I configured the second extension is if two people were sharing the same phone, I would give them seperate extensions. As for POE, none of our current switches support it and POE injectors with multiple ports are pretty expensive. I am considering purchasing a POE switch that runs -48 DC to put on our telecom rack (it has an 8-hour backup, Cox installed it when they put in the fiber that our Data and PRI come across). I would then strategically place the newer phones I buy in areas that would be important to have a working phone during a power outage.
How is the Nokia E61? I have been eyeing it for a while, but we are tied to Sprint for at least another year. I have tested the Linksys WIP300 and it works great, we are going to roll it out as a”cordless” phone that covers the whole facility once we finish installing all the wireless access points.
You are correct about the DIDs coming down the channels on the PRI, there are 23 channels for voice and one for data (transmits caller ID and call setup info). Usually if you are recieiving incoming calls you want to get at least one block of 20 DIDs that will include your main number. A block of 20 DIDs typically costs between $10 and $20 a month extra (I think we are paying $10, but I need to double-check).
I hope I answered most of your questions. God bless you too!
July 11th, 2006 at 12:25 pm
Thanks for the answers!
I absolutely love my E61. Voice quality can be choppy from time to time, but that could be my lousy AP, but that aside, it works great. From time to time, it’s nice to hit the cfwd button on my desk phone and roam around the house.. I also get my email on it via BlackBerry Connect, so it’s good…
July 11th, 2006 at 3:37 pm
Very good article! I too come from a past, installing and maintaining digital key systems(Toshiba mostly), and Asterisk @ Home(pre trixbox) was a great opportunity for us to take advantage of. We use Polycom IP501’s and some IP 4000’s. Couple with the meetme features, and other great features, we’ve been nothing less than ecstatic. One thing I can definitely attest to is having the voip network on it’s own vlan or network segment. We chose that route after noticing some echo on our side, but not at the far clients side. We use a single t1 card w/ a full t1, and 80 dids from Qwest without a glitch.
July 11th, 2006 at 5:09 pm
Jason, you are welcome for the answers, keep us posted on how the move to VoIP goes. Let me know if you have any other questions.
Rolf, which T1 card are you using, the Digium or the Sangoma?
July 11th, 2006 at 9:44 pm
I’m using the Digium 110P. We have it in a HP DL360 with 4GB of RAM, dual 2.8G procs, and two 80G SATA drives. The size of the server is overkill, but we’re not sure how quickly we’ll grow, so we wanted to be safe
.
July 12th, 2006 at 11:51 pm
Great post!!!
If we had understood phone stuff better we would have looked more closely at Asterisk last fall.
July 13th, 2006 at 12:32 am
You may want to still get a closer look at Asterisk, it can integrate with your existing PBX and provide a lot of additonal functionality at a fraction of the cost (I know those traditional systems charge a pretty penny for every feature upgrade or addon). Check out the Trixbox Virtual Appliance for VMware Server.