Considering an Asterisk installation...some questions

A business that currently has 3 physical locations is looking at moving from 3-5 external PSTN numbers and 3 separate phone systems to a single external PSTN number at the main office, and an integrated VOIP system connecting all three locations seamlessly. There are 8 phones at one location, 9 at another, and 4 at a third.

At a maximum, there would be 5-6 users of the external PSTN line at any one time.

One essential requirement is intercom over speakerphone between any two phones. From what I can tell, most of the common SIP phones do support this, so I don’t think that is a problem.

Some questions I have:

  1. Am I likely to have any stability issues deploying Asterisk in this way? Reliability is very important.
  2. Which version is most stable at the moment?
  3. Would it be more efficient to have one central * server at the main office, and connect to all the SIP phones over the WAN/LAN as needed, or have one * server at each location (total 3) with IAX over WAN between them and have each phone connected over SIP/LAN to the local * server?
  4. Whether using 1 or 3 servers, what specs would they need (roughly)?
  5. Would a 3.0 Mbps down / 1.0 Mbps up connection at each location suffice, if this was also handling email/web traffic?
  6. There are currently some feature-rich regular telephones at each desk in the main office that can do voicemail / intercom / etc. Is there any chance these existing phones could be used temporarily by connecting them via the existing internal phone lines to a * server, or would I be much better off replacing them all with SIP phones? (There’s no point keeping them if the intercom / speakerphone wouldn’t be supported). Unfortunately, I’ve forgotten what the make/model number is.
  7. Is it realistic for a person with no PBX or telephony experience, but who is very technically minded and very comfortable with Linux to set up and direct installation of such a system within a month?

Thanks for reading and answering any of these questions.

Welcome! I think Asterisk can fit your needs.

For your branch offices, the one thing that you must keep in mind is that all of them will need good Internet links with useful QoS controls to prioritize VoIP traffic. You must also make sure you have enough bandwidth for the VoIP you’ll be using and also your Internet usage.

Most IP phones support Intercom. I generally recommend SNOM and AAstra phones, both support it and work quite well. They are also easy to configure. Via the Page() app you can do some useful stuff.

  1. Asterisk can do this just fine. You might consider having small * boxes in each office with a backup PSTN line in each one- that way if any of your Internet links go down you keep in-office Intercom ability as well as some way of dialing out.

  2. I’d stick with the 1.2 branch for now, once the 1.4 branch has a few releases under it consider upgrading.

  3. see 1… Separate servers gives better redundancy. However one server will also work nicely. Just keep your UDP ports assigned right :smile:

  4. not much for that low call volume. the only high-cpu youd need is if you do G.729 recoding or lots of meetme rooms. 500mhz or better will do for each office.

  5. Yes, depending on how much web traffic and what codec you use. If you use something like gsm you will drastically reduce your BW usage. YOU NEED GOOD QOS CONTROLS ON ALL WAN LINKS THIS IS NOT OPTIONAL. Remember that :smile:

  6. The ‘feature rich telephones’ are system phones for your site PBXs and those features will only work on said PBX. You can often integrate that pbx into an * setup and use it as little more than a ‘dumb phone adapter’ but this depends on how programmable it is and what interfaces it has.
    All IP phones will be much easier to deal with but will cost more probably.

  7. If you’re a quick learner, sure. read the docs, read voip-info.org, read this forum, and ask questions here and in #asterisk when you get stuck. It should be possible, but with any system be ready for a period of working out bugs. Make sure your employer is too.
    also consider reading the book asterisk: the future of telephony. It’s available dead-tree form from O’Rilley or you can download the creativecommons licensed version from www.asteriskdocs.org.

good luck!

Thanks a lot for the quick reply! That’s exactly what I wanted to know and hoped to hear.

I’m in the process of reading through “The Future of Telephony” (it’s been excellent so far) and the wiki and so on. Asterisk looks like a fantastic technology, and I’m looking forward to giving it a try. Thanks again.

If you want to get up and running quickly I would suggest Trixbox as it’s got some very nice features and is easy to install and get up and runnign with quickly.

You could setup IAX2 trunks between the three locations and setup a dial plan so tha each location has a separate extension range. Using IAX2 means that you only need to worry about prioritising traffic on port 4569 across the internet and don’t have to worry about RTP ports. Using G.729 between the sites would reduce the bandwidth requirements and gives acceptable quality for most applications.

I have used Aastra phones quite extensively and they provide pretty good quality. I understand from others who use Snom that they may have the edge but have not done a direct comparison.

OK all sounds fine, just a little thing about money.

Many Many times I see folks look at Moving to Asterisk to save a bunch of cash. and it doing so go the cheap route…
do not follow the hype, you must spend a good bit of cash to pull this off.

Buy real Intel boxes, use real digium or like cards (no cheap x100p’s)

If you toss in a VOIP provider test them first, CHEAPER IS NOT BETTER.

And do not think for one moment that your INet has QOS, test it to be sure.

I just dealt with a setup where they where sure they had a 3 meg upload
package…well it was not even close…that best effort dsl line was under 300 k upload…they were paying for the 3 meg upload they just are not getting it.

Ping times to the Voip provider they selected based on pricing was over 500 ms.

Couldn’t agree more. In fact if you read Asterisk: TFOT it spends some time talking about using quality hardware and a good smoothed power supply! Voice is a very sensitive application and even poor power can affect the overall quality so always go through a power smoothing UPS.

If have picked up some old Compaq Proliant DL360 servers on ebay that have worked very well for Asterisk as their build quality is pretty good.