Help Deciding On Asterisk Configuration For New PBX?

Asterisk Asterisk Support
1

First of all, this is my first post, and if I’ve posted in the wrong place, please let me know where would be better and I’ll re-post.

My small company is currently growing, has about 15 employees, and we expect to be hiring more soon. We are a software development company that writes some fairly sophisticated software, so our developers should be competent with Asterisk configuration. We need to be cost conscious, and we need to grow from our two incoming lines (one POTS and other Vonage) that are connected to a cordless handset system that allows for us to move a few phones around and share those lines to a phone system that allows for incoming calls to be answered by our receptionist (or an auto-attendant if she’s not there or busy) and transferred to that individual’s phone. We then need the ability for those people to pick up their phone, get dial tone and call out. In fact, if we could have the employees answer through their computers, that would be fine
[ul]
[li]If we install Astrisk on a machine (TrixBox?) and we need three PSTN lines (or Vonage through RJ-11 analog interface) and we need two analog phone lines in house for our two existing Polycom analog SoundStation2 units, what would you recommend buying as a hardware interface?[/li]
[li]If you have a couple of 4 line PSTN boxes (PCI Cards?) that connect to the computer, can you assign any of those jacks as either an extension or trunk?[/li]
[li]If I wanted to use one line that had different rules for dialing out (i.e. one line needs a 9 to be dialed out because it’s connected to a university network and also needs a long distance code to be used if we called out on it, and a vonage line doesn’t need any of that), could I set those to automatically handle those functions if a user picks up? In this way, a user picks up the phone and dials in a consistent way, regardless of the phone line that the PBX chooses for the outgoing call.[/li]
[li]If I wanted to use Vonage for some lines, is there a way to omit the conversion from their boxes (Linksys) to analog and then out to a PSTN interface to the Asterisk system?[/li]
[li]Would anyone recommend a different service that duplicates what we’re trying to do with Vonage (make regular outgoing calls) that doesn’t require the PSTN conversion until it reaches the phone company that converts it to analog cabling. i.e. we’d love to use Skype out if that were possible.[/li]
[li]Do you have any recommendations on Mac/PC software our people could use to omit buying phones? I understand there’s no Skype integration yet, but I could imagine being fine with redirecting a call coming in on a phone line over the computer to a software phone that the employee picks up with a headset.[/li]
[li]Would it be cheaper to buy phone modules to be used for each desk and use analog phone connections and buy analog phones, or buy IP Phones?[/li]
[li]When using an IP phone system within the office, would you connect those phones to the same Ethernet network as you have for the rest of the office, or is it better to have those phones on their own network so you don’t take up too much traffic?[/li]
[li]We have four employees who are remote to us, could we set up a local extension here, so when we dial an extension (i.e. 23) the system automatically calls that employee’s analog phone line?[/li][/ul]
I have read the NerdVittles Intro, but it leaves off at what to do with the hardware for the phones themselves. I have been unable to find other resources to help me with these topics, but if you know of places to point me, I’d be happy to read through other information. Thank you so much for your help in trying to figure out an effective way for us to get set up!

2

you don’t necessarily need to connect via a POTS interface - if you were to consider using an ITSP, you would route calls to the PSTN over IP, to a remote location that would terminate your lines. it scales better and is cheaper out of the box (you don’t need any special cards, just a network connection).

as for connecting your internal phones to asterisk, you should look at an ATA, or analog telephone adapter - it has one or more RJ11 ports and one or more RJ45 ports - plug into your analog phone, the internal circuitry translates that signal to SIP/RTP, and sends voice traffic over your network to the asterisk box. cheap, very flexible, and scalable - if you need more ports, buy more ATAs. you might also consider a channel bank depending on how many lines you will need…a channel bank is nothing more than a really big ATA.

your dialplan is extremely flexible, and can handle any dialing rules you would like - trust me when i say this is asterisk’s best feature - if you can dream it, asterisk can probably do it.

as for softphones, i would start with xlite (SIP) and idefisk (iax) as both are free and work very well.

if you haven’t already, go get a copy of Asterisk - The Future of Telephony - it will give you a very good background on alot of the issues you are going to have to face, and should help get you started very easily…

good luck.

3

  1. This would be 3 FXO ports and 2 FXS ports. You could get a Digium TDM03B and TDM20B. Another option is the TDM2400P which will scale from 4 ports to 24.

  2. I believe you are talking about connect a POTS line or a phone. If so when you purchase a TDM card you can swap the modules around to fit your need.

  3. You can setup dialing ‘9’ to go out one port and then you can setup long distance to go out another port. Hope this clarifies it for you.

  4. I know people have connected their Asterisk box directly with Vonage in the past but I’m not sure if you are still able to or not.

  5. There are many providers now that you can do this with and not use an adaptor. Nufone, Voicepulse, Teliax, Broadvoice… the list goes on and on.

  6. One of the better softphones would be Eyebeam also known as x-lite(free version). www.xten.com

  7. You will have to weight this with what analog hardware would be needed. Two TDM cards would run somewhere around 600-800. Plus your analog phones you want, probably $80 each for decent ones. If you get a decent IP phone they run about $200-$250 each.

  8. In small offices it shouldn’t be a problem on the same network. Many do this without a problem.

  9. Yes

Hope this helps!

4

Thank you both so much for your help. I have a few follow-up questions if you’ll indulge me:
[ol]
[li]I’ve seen the list of ITSPs around that work with Asterisk, but could anyone recommend one?[/li][li]For Asterisk interface cards, I like the idea of the single card that is expandable with modules, and I see that one option is echo cancellation (doesn’t seem to be available on the individual cards), would you recommend to go for echo cancellation or not? I read the FAQ on echo cancellation, but I wasn’t quite sure whether it would be important for my application to external PSTN lines (or ITSP) and internal PSTN lines.[/li][li]I’ve spec’d out the most basic Dell server for about $550 (Pentium D 2.8 Dual Core, 1GB RAM, Dual 80GB SATA HDs (would use as RAID 1) and install Linux). Does this seem appropriate?[/li][li]I’ve seen TrixBox, and like the quick startup, but we do have engineers in house that know their stuff. Would you recommend starting with TrixBox, starting with something else, or going to the source?[/li][/ol]
Thanks again, so much!!

5

Trixbox is a bundle of many apps and add on’s
It has a ton of stuff most folks will never use, that being said,

Trixbox has a VMWare machine you can download and use the free server or player from vmware.com to fire it up. in less than an hour you can have a IP PBX up and running and use softphones to make calls.

But if you really want a stable, clean system then do the centos 4.x / asterisk 1.2.xx anything that TrixBox does that you like can be reproduced on a raw asterisk system with a little work (the stuff is very easy to do MYSQL and PHP with a little perl to top it off.

Edit:
Oh yea; if you want to try the vmware thing here is a custom Trixbox with some added things created by nerdvittles.com/index.php?p=144 and tweaked by the great folks at vmwarez.com

it has sip extens created and some cool feature not found else where
pbx.mississippi.com/

6

First of all, this is my first post, and if
I’ve posted in the wrong place, please let
me know where would be better and I’ll re-post.

Good morning, schalliol

This place is fine. :smile:

I can’t answer all of your questions, and I am very much still a novice at Asterisk, but let me give it a shot from my past few weeks’ experiences and see if I can help.

  • If we install Astrisk on a machine (TrixBox?)
    and we need three PSTN lines (or Vonage through
    RJ-11 analog interface) and we need two analog
    phone lines in house for our two existing
    Polycom analog SoundStation2 units, what would
    you recommend buying as a hardware interface?

Others here have suggested Digium cards, either TDM400 series (2 cards needed, for your total of five ports) or the TDM2400 series (expandable to 24 ports). I have the Digium TDM400P with one FXS (into which you plug an analog phone) and three FXO (that you plug into the wall) ports. I find that the only limitation so far is that when you make an outgoing call, Asterisk is not able to detect when the call is answered at the other end. You would need this if, for example, you auto-forward someone’s call to his cell phone and you want Asterisk to go to voicemail if the person doesn’t answer his cell phone. Digium cards apparently do not provide the ability to do this, but I’m told that Sangoma cards, e.g. the A200, do … IF your telco provides the correct signalling to allow you to detect when the call is answered (some telcos do, some do not, and with some it is an extra-cost option).

  • If you have a couple of 4 line PSTN boxes (PCI
    Cards?) that connect to the computer, can you
    assign any of those jacks as either an
    extension or trunk?

If I understand your question correctly, the answer is yes… you need to configure your PSTN cards as FXS ports to which you connect a phone, or FXO ports that you plug into the telco phone line’s wall jack. The Digium TDM400 series lets you order the card any way you want (ours here is one FXS and three FXO), and if I recall correctly the TDM2400 lets you configure the available 24 ports in groups of four.

  • Would anyone recommend a different service that
    duplicates what we’re trying to do with Vonage
    (make regular outgoing calls) that doesn’t
    require the PSTN conversion until it reaches the
    phone company that converts it to analog cabling.
    i.e. we’d love to use Skype out if that were possible.

I have come up with a way to interface a SINGLE Skype connection to a SINGLE analog phone line in the PBX, so that incoming Skype calls are routed to our auto-attendant and can be routed with touch-tone button presses just like any other call. I use a D-Link DPH-50U Skype-to-analog-phone adapter module (intended to allow regular analog phones to be used with Skype). It connects to my Skype computer via USB, and then has a “phone” jack into which you plug the phone, and a “line” jack that I connected to the FSX port on Asterisk. Incoming Skype callers hear another short dial tone and then two DTMF tones, then get our auto-attendant.

If you have Skype up and running there, call Skype user “BNPConsulting” if you’d like to hear how it works.

This should also work in making outgoing Skype calls, which is what you want; the D-Link module is designed so that you can call it from your cell phone, from wherever you are, and punch in a Skype “speed dial” number and make outgoing Skype calls including SkypeOut calls. I have not been able to get it to work that way, yet. I’m sure it is some boneheaded misconfiguration I have done to Asterisk, and I will get to it when I can. In our case the most important mode is to receive incoming Skype calls and route them within the company, and in that mode it works great.

  • Do you have any recommendations on Mac/PC
    software our people could use to omit buying
    phones?

Others here have recommended X-Lite from CounterPath (www.counterpath.com). I have evaluated it and it works fine. I have used it with a headset, and have also used it with the VoipVoice Cyberphone that Radio Shack sells for $30 as a “Skype phone”. It works fine either way.

Note, however, that the free X-Lite licensing terms don’t allow it to be used for business or commercial uses, so technically you’d have to buy their commercial EyeBeam product if you want to use it within your business.

  • Would it be cheaper to buy phone modules to
    be used for each desk and use analog phone
    connections and buy analog phones, or buy
    IP Phones?

I think it would be cheaper to buy IP phones. Another reply to your post here said that IP phones would run you around $200 to $250 each. Others, here and elsewhere, have characterized the low-priced Grandstream phones as “cheap toys”. I don’t really agree. I purchased a Grandstream BudgeTone 200 phone for $65 for evaluation, and received it a couple of days ago, and it seems fine to me as long as you don’t try to use the handset as a hammer. It’s at least as good as many of the phones I’ve used in the dozens of companies I’ve worked at in 36 years of contracting and consulting. For a basic desktop phone without a lot of bells and whistles, it should work fine.

I would suggest that you buy one Grandstream phone and evaluate it for yourself. Then if you decide to go with something more expensive, you can still use your evaluation-model Grandstream phone as one of your lesser-used extensions (in the print room, for example) … there’s absolutely nothing that says all the phones in a system have to be the same model or even the same technology.

You asked another question about using the house Ethernet network to connect your phones to your PBX, or install a separate network. With 15 employees, how many calls do you think your system is going to handle simultaneously, on average? 3? 5? 8? I have no personal experience on scaling a system like that but I can’t quite imagine you having network congestion trouble, especially if your house network is gigabit. I would suggest you try it on the house network and if it gives you any problems, then’s the time to consider installing a separate Ethernet network.

  • We have four employees who are remote to us,
    could we set up a local extension here, so
    when we dial an extension (i.e. 23) the system
    automatically calls that employee’s analog
    phone line?

Yes, you can do that and it works well. An outside call for that employee would come in one of your outside phone lines and go back out (to your remote employee) via another outside phone line, and Asterisk would tie the two lines together. We do that here, quite often, and it works well.

Better than that, though … if your remote employees have broadband Internet connections (e.g. DSL or cable), they can install a softphone like EyeBeam / XLite and use a headset, or else you can buy them the same IP phones you use at your office and they can connect directly to the PBX from their homes or remote offices. Incoming calls for them would be routed to their remote locations just like they were sitting at a desk next to the PBX. That way, if your remote employees are a long-distance call away from you, that avoids telephone toll charges, and also saves you from tying up one of your outside phone lines.

I have been unable to find other resources to
help me with these topics, but if you know of
places to point me, I’d be happy to read through
other information.

This forum is an absolute gold mine of information. :smile:

There is also a VoIP wiki at www.voip-info.org that is another gold mine of information on VoIP in general and Asterisk in particular.

Finally, you should get yourself a copy of the book “Asterisk - The Future Of Telephony”. It is an O’Reilly book that you can buy in print form AND it is also available for free as a PDF download (sorry that I don’t remember where to download it from but that information should be easy to find by searching either this forum or www.voip-info.org).

One other thing… There have been mentions in this thread about Trixbox. I have absolutely zero experience with Trixbox or Asterisk@Home, both of which are advertised to be sort of out-of-the-box, plug-and-play Asterisk implementations that require minimal configuring. I did it the hard way… installed Fedora Core 5 Linux on a a computer and downloaded and installed the sources to Asterisk and the various drivers … and in about four days’ time I went from being barely able to spell Asterisk to having a working PBX with most of the features I needed, all by using the “Future of Telephony” book and a lot of help from the fine people on this forum. It is really not hard, and in the end I think you will know a lot more about Asterisk and its capabilities than if you installed a plug-and-play solution like Trixbox or Asterisk@Home. But, that is just my opinion, without benefit of any exposure to Trix or A@H other than what I’ve read in places like here.

A final thought… someone here mentioned a VMWare Asterisk solution that runs on VMWare’s free Player. With that, you would be able to run Asterisk in a Linux “virtual machine” running on your Windows computer. I have no experience with that but I have been a VMWare user since their early beta versions and I can’t quite imagine an Asterisk system like that working well with any significant level of loading. VMWare takes up a huge chunk of PC resources and runs too slowly for many resource-intensive applications. For example, currently I run Fedora Core 3 on VMWare on Windows XP on a 2.5-GHz processor, and with many applications Fedora runs at speeds comparable to an old 200-MHz Pentium computer I had.

On the other hand, VMWare seems to run faster with processor-intensive applications than with I/O-intensive (e.g. disk-intensive) applications, so maybe Asterisk would run OK in that environment. Still, computers are cheap and Asterisk is free, so I’d suggest bypassing the VMWare route and installing Asterisk on a dedicated Linux box.

I hope this has been helpful…