Asterisk for a 4 line Voip system

Asterisk Asterisk Support
1

I found out about Asterisk from the Systm episode and ever since I have been itching to give it a try.

Recently my parents small home publishing company has been growing and they converted our basement into an office. The growth has been good but not in the eyes of the long distance phone companies who have one by one been yanking them off unlimited onto pay as you go plans. They cannot afford the kinds of plans these companies insist they use and Vonage service is so low quality that customers complain about it whenever they use that. So my solution was to start going ahead and building a low end pc for them to use as an Asterisk box.

They currently only have cable with 384 - 512k upload speeds (different speeds during peak hours than off peak). They are willing to go up to a business class cable service providing them with 1.5mbps but first I have to get something working to show them this is really something they can rely on.

Last year I switched them to Cisco managed switching and alongside this project I will be installing an IPCop mini-itx router to manage the traffic better. This should help on the QOS end since the asterisk box will be (because there is no second option for high speed access at my residence) sharing the bandwidth with 12 - 15 pc’s.

To start what I need to understand is exactly what service to use for VOIP, so I can start a test line with them and play with the quality settings until I get everything just right. I need a 100% reliable service because they make calls to many different government agencies to publish their books and if the phones don’t work then the company grinds to a bleeding halt.

Next I need to know how I can use a 4 line phone system we already have with the system. It is one of those get the phone line in one place and then everything talks to the receiver systems. I know there is TONS of documentation on this particular system but I feel like i’m drowning in wiki’s and books after the last couple of days, if anyone can provide me with a direct link or even better models that I need that would be amazing.

Now as for the PC.

This is going to be a 4 line system to start, maybe more in time. I would like to get a low end sub $500 system to start and later migrate up to whatever system they need in the future. I figure a lower end Sempron or Pentium-D with two 80gb drives in Raid-1 will do the trick. If anyone has a better idea or thinks i’m missing something please feel free to add to this. I am going to get a battery backup capable of shutting down the system even in Linux so that shouldn’t be much of a hassle.

Sorry that this is so long but I wanted to cover as much as possible because I feel the advice I need should be given by informed individuals.

.:: Edit ::.
In all of that I forgot to mention that I have bought a copy of each of these in case someone wants to reference something in them.
Building Telephony Systems with Asterisk
Asterisk : The Future of Telephony

2

Even if you can only help by answering part of one of my questions that would still be great.

3

slow down there…

why don’t you buy a relatively inexpensive intel pc, sign up for voxee, vitelity, or some other pay by the minute plan. then get a single cheap phone like a grandstream 2000, or analog telphone adapter. install asterisk@home 2.7 (or trixbox) using nerd vittles website. then see how you like it in your office for pure voip calls without disrupting what you already have working. with gsm codec you won’t need much bandwidth for a few calls at once. just make sure you use qos. you can work on integration in steps as you learn more about asterisk and see what you need. this way you are not buying tons of stuff and ripping out everything all at once.

nothing is 100% reliable :laughing:

4

Looks like I didn’t mention that I am going to get a dummy line setup and working.

The trouble is I need to convert 4 digital lines (broadvoice is where I am looking) when the project is done.

I don’t want to have to buy a new converter or converters if the test is successful to avoid wasting the cost of a single line adapter.

I don’t want to buy any new phone equipment, I want to use my existing phone system.

I read about Trixbox but it seems to have serious reliability issues when compared to Asterix and Asterixnow. I have lots of prior experience working in Linux (I’ve even got a Debian file server at home now).

My request is for what kind of converter should I be looking for? Do I need 4 independent converters or is there a 4 line model I can buy? Is there anything I should look for in one of these boxes?

Thank you for the help as is it’s already helped me come to a better understanding of what codec I should be using. Although if anyone knows of or has a chart that shows what codecs utilize in bandwidth terms that would be amazing.

5

for a 4 line system, you should take a look at this:
nslu2-linux.org/wiki/Optware … g.Asterisk

People use Linksys NSLU2 (less than $80)to run linux+Asterisk with a additional 512MB USB drive(another $15). I have one at home. It’s small, consuming less power and it is quite!

If you want to use your existing traditional phone sets, then just add a couple of Linksys PAP2($60 each).

I know it will be a little challenge to set it up. But you will benefit from it for sure.

Now people is trying to put AsteriskNOW on top of it. Then we will have a web GUI.

6

OK I think you may have been reading a lot of hype.

You should read the info in the link below
voip-info.org/wiki/view/Asterisk+Primer

You state that you use Cable, There is no such thing a Business class cable service, they have packages they call that, but DOCSYS does not really support QOS, so you can not really compare CABLE modem to say T-1’s.

Second I do not think you are going to find any VoIP provider who can match the SLA you get with copper.

Are you in the US? (At&t / Bells will be one in few months)
Who do you get copper from now?

From almost any local bell telco I deal with I can NATIONWIDE LD for 25.00 month. you will not see in the VoIP world.

If this a Business in which the phones working = Money, then you should really look long and hard at your idea here.

If you do not have a good bit of time working with asterisk, this is something you should HIRE a pro to look at for you.

Many times when someone comes to me with the setup you are giving; I send them to the telco they use and have them lay it out them, MANY times the telco comes in with a plan which is better fit to needs of the customer.

If Vonage was bad you can give up and running four phones.

To give you a start you must do a cost take off on the current bills

providers like Connect.voicepulse.com has a online calculater where you can see how much money you may or may not by switching to VOIP.

The real solution would need you keep atleast some cooper one or two as a life line for when the cable goes out (as it will) using a card from digium of course.

as for low pc junker or AMD do not do it, grap a P4 DEll or so with min or 512 ram.

There are many ways to save money on telco and many different ways to include VOIP to the mix.

I have folks who add in a service like packet8.net/
so they have more lines when needed, plus get “better” phone system as they can have IVR / Faxing / Voice mail / forward to cell phones…

So have fun C-YA around the net…

7

Okay i will start from the beginning and try to answer what I can–

Your first and biggest problem is bandwidth. For VoIP to work correctly, especially with 12-15 pc users, you NEED QOS control on the WAN link. You don’t need much for this- many of the consumer routers provide enough QoS for you. A cheap way to get started is a Linksys WRT54GL (note the L) router and load a copy of DD-WRT firmware on it. That will give you a handful of other nifty features like traffic collector and VPN.
Bad QOS is almost certainly why Vonage sucked when you tried it- if you don’t give VoIP enough bandwidth, it will suck, period.
Second, there is no such thing as a 100% reliable provider of anything. Some are better than others, but everybody has downtime. In that regard, I’d recommend quantumvoice or viatalk more than broadvoice, BV occasionally has ‘quirks’ and both QV and VT have user forums- a very good sign with VoIP carriers. Having a user forum generally means that the company is confident enough in their service that they don’t mind customer issues being publicly posted on their website, and both QV and VT are very responsive support-wise. VT is quite a bit bigger than QV, but I’ve dealt with both of them and they both work well.
The other one to consider- connect.voicepulse.com. They are a ‘wholesale’ service which means you pay a flat fee of around $11/mo for each phone number, and then per-minute on outgoing calls. However, this can often be more cost effective esp. if you aren’t on the phone 24/7. And as you can imagine, their rates are probably quite a bit better than your local telco (it varies based on where you call but around here its 8/10 of a cent per minute- thats $0.008/min). You can also set your own outbound caller id number with this service, which may or may not be useful to you.

As for what you need to get started- you need 3 things:

  1. phone service of some kind

  2. a server to run * on (and yourself / time to config it)

  3. phones for your users

  4. already mentioned, although I should add that until you go all-voip you can get a card like a digium TDM400 to interface with analog channels. The TDM400 is a modular 4-port analog card, each port can be enabled as FXO (connects to a phone LINE, red module) or FXS (connects to one or more PHONEs and provides dialtone, green module) by adding the appropriate modules. The TDM400 itself is available pre-configured in every possible configuration. If you buy a pre-configured TDM400 you can purchase the modules later to bring additional ports online in whatever format you need down the road.
    I should also mention that for * to work right you need an unlocked/BYOD provider- many providers like vonage and packet8 lock service to the ATA, so the only way to use it is via the analog port (useless with *). You want one that will give you the SIP login and let you register * directly to their server. The ones I mentioned above (BV, QV, VT, VPC) all support BYOD and I have used all of them with *.

  5. What you mentioned is fine. Asterisk doesn’t use much hardware-wise, my business asterisk system is a P2-266 that sits on the floor. Without X running, it has handled upwards of 10 calls at a time (all bridged together too) without even breaking a sweat.
    Most VoIP work is very low-cpu- it just takes the packets from one IP and sends them to another, or figures out what you are trying to dial. There are a few things that eat CPU though. First and foremost is transcoding. If * has to translate between encoding formats in realtime it will use a lot of CPU, especially if you are using G.729 codec or iLBC. Others like GSM aren’t so bad. Also, bridging (conference rooms) takes at least some CPU because audio must be mixed together. The more channels bridged, the more CPU you need. But as my example demonstrates, it’s not that bad. Your machine as spec’d will be fine for your needs, overkill even. Battery backup is a very good idea.

  6. Phones. You can do this in one of three ways (or any mix of them): VoIP phones (IP phones that plug straight into Ethernet), analog phones (dumb POTS phones on FXS ports) or by using your existing phone system somehow connected to *.
    The best option is VoIP phones. They will talk directly to * and you get useful buttons like XFER, CONF, HOLD (which will do Music on hold via *), etc. I recommend AAstra and SNOM VoIP phones, both work with * quite well. You may also try Grandstream if you are on a budget, the GXP2000 isn’t bad. If you get the low end GS, get the BT200 not the BT1xx; the 200 supports intercom and can automatically hang up the speakerphone (bt1xx gives a busy tone until you hang it up yourself).
    I try to stay away from analogs becuase they lack these, and such features must be done with hookflashes and star codes. Remember that you need one FXS port for each analog extension you have- you can plug two phones into the same port but that is one extension and if you pick up both at a time you will have a 3way call.
    And lastly integration with your existing PBX. How difficult this is and/or if you want to do it depends on what you have. If you post a model number or something you can look it up, but try searching for it on voip-info.org. Much work has been done regarding making * deal with old PBXs, and this may save you some time.

Lastly, Asterisk:TFOT is a good book and will teach you the basics, but remember it’s a few months old and things change. VoIP-info.org is an EXCELLENT resource, it’s a Wiki that has just about everything regarding * and VoIP on it, including sample configs that make * work with all kinds of stuff.

Good luck and hope that helps!