Asterisk based solution for 180 phones

Asterisk Asterisk Support
#1

Hello everyone,

it`s my first post on this forum so forgive me for any mistakes I make writting this post. I got known with Asterisk a few years ago but only a month ago I thought of giving it a try :smile: Do you think that Asterisk might do fine with a telecomunication problem presented below?

The company has 8 locations:
[]location A - 120 phones[/]
[]location B - 30 phones[/]
[]6 location with 5 phones each[/]

Number of simultanious calls:
[]between A and B - 7 calls[/]
[]between 6 locations and A or B - 4 calls[/]
[]outside the company from all locations - 9 calls[/]
[]inside A - 10 calls[/]
[]inside B - 5 calls[/]

No additional servies are needed at the moment.

Now the list of question:

[]Do you think that Asterisk will be the most cost effective platform or is there something cheaper?[/]
[]How many servers are needed? I was thinking about one in location A and one in location B. Do you think that solution could work well with only one server in location A?[/]
[]How powerfull the servers should be to handle this kind of traffic for one/two server based solution? [/]
[]Do I need to have digium training (the 5-day one) or will I be able to launch the solution on my own? I have good knowlegde from SIP and IP related topics but for sure have some lack of knowledge in linux/unix administration[/]
[]What are the main threats in developing this solution? I know that stable IP connection beetwen locations is a must, but what can be the biggest problem for a Asterisk beginer?[/]
[]What Asterisk version should I use?[/]

I hope you managed to get till the end :smile: I look forward to your answers and comments on what I wrote - I really need a helpfull hand at the beginning ")

regards,
Łukasz

#2

Did I put those questions in a wrong place?

regards,
Łukasz

#3

[]Do you think that Asterisk will be the most cost effective platform or is there something cheaper?[/]

Nothing Cheaper - but that should NEVER be the starting criteria on a decision of this size - stability and suitability are key, everything else is secondary.

[]How many servers are needed? I was thinking about one in location A and one in location B. Do you think that solution could work well with only one server in location A?[/]

I would put a server in A and B - the remote sites are another question and that is if you can get bandwidth to them that supports Voice Prioritization (QOS) - without that, don’t even try!

[]How powerful the servers should be to handle this kind of traffic for one/two server based solution? [/]

A standard QuadCore Xeon Dell with at least 4 Gig’s of RAM should be fine as long as you are not transcoding like crazy - pick a codec (Either G.729 or G.711) and stick with it across the entire enterprise - as few simultaneous calls as you have, G.729 (the Paid Digium Version) would be my choice. Recording calls is another question - if you need to record the calls, a better disk subsystem would be called for (RAID 10 or better).

[]Do I need to have digium training (the 5-day one) or will I be able to launch the solution on my own? I have good knowlegde from SIP and IP related topics but for sure have some lack of knowledge in linux/unix administration[/]

My suggestion would be to hire somebody who already does LOTS of these on a DAILY BASIS - this is FAR too complex if you are just starting out with Asterisk - Don’t give Asterisk a bad name by taking a project like this and then going down in flames!

[]What are the main threats in developing this solution? I know that stable IP connection beetween locations is a must, but what can be the biggest problem for a Asterisk beginner?[/]

WAY too many to mention here - see above answer!

[]What Asterisk version should I use?[/]

1.4 is probably the most conservative and stable right now, but you need to look at the feature sets and see if there is something that is a MUST-HAVE in the 1.6 or even the 1.8 - but not to sound like a broken record here, this is a really complex project - seek competent help and learn from them in the process - if you are not VERY comfortable with Asterisk already, a project this size is doomed to failure.

Greg

#4

I agree with Greg, Don’t do this yourself. It is a large, but not overly complicated project.

First couple of project planning questions for you?

  1. Is there a IP network in place in all locations?
  2. Is that IP network dedicated to the phone service? If not, is is QoS aware?
  3. Is there a VPN or private network between these locations?
  4. What size internet connections do you have at each site?
  5. What country(s) is this to be installed in?
  6. What is your budget to implement this project?
  7. What is the expected growth from 180 phones in the next 3 years?
    8 ) Will you be using SIP trunking or telephone company connectivity?

Give me some more to go on and we can talk about the scope of this project.

Thanks,

padapa

#5

GSnover, padapa - thanks for your answers :smile: They allowed me to see the topic from a different perspective. I didn`t realise this was such a big project.

Do you want to say by that, that Asterisk is the most stable solution of all software based solutions?

[quote=“padapa”] First couple of project planning questions for you?

  1. Is there a IP network in place in all locations?
  2. Is that IP network dedicated to the phone service? If not, is is QoS aware?
  3. Is there a VPN or private network between these locations?
  4. What size internet connections do you have at each site?
  5. What country(s) is this to be installed in?
  6. What is your budget to implement this project?
  7. What is the expected growth from 180 phones in the next 3 years?
    8 ) Will you be using SIP trunking or telephone company connectivity?
    [/quote]
    And the answers:
  1. Yes.
  2. The network is not dedicated - its for voice and data combined. We havent checked this yet.
  3. At the moment no.
  4. There are DLS links from all the locations (8Mbps - A, 2Mbps - B, 1Mbps - others)
  5. All locations are in Poland.
  6. I don`t know yet. I am the one to estimate the cost.
  7. No growth at locations A and B. Possibility of creating another big location.
  8. If you mean the connectivity between A and B we want to use SIP, all the outside calls are supposed to be conected via Asterisk to telephone company using PSTN.

regards,
Łukasz

#6

Łukasz,

Thanks for the info;

If there is not a VPN system between these locations, firewall work will have to be done to configure the phone’s & servers access between the sites. Have may want to look at Vyatta’s open source router …

[code]The company has 8 locations:
[]location A - 120 phones[/]
[]location B - 30 phones[/]
[]6 location with 5 phones each[/]

Number of simultanious calls:
[]between A and B - 7 calls[/]
[]between 6 locations and A or B - 4 calls[/]
[]outside the company from all locations - 9 calls[/]
[]inside A - 10 calls[/]
[]inside B - 5 calls[/]
[/code]

Here are a couple general options for you.

Option 1
Use a E1 Connection, ISDN or analog circuits for connections to the PSTN at site A. All other traffic needs to connect back to site A for calling out and between locations. In this scenario, a narrow band codex must be used. Verify what codec’s your phones will support. They must support G729A, GSM, iLBS or SPEEX.

Option 2
Use local asterisk servers at each site. The small sites can use mini-itx servers to control costs. Use SIP trunk using narrow band codecs to connect the small sites back to sites A and B, as well as between A and B. Dialplan management will be the primary work here. You can still send all calls out of site A. Optionally, have trunks out of site B to the PSTN too, for disaster recovery purposes.

If you would like additional help, PM me for rates and such.

Best wishes for a successful project.

padapa

#7

Thanks a lot for your help :smile: I see that a test environment is needed here. I will try to run on it all the scenarios to see what are my skills :smile:

regards,
Łukasz

#8

I have done a multiple site setup over VPN and non-VPN previously. The VPN encapsulates the Voice stream over the UDP for better quality in general. It work either way however.

If I were contracted to setup a solution like this I would have different extension ranges at each site that allowed multiple SIp trunks to connect to each other for site to site dialing. Its a huge star layout.

A Test environment is needed for someone who does not know how to properly implement something like this. I have no inhibitions about setting something like this up, but as a previous poster noted I would be eager to find out about the VPN status and whether or not the routers enjoyed the company of SIP.

Thanks,

-Jake
www.voipcitadel.com