Building a large PBX system/Asterisk Guru for hire needed

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Building a large PBX system/Asterisk Guru for hire needed

Postby pbxconsultant » Thu Jan 10, 2008 3:00 pm

I like to start by thanking anybody who will reply to this post. I truly appreciate it. I am a LAMP system Professional, a network specialist, and I also do consultation and installation for PBX systems. I have a big project coming up that requires over 1000+ IP lines with the support of around 400 concurrent calls.
I am planning to hire an asterisk professional to help me build that system (more information, if you are interested below), however before I go ahead and spend money, and time on that I want to make sure that the project is feasible both economically and technically, which is the subject of this post.
I have read a lot, installed asterisk and asterisknow, toyed with SER and got it working in its simple form. I got the feel of the system however uncertainty still looms over my head. I will start with more information about the project then I will proceed with my questions.

1-The project is a hotel/motel (resort) project
2-1000+ SIP stations and support for 400 concurrent calls.
3-It’s all enclosed in one location, i.e. there are no NATing involved, no remote users no connections to other sites over VOIP.
4-The SIP phones in the rooms are connected by a100 mb/s CAT 5e or CAT 6 cables carrying IPTV, internet and VOIP to the rooms. And tied together with gigabit switches and fiber optics (to tie different buildings). Which tells me I have no bandwidth issues, and I don’t have to resort to any CPU hog CODECs or ones that require licensing. Also I don’t think I have to resort to transcoding except maybe for the voice mail which I will explain below.
5-All outgoing or incoming calls to the outside world will be handled by 2 E1 trunks (up to 4 max), and 16 CO lines.
6-There will be another 16 single line ports (analogue) FXS stations for fax machines/modems..
7-Voicemail is optional as messages will always be handled by the receptionists. However if we use, it will handle very limited voice mail messages for management/housekeeping, etc. I can use the voice processing card form DIGIUM to handle that if possible.
8-The playback() features will be limited to music on hold and voicemail functions. No Automated attendant, or IVR needed (all handled by the reception which is available 24x7).
9-Most of the features required are already available with asterisk however some extra features for the hotel motel functionality which I haven’t seen on asterisk will be required. For instance:
a. Wakeup call.
b. Check in / checkout (with station lock)
c. PMS interface to deal with the hotel software system and to send out call accounting information to that software (requires custom programming to implement the protocol and it’s connected to it using a serial interface).

My questions are:
1-Can I run this system on asterisk? Will asterisk be able to handle the load?
2-Will I have to resort to using SER as a proxy in the middle, or can I have asterisk handle the whole load?
3-How reliable will the system be compares to ready systems like the ones from Toshiba?
4-And of course one the most debated/discussed question (your best guestimate will be much appreciated as I still can’t see any definite answers) What kind of computer hardware will be able to handle this (I know what I need when it comes to the DIGIUM hardware; analogue and digital cards), and can I fit it on one system on a dual proc Quad core 3.2 GHz Intel Xeon system (1600 MHZ system bus) with a supermicro motherboard for instance or do I have to resort to multiple servers.
5-Are features like the ones mentioned in item 9 above can be easily implemented?
6-Are my conclusions when it comes to codec choices and transcoding correct?

If you think you are capable of building that system, and you live in Canada somewhere around the Greater Toronto Area or you are willing to travel to Ontario, Canada please do not hesitate to contact me.

Thanks for taking the time to read this post, and replying to it.
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Postby smaikol » Mon Feb 04, 2008 8:33 am

Hi pbxconsultant,

Yuor project is wide, that's for sure. 400 concurrent calls are a lot so, for sure, asterisk as it is is NOT the answer, but, you must deal with an environment with SER+Asterisk. Taht is, SER for managing calls and asterisk to implement services.

Ofcourse you need also to evaluate a cluster implementation since if i don't remeber wrong ser can deal with 150 concurrent calls per second that is a bit far from your expected 400, so cluster means, ser+asterisk+mysql, maybe in realtime mode. As for the other functions, faxes and so on...more than digium cards i suggest you patton sip gateways, they are sip\pstn(isdn or analog) so you can both use directly with ser as ua or with asterisk as trunk termination or extensions.

i'm far far away form canada, so i cannot be of help to yuo to implement this, but we can exchange informations and advices if you agree.

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Postby bkruse » Mon Feb 04, 2008 9:26 am

Moved to jobs section
Site Admin
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Postby sub_zero255 » Mon Feb 04, 2008 5:56 pm

Did you find answers to all your questions ? have you make this done ?
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Postby dgarstang » Mon Feb 04, 2008 11:38 pm

I'd like to correct a couple of things that smaikol said.

1. The original poster said he wanted 400 concurrent calls, not 400 calls per second. They aren't the same thing.

2. It's my understanding that SER/OpenSER can handle 3,000-5,000 new call setups per second (not 150), depending on whether you are handling calls in statefull or non stateful mode.

3. I would advise against using MySQL with OpenSER directly. OpenSER blocks on database access. A slow database or a network problem means slow calls.

4. 400 concurrent calls isn't a lot, really... The number of concurrent calls a single system can handle is dependent on several factors, but I think 2-3 servers should handle 400 concurrent calls (without transcoding).

5. Dependant on call ramp-up speed, I really don't see OpenSER being a critical requirement here.

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Postby cadillackid » Tue Feb 05, 2008 1:29 pm

there is a company in wisconsin developing an * based system.. they are Innovation Voice Technologies.. this company offers systems strictly to the hotel / motel market and has already created the PMS interface / Hotel Lodging features(wakeup, maid status, etc).. their servers are designed for large SIP implementations as well as reliability and includes voicemail as well.. you may want to talk to them as their product is fairly new but i have had a chance to see it and beat on it a little and think that it can be a viable solution.

this product appears to be much more cost effective than a legacy solutions such as a mitel / Nortel / Avaya.. and of course allows you to use many different selections of sets...

My company deals in mainly legacy PBX along with VOIP solutions by Mitel.. we also will be handling the new Innovation system when it is fully available.. I myself think you will put a lot of $$ into developing the hotel motel features and reporting functions for a bare asterisk system and it might be wise to check into something like i mentioned above...

using a mitel / nortel etc will end up killing you in licensing as they are steep on the software costs...
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Postby pbxconsultant » Sat Feb 09, 2008 8:51 pm

Hello Christopher,
I just sent you a pm please check it.

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PBX help

Postby voipguy » Mon Apr 14, 2008 12:41 pm

did you find someone?
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