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Wannabe Geek

Topic # 21462 27-Apr-2008 21:38
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Hi All

Been lurking here for ages, but this is my first post.

I run a small general IT services company with 5 staff.  Although I am technically competent, I have limited experience with Linux and no experience with Asterix or VOIP.

I would like to put in an Asterix based PBX system that can do the following:

1.  Incoming calls on our existing Analog lines (2 of them, 1 with faxability) are handled by the PBX
2.  Incoming callers get prompted to dial an extension number or hold for an operator.
3.  Incoming calls can be answered by anyone, transferred between extensions, put on hold, etc
4.  Unanswered calls go to a voicemail system
5.  All internal extensions are IP phones
6.  Outgoing calls can be directed over either the POTS lines or our ADSL connection to a SIP provider (i.e. press 1 for a POTS line or 2 for a VOIP line)
7.  We're able to make simultaneous outgoing calls on the SIP lines .
8.  We're still able to send / receive faxes
9.  The system is rock solid reliable and the quality of the VOIP calls is equal to that of the PSTN calls.

From the research I have done so far, I believe all this is reasonably achievable without too much aggravation.  I understand that I’ll need a TDM400P card with a 2 port FSO module to hook the PBX into the PSTN.

If I may, I’d like to ask a few questions:

1.  Is this all doable?  Are there any big holes in the list above or potential problems that I might not have anticipated?

2.  I plan to do this in 3 stages:
 a) testing / experimenting; 
 b) PBX and POTS lines only;
 c) PBX , POTS and SIP lines.
Does this seem a sensible approach?

3. While I’m testing, I’ll just use an old PC – Celeron 1.8 with 512mb RAM and a 40gb drive – it this spec going to work?

4. Again, while testing I’d like to use a softphone, preferably a free one – any suggestions.

5. Any other advice or suggestions will be most welcome!

6. Can you recomend a suitable SIP provider?

Thanks in advance


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  Reply # 128227 3-May-2008 17:17
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DaveWhite: Although I am technically competent, I have limited experience with Linux and no experience with Asterix or VOIP.

I think you could be asking for trouble Dave.  Asterisk is a very powerful system, and I believe this is also it's biggest weakness - for someone with no experience, it could end up being a long frustrating road to success.

A good place to start would be Tony's excellent tutorial on Trixbox which is available here.

To anwer your other queries
  1. All of this is certainly doable but I would change number 6 - make the system default VoIP, and push 1 to force out over PSTN.
  2. I'd change this and make the PSTN integration (which is the most important part of your communications with clients) the last step.  Run the system as pure SIP to start off with so that you and your staff can become familiar with the phones and system before swapping your phone lines in.
  3. Not too sure as Asterisk isn't my forte - you may want to get a higher spec'd computer and start testing on that - when you are ready to move from testing to production you don't have to move everything from one server to another.
  4. My opinion is you can't go wrong with a desk phone, and I'd steer clear of soft phones.
  5. .
  6. WorldxChange or 2Talk in that order.
I think your best course of action would be to get a third party company (shameless plug) to install a new VoIP phone system for you, then ask for access so you can maintain it on a day-to-day basis.  My opinion is your telecommunications is not really something you want to get wrong - frustrated customers will always hang up and move on.

Hope all works out well.

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