Need Help Designing Implementation

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I’d like to replace my current VOIP provider with an Asterisk based solution. I have some ideas I want to run by the list to see if they are possible, and get answers to a couple questions.

I want to setup two Asterisk servers that are linked to each other:
– The first server would be my “external” (public) server and would live in a real data center. The second server would be my “internal”
(private) server and would live in my house.
– The external server would receive all incoming calls and handle the voice mail stuff.
– The internal server would run all the phones in my house (VOIP or Analog-via-FXS). All outgoing calls would be routed out through the external server.

I also want to add the following additional functionality:
– If the external server looses connectivity to the internal server while a call is in progress, the external server should place the call on hold while it tries to reach us via our cell phones. A message should be played informing the remote party that the connection had been lost and it is trying to re-establish it now. If it can’t reach us, it should inform the remote party that the connection could not be re-established and allow the remote party to leave some closing remarks on the voice mail system.

– If a call comes in and no one is at home to take the call (or if all lines at home are busy), it should ring all of our cell phones and whoever answers the call first gets the call. If no one answers the call via the cell phones after 3 rings, it should route the call to the voice mail system. I say 3 rings on the cell phone because I do not want the cell phone voice mail to take the call.

– I also would like the system to automatically route all calls directly to voice mail depending on the time of day (say 10PM to 8AM).
I would like specify in a “white list” specific phone numbers that are allowed to ring through regardless of time of day (i.e. her parents, my parents).

– I would like the VOIP phones to turn on the voice mail waiting indicator light if the external server has new voice messages.

Is all of this possible? If not, which part’s are not (and how much work do you think would be needed to make those parts work)?

2 thoughts on - Need Help Designing Implementation

  • That all seems perfectly doable.

    I don’t think that’s doable without quite a lot of work – but others may be able to advise further. To elaborate a little, it’s easy to detect whether a route is usable when a call is placed, but detecting a call failure *during* the call is much more difficult.

    That’s easy, though remember asterisk does things in seconds rather than
    “rings”. You should also remember there’s a delay in processing the call through the mobile networks before the phone actually starts “ringing” – in the UK that averages around 7 seconds between the call being sent to the mobile network from your server, and the phone ringing.

    Shouldn’t be difficult.

    I believe this is doable in the newer versions of asterisk, but not the older versions. Again, someone else will hopefully chip in here, since our stuff is still running 1.4 🙂

    As is so often the case, (almost) anything is possible if you’re prepared to spend time doing it. How much is worth doing depends on your time, and what else you might prefer to be doing with it…

    FWIW, you might want to think about whether you actually need a separate asterisk box at home. In my experience, unless you have many dozens of extensions, you’re almost better off (and certainly no worse off)
    connecting your SIP devices at home (assuming you’re using SIP) directly back to the * server in the datacentre. One less box to maintain, and things like MWI will “just work” without having to play with the messaging interfaces.

    Kind regards,