Looks like the week for dial plan help...`

Discussion in 'UK VOIP' started by linker3000, Oct 4, 2006.

  1. linker3000

    linker3000 Guest

    OK guys - here's another one...

    I've set up a Linksys/Sipura 3102 with Asterisk. It works OK but not
    reliably - ie: I can use it to ring other IP phones on the network or
    can call it from other phones for a few times and then it needs a
    reboot. I'm currently playing with the SIP settings but any ideas would
    be appreciated. Haven't even hooked it up to the PSTN yet.

    I've also found that although I have enabled the Web interface from the
    'Internet' side, the interface dies after a minute or so, but it works
    fine from the 'local' side.

    Anyway - can anyone advise on a dial plan that will do the following:

    All outbound calls to go out via VoIP EXCEPT 999 and 112 which should
    use the PSTN line. Calls starting 9 also to go out via PSTN (eg: '9' for
    a PSTN line - like 901234 567 8900).

    linker3000, Oct 4, 2006
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  2. linker3000

    Brian A Guest

    Are you wanting dial plan help for the SPA-3000 or do you want help
    connected with Asterisk?
    If it is SPA-3000 can I suggest that you learn the basics of how to
    construct a dial plan. Dialing 9 to route all the numbers that need to
    go via PSTN is OK for the exception, where your VSP is down, but not
    for day to day operations. If you construct your dial plan properly
    you should have to think whether a number is going via VOIP or PSTN -
    the dial plan should take care of that. Further, as you are developing
    you system you may want to make changes, coming back here every time
    to get a new dial plan isn't going to be very good regime.
    Can I suggest that you take a look at some notes I have made and, if
    after that, you are stuck then please come back here for help.
    Check out the notes on
    Remove 'no_spam_' from email address.
    Skype Free Zone!!
    Brian A, Oct 5, 2006
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  3. linker3000

    Daviey Guest


    Will your PSTN be attached to the * server or connected to the Sipura

    Open up a shell to your * box. If your using Trixbox i believe it
    includes a Java SSH client. If not download putty* and login to your
    asterisk server (using asterisk's IP). once in type 'asterisk
    -rvvvvvv'. This will get you into the asterisk console. When you
    make/recieve calls you can see what asterisk is doing. Watch for when
    the line ceases and report back what appears on the screen.

    Also commands such as 'sip show registry' and 'sip show peers' will give
    you some idea of the connection status of the extenions(peers) and voip

    Let us know how you get on!

    *Putty client: http://the.earth.li/~sgtatham/putty/latest/x86/putty.exe
    Daviey, Oct 5, 2006
  4. linker3000

    Linker3000 Guest

    Hi Brian,

    I will have a look at your notes - thanks.

    This setup is for a small business with one incoming phone line hence
    all outbound calls are planned to go via VoIP to keep the line free for
    one inbound call - not ideal but best under the current circumstances.
    The system is unlikely to develop any further than this.
    Linker3000, Oct 6, 2006
  5. linker3000

    Linker3000 Guest

    Hi Daviey,

    The PSTN will be integrated with the * (Trixbox) system via the SPA3102
    - at the moment, the company takes a call on the PSTN line with a DECT
    phone and they walk it to the recipient so I want instead for the
    inbound PSTN call to be handled by Trixbox so they can transfer it to
    the required extension. All other aspects of the system are working fine
    - 7 extensions all with inbound 0845 numbers routing to the desks (apart
    from inbound fax detection but I an working on it).
    Linker3000, Oct 6, 2006
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