[GrandStream HT-488] How to configure?

Discussion in 'VOIP' started by Vincent Delporte, Nov 28, 2006.

  1. Hello

    After a bit of fiddling, I got to the web interface and can more or
    less successfully receive a call through an Asterisk server using this
    SIP gateway.

    I have the following questions, though:

    1. HT488 doesn't detect that the remote end has hung up: What should I
    use for France as regional settings in the FXO Port section? Here are
    the default settings that show up in the web interface:

    PSTN AC Termination: 320 Ohm + (1050 Ohm || 230 nF))
    PSTN Disconnect Tone: Frequency: f1 480 f2 620
    PSTN Disconnect Tone Cadence: All 0's
    PSTN Silence Timeout : 60

    2. No CID is returned, even though it works OK if I plug a caller
    ID-capable modem on the phone line. HT488 doesn't handle CID, or is it
    linked to regional settings above?

    3. How to secure access to PSTN line so that unauthorized users from
    the Net don't use my PSTN line?

    Thank you.
    Vincent Delporte, Nov 28, 2006
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  2. Vincent Delporte

    ldemsp Guest

    I don't live in France but as far as I know, the busy tone in France
    is 440 Hz with 0.5/0.5 sec for on/off rate. Try to set your equipment

    If it is not the busy tone what is played by your exchange when the
    remote end has hung up, and nobody knows what is played actually,
    (=even the telco guys can't help you) than the actual tone is to be
    analysed, and you should configure the HT488 accordingly.
    ldemsp, Dec 3, 2006
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  3. Thanks, I'll try to change the settings.

    What about caller ID from the FXO port? Back in Oct 2005, someone said
    that CID is only available for the FXS port, but support for FXO might
    be added in future firmware... but the unit is already up to date.

    If no CID, I can't use that equipment :-/
    Vincent Delporte, Dec 3, 2006
  4. Vincent Delporte

    ldemsp Guest

    I don't even understand the question.
    CID can only work if it is received from the FXO port, and it is
    relayed to either the FXS port, or forwarded as a source ID to the
    VoIP termination target. Or the alternate direction, when the call can
    come from the VoiP leg, and the CID is shown on the FXS port again.

    I really don't understand what are you waiting for. You will never
    have the equipment transfer CID from the FXS or from VoIP towards the
    FXO side. It is nonsense: the switch (PBX, or local exchange) will
    give you the CID according to your call number, not by the info it
    gets from the station.

    It expects and understands hook on/hook off/DTMF/hook-flash from the
    station side, and nothing else.
    ldemsp, Dec 4, 2006
  5. If I hook up the HT to the network and call in from the PSTN, it
    doesn't forward CID to the Asterisk server. There is CID-related items
    in the FXS section, but nothing in the FXO section. I'll see if the
    former is enough to have the unit send CID information to Asterisk, or
    if I have to change regional settings.

    Vincent Delporte, Dec 5, 2006
  6. Vincent Delporte

    ldemsp Guest

    I see. You are asking if the equipment can understand the CID from the
    FXO interface in order to translate it to FXS interface (and to the
    network) or it just rings thru.

    Theoretically it understands CID, and should forward it with the call.
    In France you should use ETSI-FSK type setting. Check if you have
    unset block caller ID. *31 is the unsetting from the phone, and you
    should set the "Send Anonymous" option to No.
    ldemsp, Dec 5, 2006
  7. No. I don't use the FXS port at all (no handset connected to it). I
    just use the FXO port and the WAN port so that the HT488 is used as an
    SIP gateway for an Asterisk server.
    CID works on this line, but I'll see if making regional changes in the
    FXS section has an effect on the FXO section.

    Vincent Delporte, Dec 7, 2006
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