BT Diverse 4010 doesn't show CID from PSTN callers when connectedthrough VOIP GATE 102

Discussion in 'UK VOIP' started by dmitri, Feb 14, 2007.

  1. dmitri

    dmitri Guest

    Please can anybody help! I've asked already but havn't got a clear answer.
    I tried the setting that sipgate recommends for Grandstream Handytone-486.
    Everything( inbound CID to incoming VOIP number, all outgoing CID) works
    fine but incoming CID to PSTN number don't show up! I tried different
    cables, BT-to-RJ11 adapters,microfilters between ATA and phone, etc. etc.

    DECT: BT Diverse 4010
    ATA: VOIP GATEWAY 102 (clone or similar to Grandstream
    Handytone-486/488)with the following settings
    FXS Impedance: CTR21(270R+750||150nF)
    Onhook Voltage: 48V
    Caller ID Scheme: ETSI-FSK
    DTMF Send Type: Via SIP INFO
     
    dmitri, Feb 14, 2007
    #1
    1. Advertising

  2. dmitri

    DGB Guest

    Re: BT Diverse 4010 doesn't show CID from PSTN callers when connected through VOIP GATE 102

    In news:,
    dmitri <> typed:
    > Please can anybody help! I've asked already but havn't got a clear
    > answer. I tried the setting that sipgate recommends for Grandstream
    > Handytone-486. Everything( inbound CID to incoming VOIP number, all
    > outgoing CID) works fine but incoming CID to PSTN number don't show
    > up! I tried different cables, BT-to-RJ11 adapters,microfilters
    > between ATA and phone, etc. etc.
    >


    If I'm understanding your symptons and set-up correctly, it sounds to me as
    though you're asking the ATA to do something it's not designed to do.

    When no calls are in progress the phone will be connected to the VOIP line
    of the ATA, so that on picking up the phone you can dial straight out on the
    VOIP line. The ATA swiches the phone to the PSTN line either by dialling **
    or when the PSTN line starts to ring. However, BT CLI is sent before the
    first ring, at which point the phone is still connected to the VOIP line,
    therefore the BT CLI FSK data isn't routed to the phone.

    --
    Don
     
    DGB, Feb 14, 2007
    #2
    1. Advertising

  3. dmitri

    dmitri Guest

    Re: BT Diverse 4010 doesn't show CID from PSTN callers whenconnected through VOIP GATE 102

    On Wed, 14 Feb 2007 20:03:37 +0000, DGB wrote:

    > In news:,
    > dmitri <> typed:
    >> Please can anybody help! I've asked already but havn't got a clear
    >> answer. I tried the setting that sipgate recommends for Grandstream
    >> Handytone-486. Everything( inbound CID to incoming VOIP number, all
    >> outgoing CID) works fine but incoming CID to PSTN number don't show
    >> up! I tried different cables, BT-to-RJ11 adapters,microfilters
    >> between ATA and phone, etc. etc.
    >>

    >
    > If I'm understanding your symptons and set-up correctly, it sounds to me as
    > though you're asking the ATA to do something it's not designed to do.
    >
    > When no calls are in progress the phone will be connected to the VOIP line
    > of the ATA, so that on picking up the phone you can dial straight out on the
    > VOIP line. The ATA swiches the phone to the PSTN line either by dialling **
    > or when the PSTN line starts to ring. However, BT CLI is sent before the
    > first ring, at which point the phone is still connected to the VOIP line,
    > therefore the BT CLI FSK data isn't routed to the phone.
    >


    Thanks, Don, it does make sense. But I found the following thread at
    http://www.velocityreviews.com/forums/t235141-spa3000-and-cli-query.html:

    .......

    I can confirm that the FXO port on the Sipura 3000 will indeed work with
    BT's CLID. There is a regional setting for Caller ID Method which I have set
    to ETSI FSK With PR(UK). I can call my PSTN line from my mobile and the
    Sipura is correctly passing the caller ID onto my * box.

    There is only one regional setting that I can see - so it looks like the FXO
    and FXS ports use the same setting.

    Software version: 2.0.11(GWg)
    Hardware version: 2.0.1(0875)

    HTH
     
    dmitri, Feb 14, 2007
    #3
  4. dmitri

    Jono Guest

    Re: BT Diverse 4010 doesn't show CID from PSTN callers when connected through VOIP GATE 102

    dmitri brought next idea :
    > On Wed, 14 Feb 2007 20:03:37 +0000, DGB wrote:
    >
    >> In news:,
    >> dmitri <> typed:
    >>> Please can anybody help! I've asked already but havn't got a clear
    >>> answer. I tried the setting that sipgate recommends for Grandstream
    >>> Handytone-486. Everything( inbound CID to incoming VOIP number, all
    >>> outgoing CID) works fine but incoming CID to PSTN number don't show
    >>> up! I tried different cables, BT-to-RJ11 adapters,microfilters
    >>> between ATA and phone, etc. etc.
    >>>

    >>
    >> If I'm understanding your symptons and set-up correctly, it sounds to me as
    >> though you're asking the ATA to do something it's not designed to do.
    >>
    >> When no calls are in progress the phone will be connected to the VOIP line
    >> of the ATA, so that on picking up the phone you can dial straight out on the
    >> VOIP line. The ATA swiches the phone to the PSTN line either by dialling **
    >> or when the PSTN line starts to ring. However, BT CLI is sent before the
    >> first ring, at which point the phone is still connected to the VOIP line,
    >> therefore the BT CLI FSK data isn't routed to the phone.
    >>

    >
    > Thanks, Don, it does make sense. But I found the following thread at
    > http://www.velocityreviews.com/forums/t235141-spa3000-and-cli-query.html:
    >
    > ......
    >
    > I can confirm that the FXO port on the Sipura 3000 will indeed work with
    > BT's CLID. There is a regional setting for Caller ID Method which I have set
    > to ETSI FSK With PR(UK). I can call my PSTN line from my mobile and the
    > Sipura is correctly passing the caller ID onto my * box.
    >
    > There is only one regional setting that I can see - so it looks like the FXO
    > and FXS ports use the same setting.
    >
    > Software version: 2.0.11(GWg)
    > Hardware version: 2.0.1(0875)
    >
    > HTH


    I'm not sure how the GATE is supposed to integrate with the PSTN,
    however, the SPA3000 is essentially in two parts - a PSTN side & a VoIP
    side.

    Whenever a call comes in on the PSTN, it is then passed to the VoIP
    side of the ATA, presumably using some VoIP internally.

    Only when the power is off (and a relay shuts) is the attached phone
    actually connected to the PSTN.

    The SPA clearly sees the CLI before the ring, then passes the call to
    the VoIP side together with the ID intact - however, there is a one
    ring delay between when the PSTN line rings & the connected phone
    rings.
     
    Jono, Feb 14, 2007
    #4
  5. dmitri

    DGB Guest

    Re: BT Diverse 4010 doesn't show CID from PSTN callers when connected through VOIP GATE 102

    In news:,
    Jono <> typed:
    >
    > I'm not sure how the GATE is supposed to integrate with the PSTN,
    > however, the SPA3000 is essentially in two parts - a PSTN side & a
    > VoIP side.
    >
    > Whenever a call comes in on the PSTN, it is then passed to the VoIP
    > side of the ATA, presumably using some VoIP internally.
    >
    > Only when the power is off (and a relay shuts) is the attached phone
    > actually connected to the PSTN.
    >
    > The SPA clearly sees the CLI before the ring, then passes the call to
    > the VoIP side together with the ID intact - however, there is a one
    > ring delay between when the PSTN line rings & the connected phone
    > rings.


    That's interesting. I haven't actually tried to get CLI from a BT line
    plugged into the PSTN port of my Grandstream 486, since I already have stand
    alone caller display units directly connected to the BT incoming line (one
    of which also indicates CLI of calls waiting). What I have found, though,
    is that the facility on the 486 to access the PSTN line only works properly
    when you have a "real" PSTN line connected, and not a PABX extension, which
    is how I first tried configuring it. I think it must be something to do
    with ATA needing a full 50v line rather than the 24v it gets from the PABX.
    The 486 also senses when no line is connected to the PSTN port, in which
    case dialling ** gives an engaged tone.
    --
    Don
     
    DGB, Feb 15, 2007
    #5
  6. dmitri

    speckled hen Guest

    Re: BT Diverse 4010 doesn't show CID from PSTN callers when connected through VOIP GATE 102

    On Feb 15, 12:35 am, "DGB" <> wrote:
    > Innews:,
    > Jono <> typed:
    >
    >
    >
    > > I'm not sure how the GATE is supposed to integrate with the PSTN,
    > > however, the SPA3000 is essentially in two parts - a PSTN side & a
    > > VoIP side.

    >
    > > Whenever a call comes in on the PSTN, it is then passed to the VoIP
    > > side of the ATA, presumably using some VoIP internally.

    >
    > > Only when the power is off (and a relay shuts) is the attached phone
    > > actually connected to the PSTN.

    >
    > > The SPA clearly sees the CLI before the ring, then passes the call to
    > > the VoIP side together with the ID intact - however, there is a one
    > > ring delay between when the PSTN line rings & the connected phone
    > > rings.

    >
    > That's interesting. I haven't actually tried to get CLI from a BT line
    > plugged into the PSTN port of my Grandstream 486, since I already have stand
    > alone caller display units directly connected to the BT incoming line (one
    > of which also indicates CLI of calls waiting). What I have found, though,
    > is that the facility on the 486 to access the PSTN line only works properly
    > when you have a "real" PSTN line connected, and not a PABX extension, which
    > is how I first tried configuring it. I think it must be something to do
    > with ATA needing a full 50v line rather than the 24v it gets from the PABX.
    > The 486 also senses when no line is connected to the PSTN port, in which
    > case dialling ** gives an engaged tone.
    > --
    > Don


    I have caller ID on my 2 lines on a Linksys spa2102 and it works fine.
    make sure the caller ID method is set to:
    etsi Fsk With PR (UK)
    on default mine was set to something else
     
    speckled hen, Feb 15, 2007
    #6
  7. dmitri

    Jono Guest

    Re: BT Diverse 4010 doesn't show CID from PSTN callers when connected through VOIP GATE 102

    speckled hen explained :
    > I have caller ID on my 2 lines on a Linksys spa2102 and it works fine.
    > make sure the caller ID method is set to:
    > etsi Fsk With PR (UK)
    > on default mine was set to something else


    The original OP has CLID working using VoIP but not on inbound PSTN
    calls.
     
    Jono, Feb 15, 2007
    #7
  8. dmitri

    Dmitri Guest

    Re: BT Diverse 4010 doesn't show CID from PSTN callers whenconnected through VOIP GATE 102

    On Wed, 14 Feb 2007 23:34:18 -0800, speckled hen wrote:

    > On Feb 15, 12:35 am, "DGB" <> wrote:
    >> Innews:,
    >> Jono <> typed:
    >>
    >>
    >>
    >> > I'm not sure how the GATE is supposed to integrate with the PSTN,
    >> > however, the SPA3000 is essentially in two parts - a PSTN side & a
    >> > VoIP side.

    >>
    >> > Whenever a call comes in on the PSTN, it is then passed to the VoIP
    >> > side of the ATA, presumably using some VoIP internally.

    >>
    >> > Only when the power is off (and a relay shuts) is the attached phone
    >> > actually connected to the PSTN.

    >>
    >> > The SPA clearly sees the CLI before the ring, then passes the call to
    >> > the VoIP side together with the ID intact - however, there is a one
    >> > ring delay between when the PSTN line rings & the connected phone
    >> > rings.

    >>
    >> That's interesting. I haven't actually tried to get CLI from a BT line
    >> plugged into the PSTN port of my Grandstream 486, since I already have

    stand
    >> alone caller display units directly connected to the BT incoming line (one
    >> of which also indicates CLI of calls waiting). What I have found, though,
    >> is that the facility on the 486 to access the PSTN line only works

    properly
    >> when you have a "real" PSTN line connected, and not a PABX extension,

    which
    >> is how I first tried configuring it. I think it must be something to do
    >> with ATA needing a full 50v line rather than the 24v it gets from the

    PABX.
    >> The 486 also senses when no line is connected to the PSTN port, in which
    >> case dialling ** gives an engaged tone.
    >> --
    >> Don

    >
    > I have caller ID on my 2 lines on a Linksys spa2102 and it works fine.
    > make sure the caller ID method is set to:
    > etsi Fsk With PR (UK)
    > on default mine was set to something else


    Do you mean "Polarity Reversal" by PR(UK)? Yes, I did set it. It actually
    says "Polarity Reversal" without specifying (UK) if it matters.

    Thanks,

    Dmitri
     
    Dmitri, Feb 15, 2007
    #8
    1. Advertising

Want to reply to this thread or ask your own question?

It takes just 2 minutes to sign up (and it's free!). Just click the sign up button to choose a username and then you can ask your own questions on the forum.
Similar Threads
  1. SKhan3
    Replies:
    1
    Views:
    364
  2. Rob Walker
    Replies:
    9
    Views:
    1,181
    Rob Walker
    Sep 13, 2006
  3. dmitri

    Any experience with Gate 102 ATA

    dmitri, Jan 15, 2007, in forum: UK VOIP
    Replies:
    8
    Views:
    653
    dmitri
    Feb 11, 2007
  4. dmitri
    Replies:
    4
    Views:
    830
  5. richard

    Still more flack over the Gate's-gate affair

    richard, Jul 30, 2009, in forum: Computer Support
    Replies:
    7
    Views:
    629
    Evan Platt
    Aug 2, 2009
Loading...

Share This Page