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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-486/48with the following settings
FXS Impedance: CTR21(270R+750||150nF)
Onhook Voltage: 48V
Caller ID Scheme: ETSI-FSK
DTMF Send Type: Via SIP INFO

In news:. uk,
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

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

> In news:. uk,
> 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/forum...li-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 brought next idea :
> On Wed, 14 Feb 2007 20:03:37 +0000, DGB wrote:
>
>> In news:. uk,
>> 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/forum...li-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.

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

On Feb 15, 12:35 am, "DGB" <m...@ntlworld.invalid> wrote:
> Innews: d,
> Jono <notha...@blueyonder.invalid> 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 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.

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

> On Feb 15, 12:35 am, "DGB" <m...@ntlworld.invalid> wrote:
>> Innews: d,
>> Jono <notha...@blueyonder.invalid> 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