dmitri brought next idea :
> On Wed, 14 Feb 2007 20:03:37 +0000, DGB wrote:
>
>> In news:wZOdnd9m3vEwwk7YnZ2dnUVZ8qXinZ2d@eclipse.net. uk,
>> dmitri <me@my.biz> 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.