On 03-07-2007, T i m <news@spaced.me.uk> wrote:

> On Tue, 03 Jul 2007 19:19:48 +0100, Jono <nothanks@blueyonder.invalid>
> wrote:
> 
>
> And is that not typical of other similar spec (free local number /
> free incoming PSTN / SIUP calls) SIP providers Jono?

Not paticularily. Voiptalk, Gradwell and Voipfone don't block SIP-to-SIP
calls. Vonage and Sipgate do. It does not correlate with freeness.

> So, if I went with 'another' SIP provider I guess I would get two
> 'things' from my registration, a STD number for ordinary PSTN users to
> call me on and a number linked 'address' for some (all) SIP clients to
> use?

The important item is the SIP address, user@domain. Without it you are
not contactable. At some point a PSTN number has to point to it. The
phone call would not be delivered otherwise.

> How would someone with a stand alone SIP phone call me over VoIP, I
> mean, what would they enter on their analogue phone if I only had an
> address?

This depends on the ATA used and is not straightforward. If your Sipgate
ID was 1112223 I would need to dial 111222333@sipgate.co.uk, and the
letters present a problem with an analogue phone. It can be done if the
ATA has a quick dial or address book facility. Or, rather tediously,
dial 111222333*217*10*79*23 from the handset. * replaces @ and . on my
ATA.

--
Brian

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