How NOT to dial with Sipura's dial plan ?

I have a SPA-3000 whose PSTN interface is connected to an analog PBX
extension, and the VOIP interface is SIP registered.

I'd like VOIP incoming calls to be answered and connected to the PBX
dialtone, so I managed to put something in the dialplan like <:>SO but
it doesn't work.

I solved the problem by automatically dialling some dummy digits that do
nothing harmless to the PBX, but the goal would be NOT to dial anything.

Has anyone a clue about that ?

 

I THINK I understand what you are getting at.
Normally the default dialtone would be for dialling the 'main'
provider, that is the one that is registered as your incoming
provider.
However, if you have working what you are wanting by dialling some
digits then there is an alternative - a 'warm' line.
In this case you can get the digits that you want to dial to be
dialled automatically. Then it will be up to the configuration of the
exchange to accept the numbers dialled after that.
Example:

If you want a 'warm line' do this:-
P4 <:12345>
Here, if no number is dialled in the first 4 seconds of 'off hook'
then 12345 is automatically dialled. Simply don't dial anything until
you are routed, then dial the number you want via your PBX.
If you want this to be routed via provider 4, on your gateway
settings, for example, then add <gw4> to make
P4 <:12345><gw4>
So, in summary, for this example, pick up the phone, wait 4 seconds,
the dial tone changes, then dial via your PBX.

 

I should also add that if you don't need to dial any digits then the
following will work.
P3, xx. <gw3>
Here, after 3 seconds, a new dial tone is presented (sounds like the
'unobtainable' tone). Then dial away. If you want you can put it all
together:-
P3, <0:0044>[1-2]xx. <gw3>
Here a UK number, starting 01 or 02, can be dialled, via gateway 3,
after wating for the second dial tone which is presented after 3
seconds of 'off hook'.

 

No, you didn't understand. I want NO dialling be made by the SPA-3000.
Dialling is to be done in audio DTMF after the (PBX) analog line
dialtone (or voice message, if any) is heard at the remote end (where
BTW there is a SPA-2100 and a cheap analog phone).

The correct answer was: <gw> which I found after some (many) trials.

This way, I have a PBX extension in a remote location and everything is
transparent for the user, including tones, voice messages and hookflash.
The drawbacks are:
- getting the dialtone takes 1~2 seconds
- CID is not transmitted
- ringing patterns are not transmitted (PBX ringing varies
depending wether the call is internal or from an inbound trunk)

But these are small annoyances. We'll try to improve this later (mainly
by setting direct peer-to-peer IP calls and not through a SIP
registrar).

Note: if you plan to do so, restrict the SPA-3000 access with the "Voip
caller ID pattern" field (in the PSTN tab), otherwise anybody calling
the gateway will access all your PBX features, including outbound
trunks.