[PhoneBoy]

Being slightly paranoid about disclosure requirements, I had moved a thread into the SPA3000 forum that contained a legitimate question from muppetmaster. Since the thread itself might also contain other sensitive information now, here is the question with an answer:

How does the Sipura 3K allow one to forward calls coming in from the PSTN to be serviced either by a SIP provider (ie - FWD) or by an Asterisk instance? Can this be a viable alternative to a Zaptel modem with an Asterisk instance. Where you just run Asterisk and use the Sipura 3K as the interface to the PSTN? How would that analog interface be addressable for an Asterisk dial plan? Etc, etc.

On release, the SPA3000 should support a "hard coded" forwarding from SIP to PSTN as well as PSTN to SIP.

Basically, once you call in on your PSTN line and authenticate, you get "SIP dialtone" for lack of a better description. You can dial whatever number(s) are permitted by the dialplan associated with the PIN you used to authenticate. I assume the SIP endpoint can be associated with an Asterisk box. To go the other way (from Asterisk to PSTN), you can cause Asterisk to authenticate with the SPA3000 using HTTP Digest, which would allow a PSTN call via one-stage dialing.

-- PhoneBoy

[muppetmaster]

I am not sure if I follow.

Here is the scenario I am looking for:

I would like to avoid using an Asterisk instance with a Zaptel FXO card and instead use the Sipura 3000 a a compatible PSTN gateway. This would mean that the Asterisk would either need to register with the Sipura 3000 as a SIP end-point and pick-up any calls that come in on the PSTN, or that the Sipura 3000 would need to be able to forward all calls to a specific SIP URL. The Asterisk instance would then run it through a standard dialplan and deliver appropriate services.

Likewise I would like the Asterisk to be able to seamlessly send calls through the Sipura 3000 as the PSTN gateway.

Are these scenarios possible without delivering 'dialtone' which requires an additional set of digits? Thereby relegating the Sipura 3000 to a PSTN bi-directional gateway with the intellegence at the Asterisk instance?

[PhoneBoy]

An incoming PSTN call could theoretically be forwarded to any SIP address.

For outgoing PSTN calls, you would configure Asterisk to make a PSTN call in much the same way as you might configure it to make a call via a SIP provider, i.e. Asterisk would place the call as number@SPA3000-ip and provide authentication credentials in the process. In this case, there isn't a second dialtone to contend with.

[muppetmaster]

Originally Posted by PhoneBoy

An incoming PSTN call could theoretically be forwarded to any SIP address.

For outgoing PSTN calls, you would configure Asterisk to make a PSTN call in much the same way as you might configure it to make a call via a SIP provider, i.e. Asterisk would place the call as number@SPA3000-ip and provide authentication credentials in the process. In this case, there isn't a second dialtone to contend with.

Then I've got to get me one!

[PhoneBoy]

We're still selling the beta units I think.

[muppetmaster]

How likely are any hardware changes prior to the GA release? Also, when is the GA release anticipated?

[PhoneBoy]

I am unaware of any hardware changes that they are planning.

[MustardMan]

I don't see any 3000's for sale anywhere on this site. If they are please point me to the proper link

[PhoneBoy]

We are sold out of beta units. GA units are supposed to be available in the August timeframe, last I heard.