I've got a way to get the
SPA-3000 to use the FXO port to take inbound from PSTN (grabs and passes telco caller-ID name/num as well) and pass to Asterisk for add'l handling.
Sure, the
SPA-3000 does a great job of 'front-ending' inbound PSTN calls, and can even pass-through to the built-in FXS port, or external VoIP service, but I needed Asterisk to get the call BEFORE it was "answered" and handled/routed by the
SPA-3000.
Would seem to be a simple mode of operation, yet everywhere I looked it didn't seem possible to do just that.
I wanted to use it as a 'simple' FXO <-> SIP gateway to Asterisk AND also use the FXS port as an Asterisk extension.
Here's how:
(I'm only detailing the tricky part .. the rest is really basic Asterisk and/or
SPA-3000 setup)
1. Setup Asterisk and
SPA-3000 so both the PSTN line (FXO) and Line1 (FXS) are registered with Asterisk as different extensions (i.e. FXO user ID=10 and FXS user ID=2000) on different ports (5060/5061).
In this example I'll use Asterisk extension "99" as the place I want to send the inbound PSTN call to.
2. PSTN Line tab:
PSTN-To-VoIP Gateway Setup
PSTN-To-VoIP Gateway Enable: NO
PSTN Ring Thru Line 1: YES
PSTN CID For VoIP CID: YES
(here's one of the tricks to make it work)
PSTN CID Number Prefix: A
(I used 'A' but I suppose you could pick any ALPHA character that WOULDN'T be expected as a valid caller-ID NUMBER)
FXO Timer Values (sec)
PSTN Ring Thru Delay: 3
3. User 1 tab:
Selective Call Forward Settings
Cfwd Sel1 Caller: A*
Cfwd Sel1 Dest: 99
4. In Asterisk (in the context that you've defined exten 99):
exten => 99,1,SETCIDNUM(${CALLERIDNUM:1})
exten => 99,2,Dial(SIP/${exten})
(for example)
Here's what happens:
Call rings FXO port.
Wait three seconds so that caller-ID gets sent (you might need to increase this, but 3 secs seems to work fine for me) to the
SPA-3000.
PREFIX the caller-ID NUMBER with a LETTER before passing it to LINE 1
(so if original caller-ID was 5559991212, it's now A5559991212, not a 'valid' caller ID number, but
SPA-3000 and Asterisk don't seem to care, thankfully).
SELECTIVELY forward ONLY calls with caller-ID NUMBER that begin with A (actually this should be EVERY inbound PSTN call) to Asterisk extension 99
As soon as Asterisk gets the call, STRIP the 'invalid' A off and we're left with a good, original callerID number. Send the call out to a device (can be the
SPA-3000 FXS (exten 2000) or port if you want!)
The call is still UNANSWERED at this point.
FXS port starts to ring, and original PSTN-provided caller-ID is sent as usual.
Answer extension 99 (or send it voicemail) and FXO finally goes off-hook.
You can make calls to extension 2000 and not worry about them being bounced back to extension 99 since no "normal" caller-ID NUMBER should ever (??) start with "A"
Above all, I think this could be made a whole lot more intuitive and fool-proof if Sipura just added a feature into the firmware.
SPA-3000 FXO gateway to Asterisk (without 'answering' first) 
Linear Mode
