SPA3102 == 2 x UACs?

[Steve Brown]

Basic question: Can the SPA3102 be used as 2 independent UACs: one for FXS, one for FXO?

I'm trying to place Asterisk in between the FXS/FXO ports for call routing which mostly works...

PSTN -> Asterisk -> Line1 works a charm, including CID, only real trick was changing dialplan for PSTN Line (S0<:4801234567890@sipserver:5060>)

Line1 -> Asterisk -> PSTN not so much, I receive a secondary continuous tone (NOT PSTN dialtone) which is waiting for digits: wait 10sec and receive reorder, dial additional digits and they appear in debug and any additional digits dialed after receiving tone are passed to PSTN correctly; if I dial the original number a second time from this secondary tone it is routed via PSTN as expected.

What I really would like to see happen: SIP INVITE from Asterisk to PSTN port (sip:6021234567890@spa3102:5061) goes offhook FXO and passes SIP INVITE called party (6021234567890) number to FXO via DTMF digits, which seems reasonable as a VoIP-PSTN Gateway function

So do I have a fundamental functional misunderstanding of how SPA3102 should work? or is it config issue?

[Andrey Cassemiro]

Hi Steve,

Please can you tell me how is configure the field One Stage Dialing at the PSTN Line Tab?

It must be set to "yes" and the PSTN Line extension must be differente from the Dialed Number.

If your extension is 403 and the number starts with 403 it will make the SPA3102 start a two stage dialing that request you to insert the number.

If they are different and the OSD is set to yes the SPA must do what you expect.

I hope it help you.

Regards.

Andrey Cassemiro.