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?
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