So another piece of my Asterisk/TrixBox puzzle was completed today -- or rather, almost completed. I received the Linksys SPA-3102 FXO/FXS SIP ATA, which will be the bridge between Asterisk and one of the incoming POTS lines to the lab. I probably should have ordered the SPA-3000, since I really don't need the routing/NAT capabilities present in the SPA-3102, but then again, it's a lab. I have it working inbound and outbound with Asterisk, and it's also driving a few analog phones that are referenced as a single addressable Asterisk extension. It was a bit of a struggle, but not much. Some notes on this follow:
The Asterisk server is set under Proxy, and the username/password is referenced in the SPA-3102 PSTN Line Subscriber Info section. You will show a failed registration on the PSTN line, but as long as you've flagged
Make Call without Regand
Ans Call without Regas
Yes, it will work. I'd rather have this setup than cluttered extensions on Asterisk that aren't used.
(S0<:4155551212)as Dial Plan 2 under PSTN Line with a corresponding flag to use DP2 under PSTN-to-VoIP Setup will generate a 4155551212 DID to Asterisk for an incoming call. Obviously, you can put whatever you want in that field.
(*x|*xx|xxx|11|0|00|[2-9]xxxxxx|1xxx[2-9]xxxxxxS0|xxxxxxxxxxxx.)This speeds up recognition of internal extension and feature code dialing, as well as just about everything else. With this dialplan, calls from the FXS port are routed to Asterisk, which has outbound routing configured to prefer the SIP trunks over the PSTN lines for long-distance, and the PSTN lines for all local calling, including emergency dialing.