I'll post some details about my setup (I guess it's quite similar to yours - I only have GSM gateway on FXO interface - you will have your phone line). I have incoming calls from GSM gateway going to user housephone - this is dummy LMCE user (won't show on floorplan), but it's purpose is to have only one "user" receiving all incoming calls - so I can go in web-admin and set all call routing features only for that user (I came to this solution for me, cause we're 4 in my family, but we all use 2-3 common phones, so making routes for each user is too complex for our situation)...
Basically you have to imagine sipura in your case as two devices :
1. from FXO port (your line) to VOIP - Asterisk gateway -this is basically representing SIP extension to Asterisk (240) - "PSTN" line in Sipura web interface (I have also under Dial plan #8 line : (<S0:401>) - that means that on incoming call, extension 401 will be called)
2. from FXS (you can attach phone or doorphone in my case) to VOIP Asterisk gateway (another sip extension - 209 in my case) - "Line 1" in Sipura web interface
You must setup those two (I think that Voxilla configurator will setup this for you already)...
Now on Asterisk (LMCE) side :
1. I have SIP trunk defined for sipura device :
put setting data for 240 "PSTN line" in Peer details. This trunk will be used for your outgoing routes (so call will go to your phone line)
2. extension 401 is custom extension defined only (no real device) and I have this line in there :
This device uses custom technology.
That means that whenever I receive GSM call from Gateway (this is equal to your phone call), call will be forwarded to extension 306. Extensions 3XX are not visible anywhere in FreePBX, but are created in Asterisk config files for each LMCE user. So extension 306 is extension for my previously mentioned user housephone (where all incoming calls go to).
What happens with calls on that extension, is determined by call routing setup you can do on web-admin - there you can specify behaviour regarding house security mode, Caller id, etc....
Comment: maybe calling 306 in first place instead of 401 in dialplan of Sipura first will also work, but haven't tried it...
3. outbound routes setup in Freepbx:
this where you determine how and where your outgoing calls with go. You can easily use your defined trunk as part of outbound routes...
Beware: this setup is done more in Freepbx than via web-admin, phone lines ,etc... works for me, cause I used Asterisk before and I want to setup few extra things in Freepbx (like more trunks in certain order for outgoing calls etc...), so maybe similar result can be achieved via web-admin in more proper way...