To see the SIP conversation in your SPA syslog, set PSTN Line -> SIP Debug Option to full. With luck, you’ll see some activity on attempted outbound calls.
I don’t understand your ‘sip show peers’ output at all. First, I’d expect extension 3000 to show there (assuming that it is registered ok and working), though this is likely unrelated to the trouble you’re having with the FXO. Also, I don’t know why you are seeing two entries for the trunk; my system shows only one.
This is the only SPA I have and it’s in production, so I don’t want to experiment with settings, but these are the ones that are working here:
I use the same name for General -> Trunk Name and Outgoing -> Trunk Name. Yours have different capitalization, though I have no idea whether that matters. I also use the same name as the SPA username (and also don’t know whether that matters). So, for example, I would have for PEER Details:
host=dynamic
username=PSTN_ChanSIP_Trunk
secret=3001
type=peer
And for USER Details:
secret=3001
type=user
context=from-trunk
Of course, with these settings you would change in the SPA, PSTN Line-> User ID to PSTN_ChanSIP_Trunk .
After these changes, restart Asterisk (or reboot the entire server). With luck, you should see for sip show peers:
PSTN_ChanSIP_Trunk/PSTN_ChanSIP_Trunk 192.168.168.173 D Yes Yes 5062 Unmonitored
and no other entries for the PSTN Line side of the device.