Check that in Asterisk SIP Settings, External Address and Local Networks are correctly set.
Check that the LAN address of the PBX matches the address to which you are forwarding ports in the USG.
If using a trunk with registration, confirm that expiry or qualify frequency is keeping the NAT association open (UDP timeout in USG must be longer).
If the above doesn’t help, please post:
ISP? Modem make/model? Static public IP address?
Does USG get a public IP on its WAN interface? Trunking provider(s)?
What does caller hear on an attempted incoming call?
On a failed outgoing call, have you confirmed that outbound audio is also not present?