Hello
I have freepbx 16 with only one trunk (fxo grandstream gxw4108 192.168.1.aaa)
I have no problems with outging call, but i dont receive any incoming calls
The incoming calls are being handled by chan_pjsip, but you have only configured the obsolete chan_sip. When both exist, FreePBX prefers chan_pjsip for the standard SIP port 5060.
The preferred solution is to remove chan_sip and do all your SIP endpoints with chan_pjsip.
You will need to tell the other end of the trunk that they need to talk to port, I think, 5070, not the default port 5060. If you are registering with them, and they are sending to port 5060, their SIP implementation is broken (or you have a broken router).
In certain network configurations, it may be possible to bind the two to specific interfaces, but this will confuse anyone who maintains it, in future.
If it is really impossible for them to work with chan_pjsip, you need to provide detailed reasons, so the Asterisk developers know what they have to change, to make it work.