Configure FreePBX HT813 PJSIP Routing

My FreePBX server is up and reachable; what I still need is a clean configuration that brings a Grandstream HT813 into the mix and routes calls smoothly between 1-2 analog handsets and my existing PJSIP extensions.

Scope
• Register the HT813 to FreePBX so the two analog phones plug in and work exactly like local extensions.
• Create and test dial-plan logic for the PJSIP side that supports call forwarding, on-demand and automatic call recording, voicemail boxes, and an IVR menu.
• Make sure analogue-to-VoIP and VoIP-to-analogue audio quality is solid, caller ID is preserved, and fails over gracefully if the ATA reboots.
• Provide brief notes on what you changed and where, so I can maintain the setup later.

I can give you SSH / GUI access at an agreed time; once the phones ring both ways and the feature set above is confirmed, we are done.