Asterisk IVR + SIP Setup

I have an Asterisk instance that needs a fully functional IVR linked to our SIP trunks. The core requirement is simple: callers should be greeted, routed through a concise menu, and seamlessly bridged to the correct internal extensions or external numbers. I already have the server and SIP credentials; what I need is your expertise in writing the dial-plan, recording or loading the voice prompts, and making sure calls flow reliably both ways.

Configuration must be clean, well-commented, and resilient so future menu changes are straightforward. Once everything is in place I’ll run live tests with you to confirm:

• IVR menu plays correctly and follows the designed call flow
• SIP trunk registers and handles inbound and outbound calls without drops or codec issues
• Basic documentation shows the dial-plan logic and any custom scripts or AGI you introduce

If you’ve deployed similar Asterisk IVR-SIP solutions, I’d love to see a brief example or two when you reply.