Asterisk IVR Setup for SIP/PRI

I need an experienced telephony engineer to bring up a new IVR on our Asterisk-based server that will answer calls coming in on both a SIP trunk and a PRI. The core requirement is a clean, dependable call-routing tree—callers should reach the right destination every time—with call recording and basic, built-in reporting turned on from day one.

Environment
• Linux box already running Asterisk (remote SSH available)
• One SIP trunk + one PRI (credentials and circuit details ready)

Scope of work
1. Configure the IVR in Asterisk, activate call routing options, and confirm that both trunks follow the same logic.
2. Enable call recording for all menu paths, store files locally in an organised directory structure, and verify playback.
3. Turn on the stock call‐detail reporting in Asterisk so I can pull simple traffic and recording stats without third-party add-ons.
4. Stress-test inbound and transfer legs, then tidy up dial-plan code for maintainability.

Optional enhancements such as layered menu navigation or voicemail drop boxes can be left commented in the dial-plan so I can expand later, but they are not part of the immediate go-live.

Acceptance criteria
• Dial-plan reloads without errors.
• Calls from both trunks reach the test extensions through the IVR in under two rings.
• Recordings appear in the defined folder with correct time-stamps.
• CDR reports show trunk, DID, duration, and recording link.

I will grant remote access as soon as we agree on the implementation steps and timeline, and I’m ready to move quickly once you’re set.