Asterisk SIP Infrastructure Setup

I have most of the mobile application ready; what I still need is a solid Asterisk-based voice layer that simply works every time a user places or receives a call. My priority is configuration and integration: connecting SIP trunks, wiring up DID numbers, mapping inbound and outbound routes, and ensuring that every two-way call is recorded and pushed securely to my chosen cloud storage bucket.

The core stack will run on a Linux server with Asterisk at its heart. You’ll handle the dial-plan logic, trunk registration, codec selection, security hardening, and the APIs that let the mobile app trigger voice calls and fetch recording links. I already know the basic architecture I want, so we can move quickly into hands-on setup, iterative testing, and final roll-out.

Deliverables:
• Fully configured Asterisk PBX with SIP trunks and DIDs operational for voice calls
• Reliable two-way call recording automatically uploaded to cloud storage with secure access controls
• API or webhook endpoints the mobile app can call to initiate sessions and retrieve recording URLs
• Documentation of configurations, commands, and any custom scripts for future maintenance

Successful completion means a user can tap “Call” inside the app, the call routes through our infrastructure, is recorded end-to-end, and the recording becomes instantly available through a secure cloud link. If you have proven experience doing exactly this, let’s get started.