FreePBX Server Configuration

FreePBX / Asterisk Configuration Expert Needed We have a Hostinger VPS with FreePBX already installed and accessible. We need an experienced VoIP engineer to configure it for integration with our SaaS CRM. Scope Secure and optimize FreePBX for production Configure PJSIP/WebRTC for browser calling Create test extensions (101, 102, 103) Configure ARI (Asterisk REST Interface) Configure call recording Configure TLS, SRTP, NAT, RTP, codecs (Opus/G711) Prepare the server for an Orange Morocco SIP Trunk (credentials will be provided later) Configure inbound/outbound routes once SIP credentials are available Provide basic documentation (extensions, ARI credentials, firewall/ports) Important This is not a PBX deployment for end users. The PBX will be controlled programmatically by our CRM through ARI/API. Experience with Asterisk, FreePBX, WebRTC, PJSIP, ARI, SIP Trunks, and production VoIP deployments is required. Bonus: Experience with Orange SIP, GSM gateways (OpenVox/Dinstar/Yeastar), or CRM telephony integrations.