VoIP Engineer (Asterisk / FreePBX) — Telephony Migration to Git Management, Turnkey Project

City and office address: Fully remote, candidate location is not important Format: Remote Employment: Part-time / Project-based Salary range: $25–50/hour (net) during the system acceptance phase; afterwards — fixed price for agreed-upon stages Time zone: Europe ±2 hours from CET — preferred

About the Project

Telecommunications business in France: paid consultation lines and office telephony (~30 internal numbers). The system has been operational for many years but was maintained by a single person, is manually managed, and is poorly documented.

The task is to take over the infrastructure and prepare it for handover to a new specialist. You will not be responsible for support after the project: the team should implement planned changes via Git following a clear process, and all information should remain in the repository and instructions, not in the engineer's head.

Tech Stack

Asterisk/FreePBX, Kamailio, FastAGI with routing logic via MariaDB, SIP trunks from multiple operators, fail2ban, FOP2, Zabbix, Proxmox virtual machines.

What Needs to Be Done

Stage 1 — System Takeover (current specialist has limited availability): — Accept access credentials, configurations, routing rules, and common incident procedures; — Create a telephony map: what works where and what it's responsible for.

Stage 2 — Migrate Management to Git: — Store FreePBX/Kamailio configurations, dialplan, and FastAGI scripts in Git with clear comments; — Prepare a staging environment for testing changes before production deployment; — Establish a process: branch → staging → deploy → rollback with a single command; propose the automation level yourself; — Write runbooks for typical operations and incidents for an engineer unfamiliar with the system; — Resolve causes of chronic problems: office IPs being banned by fail2ban, certificate expiration issues, disk full due to recordings.

Acceptance Criteria

A specialist who did not participate in the project should be able, using only the repository and instructions, to independently: — Add a new number; — Change the IVR; — Upload an audio prompt; — Restore service after a simulated failure in the staging environment.

Without your assistance. If they cannot, the project is not accepted, and further revisions are required.

Requirements

— 3+ years of experience with Asterisk/FreePBX in production: dialplan, AGI/FastAGI; — Confident SIP diagnostics using sngrep/tcpdump; — Linux, systemd, fail2ban, certificates, MariaDB; — Git is mandatory; Ansible or other IaC is desirable; — Ability to write clear instructions — half of the result; — Kamailio — strong plus.

Conditions and Payment

The project is divided into small parts with fixed prices and verifiable results.

Stage 1 is paid hourly. Result: telephony map and breakdown of the project into 6–10 parts with prices. The readiness of each part should be verifiable within 15 minutes.

Other parts are paid upon acceptance. 15% of the total amount — after the first month of independent operation by our side.

All access credentials and secrets will be stored in the company's vault, and code from day one will be in our GitLab. Payment will be made to an EU legal entity via invoice or directly to the freelancer.

Selection Process

Short call and 2–3 practical case studies based on real incidents: SIP trace analysis, PBX failure, fail2ban issues.

In your application, please provide:

1. The most complex Asterisk case you've resolved and how you did it;
2. How you would organize FreePBX management via Git for team operation.

Company Name: Eastern Country Digital Services s. r. o. Contacts: @tarasenko_r