#vacancy #VoIP #Asterisk #FreePBX #Kamailio #SIP #Linux #remote #projectwork

VoIP Engineer (Asterisk / FreePBX) — Git-managed telephony migration, turnkey project

City and office address: fully remote, candidate location not important Format: remote Employment: part-time / project-based Salary range: $15–30/hour (net) during system acceptance phase; then — fixed fee for agreed stages Time zone: Europe ±2 hours from CET — desirable

About the project

Telecom business in France: paid consulting lines and office telephony (~30 internal numbers). The system has been running 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 transfer to a new specialist. You will not be responsible for support after the project: the team should be able to make 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 several operators, fail2ban, FOP2, Zabbix, Proxmox virtual machines.

What needs to be done

Stage 1 — Take over the system (current specialist has limited availability): — Accept access credentials, configurations, routing, and typical incidents; — Create a telephony map: what works where and what it is responsible for.

Stage 2 — Transition management to Git: — Place 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 degree of automation yourself; — Write runbooks for typical operations and incidents for an engineer unfamiliar with the system; — Resolve the causes of chronic problems: office IPs being banned by fail2ban, certificate expiration, disk full due to recordings.

Acceptance Criteria

A specialist who did not participate in the project should be able to independently: — add a number; — change IVR; — upload a voice prompt; — restore service after a simulated failure on the staging environment.

Using only the repository and instructions. Without your help. If not successful — the project is not accepted, further work is 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 — desirable; — Ability to write clear instructions — half of the result; — Kamailio — a strong plus.

Conditions and Payment

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

Stage 1 is paid hourly. Result: a telephony map and breakdown of the project into 6–10 parts with prices. Each part's readiness 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 are in the company's vault, code from day one — in our GitLab. Payment to an EU legal entity via invoice or directly to a freelancer.

Selection Process

Short call and 2–3 practical cases based on real incidents: SIP trace, PBX failure, fail2ban.

In your application, please write:

1. the most challenging Asterisk case you've encountered and how you solved it;
2. how you would organize FreePBX management via Git for team collaboration.

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