#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; then — fixed price per agreed milestones 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 operating for many years but was maintained by one person, is manually managed, and is almost undocumented.

The task is to take over the infrastructure and prepare it for transfer to a new specialist. After the project, you will not be responsible for support: the team should 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 — System Takeover (current specialist has limited availability): — Accept accesses, configs, routing, and typical incidents; — Create a telephony map: what works where and what it's responsible for.

Stage 2 — Migrate Management to Git: — Place FreePBX/Kamailio configs, 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; — Eliminate the causes of chronic problems: office IPs banned in fail2ban, expiring certificate, disk space full with recordings.

Acceptance Criteria

A specialist, not involved in the project, solely using the repository and instructions must be able to: — add a number; — change IVR; — upload a voice prompt; — restore service after a simulated incident on the staging environment.

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 the result; — Kamailio — 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: telephony map and breakdown of the project into 6–10 parts with prices. Each part's readiness should be verifiable in 15 minutes.

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

All accesses 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 on real incidents: SIP trace, PBX failure, fail2ban.

In your application, please write:

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 collaboration.