We are on the lookout for a VoIP/Asterisk Engineer to join our Engineering Department.
Responsibilities:
- Deploy and configure Asterisk 18+ (22 preferred) for test and production environments;
- Set up and maintain XCally / Motion integrations with Asterisk core;
- Manage SIP / PJSIP trunks, routing logic, and IVR workflows;
- Configure channel-based stereo recording (agent left / customer right);
- Enable WebRTC connectivity (WSS/TLS, ICE, STUN/TURN);
- Integrate Asterisk via AMI / ARI and maintain CDR / CEL data pipelines;
- Troubleshoot call-flow, audio, and codec issues;
- Ensure security and monitoring using TLS, Fail2Ban, Prometheus, and Grafana.
Requirements:
- Solid experience with Asterisk 18+ (22 preferred);
- Strong understanding of SIP / PJSIP, RTP / SRTP, and NAT principles;
- Experience with XCally / Motion and FreePBX;
- Practical knowledge of Docker / Linux-based systems;
- Understanding of WebRTC signaling and browser endpoints;
- Familiarity with AMI / ARI APIs and CDR / CEL integration;
- Scripting skills in PHP / Bash;
- Strong VoIP troubleshooting background;
- Knowledge of VoIP billing, E.164 number formatting, and security practices.
Nice to Have:
- CRM / CSP integrations, WebRTC SDK (JS/TS), cloud environments (AWS, Hetzner, GCP)
We will tell you more about all the benefits on the interview :)
This position is planned to be created (promising).