We are looking for an experienced Dialer Engineer / SIP Engineer to set up and scale VoIP-based contact center solutions using Asterisk or similar technologies.
Requirements
- 6+ years of hands-on experience with Asterisk, Kamailio/OpenSIPS, FreeSWITCH, or similar VoIP systems.
- Deep knowledge of SIP protocol, RTP, WebRTC, STUN/TURN, and NAT-related issues.
- Experience designing and deploying large-scale dialer systems on cloud infrastructure.
- Solid experience in Linux system administration and shell scripting.
- Familiarity with SBCs (e.g., Acme Packet, Sansay, Audiocodes) and SIP debuggers (sngrep, Wireshark).
- Exposure to call center metrics, DNC lists, retry logic, concurrency management.
- Experience with REST APIs, MySQL/Postgres, and message queues (RabbitMQ/Kafka) is a plus.
- Bonus: Experience with voice biometrics, conversational IVR, AI-based call scoring.