Join the Innovation at Moon Technolabs !!

We're looking for a skilled Sr. Asterisk / FreeSWITCH VoIP Developer to build, optimize, and manage scalable VoIP communication systems while ensuring high performance and reliable connectivity.

Asterisk / FreeSWITCH VoIP Developer - 5+ Yrs

Location: Sola, Ahmedabad

Work Mode: Onsite Only

Primary Responsibilities:

- Troubleshoot SIP signaling, RTP media flow, one-way/no-audio issues, codec negotiation, jitter, latency, packet loss, and NAT traversal problems.
- Perform advanced VoIP and network troubleshooting using tools like sngrep, tcpdump, Wireshark, RTP analysis, traceroute, ping, and network monitoring tools.
- Design, develop, and maintain scalable VoIP and telephony solutions using Asterisk and/or FreeSWITCH.
- Configure and manage SIP trunking, dial plans, IVR systems, call routing, conferencing, and PBX features.
- Develop and optimize real-time voice communication systems for high availability and low latency.
- Integrate telephony systems with CRMs, APIs, third-party services, and AI-based voice platforms.
- Troubleshoot SIP signaling, RTP media flow, NAT traversal, codec negotiation, and call quality issues.
- Implement and maintain WebRTC-based communication solutions and SIP over WebSocket (WSS).
- Develop custom modules, AGI/ESL scripts, and automation tools for telephony workflows.
- Monitor system performance, identify bottlenecks, and optimize server scalability and reliability.




- Work with DevOps and infrastructure teams for deployment, monitoring, and production support.
- Participate in requirement analysis, architecture planning, estimation, and technical documentation.
- Mentor junior developers and support technical decision-making across VoIP projects.
- Ensure system security, failover mechanisms, and compliance with communication standards.

Technical Requirements:

- Solid hands-on experience with Asterisk, FreeSWITCH, FusionPBX, or FreePBX.
- Deep understanding of SIP, RTP, SRTP, WebRTC, ICE, STUN/TURN, and NAT traversal.
- Experience in configuring SIP trunks, SBCs, gateways, IVRs, queues, and call center solutions.
- Proficiency in scripting/programming using Python, Lua, PHP, Node.js, or Bash.
- Experience with ESL (Event Socket Library), AGI, AMI, or ARI integrations.
- Knowledge of VoIP troubleshooting tools such as sngrep, tcpdump, Wireshark, and RTP analysis.
- Experience with Linux server administration (Ubuntu/CentOS/Debian).
- Familiarity with Kamailio/OpenSIPS and SIP proxy concepts is a plus.
- Experience integrating telephony with cloud providers like Twilio, Telnyx, Plivo, or similar.
- Knowledge of databases such as MySQL/PostgreSQL and API integrations.
- Understanding of containerization and deployment using Docker/Kubernetes is preferred.
- Familiarity with CI/CD and monitoring tools for production systems.

📌 Sr. Asterisk / FreeSWITCH VoIP Developer (Junagadh)
🏢 Moon Technolabs
📍 Junagadh