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We're on the lookout for a Telephony Engineer with solid experience in Asterisk and Kamailio to design, build, and scale our SIP-based voice infrastructure. The ideal candidate will collaborate with cross-functional teams to ensure robust telephony services, seamless integration with microservices, and high-availability architecture.
Join us in shaping the future of AI-powered communication.
Location: Gurugram (On-site preferred)
Experience: 3+ years
Apply now: [Confidential Information] | 9004735381
Check out the complete job description below:
Key Responsibilities:
- Design, configure, and maintain Asterisk PBX systems, including dialplan development (PJSIP, ARI/AMI)
- Implement and manage Kamailio SIP routing, dispatcher, and failover configurations
- Develop and optimize call routing, IVR systems, queue management, and call recordings
- Troubleshoot SIP and RTP issues using tools like Sngrep, Tcpdump, and Wireshark
- Integrate telephony APIs with Node.js microservices for dynamic call handling
- Architect and implement high-availability, multi-tenant telephony solutions
- Collaborate with development and infrastructure teams to ensure system scalability and reliability
Key Requirements:
- 3+ years of experience working with Asterisk (PJSIP, ARI, AMI) and Kamailio SIP server
- Strong understanding of SIP, RTP, WebRTC protocols
- Proficiency in SIP routing, failover, and dispatcher configuration
- Hands-on experience with telephony troubleshooting tools (sngrep, tcpdump, Wireshark)
- Experience with telephony API integration in Node.js environments
- Knowledge of Linux server administration and scripting
- Familiarity with Docker, Kubernetes, PostgreSQL, Redis
- Excellent problem-solving and communication skills
Job ID: 131564791