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  • Posted 4 months ago
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Job Description

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

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Job ID: 131564791