Senior VoIP Engineer – Asterisk & FreeSWITCH

We are looking for an experienced Senior Telephony / VoIP Engineer with strong expertise in Asterisk or FreeSWITCH-based telephony infrastructure. The ideal candidate should have hands-on experience in designing, deploying, and maintaining VoIP and SIP-based systems, and should be capable of integrating telephony platforms with modern backend systems. You will work closely with our engineering team to support an AI-driven voice automation platform, focusing primarily on building and maintaining the telephony layer.
Location: Remote / India (Nagpur preferred) Experience: 4–8 Years Employment Type: Full-time

Deploy, configure, and maintain Asterisk or FreeSWITCH servers in production environments.

Set up and manage SIP trunks with telecom providers such as Airtel, Jio, Tata, Exotel, Route, etc.

Design and implement dialplans, IVRs, inbound/outbound routing, and call flows.

Manage SIP signaling, SDP negotiation, and RTP media streams.

Configure codecs, transcoding, jitter buffers, and NAT traversal for optimized call performance.

Implement call recording, call bridging, failover routes, and CTI integrations.

Integrate telephony systems with backend services using WebSocket or gRPC audio streaming.

Implement programmatic call control using ARI/AGI/EAGI (Asterisk) or ESL/FS API (FreeSWITCH).

Monitor and troubleshoot SIP/RTP performance and call quality issues.

Ensure high availability, reliability, and performance optimization for telephony systems.

Minimum 4–8 years of hands-on experience with Asterisk or FreeSWITCH in production environments.

Strong understanding of SIP, SDP, RTP protocols.

Experience working with VoIP codecs such as G.711, G.729, and Opus.

Hands-on experience with dialplans, IVRs, call routing, and DID management.

Experience integrating SIP trunks with telecom carriers or CPaaS providers.

Strong knowledge of Linux environments such as Ubuntu or CentOS.

Ability to analyze SIP traces and debug telephony issues effectively.

Familiarity with scripting or programming using Python or Node.js.

Experience monitoring telephony systems using tools such as sngrep, tcpdump, Homer/HEP, and CDR analysis.

Strong problem-solving, debugging, and communication skills.

Python / Node.js

SIP Trunk Integration

WebSocket / gRPC

Telephony Monitoring Tools (sngrep, tcpdump, Homer)

CPaaS Platforms (Twilio, Exotel, Route Mobile, Plivo, Knowlarity)

Kamailio / OpenSIPS

Audio Processing Tools (GStreamer / FFmpeg)

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Senior Telephony / VoIP Engineer (Asterisk / FreeSWITCH)

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Senior Telephony / VoIP Engineer (Asterisk / FreeSWITCH)

We are looking for an experienced Senior Telephony / VoIP Engineer with strong expertise in Asterisk or FreeSWITCH-based telephony infrastructure...

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Senior Telephony / VoIP Engineer (Asterisk / FreeSWITCH)

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