VoIP Software Engineer (Asterisk and FreeSWITCH Development)

Specialization: Asterisk, FreeSWITCH, Kamailio and SIP

Employment Type: Full Time

Department: Engineering / Telecom Software Development

Experience: 3+ years preferred

Location: Remote

About the Role

We are looking for an experienced Senior VoIP Software Engineer with strong hands on development experience in Asterisk, FreeSWITCH, and Kamailio to build, enhance, and maintain scalable VoIP and telecom communication platforms.

The ideal candidate will have a strong understanding of SIP signaling, call routing, real time communications, telephony application development, and the integration of VoIP platforms with backend applications and APIs.

Key Responsibilities

• Develop, customize, and maintain VoIP applications using Asterisk and FreeSWITCH.

• Configure and develop SIP routing logic, registrations, proxies, and load balancing using Kamailio.

• Build and maintain IVR systems, automated dialers, call queues, call recording, conferencing, and advanced call routing solutions.

• Integrate telephony platforms with CRM systems, REST APIs, databases, and third party communication services.

• Work with Asterisk ARI, AMI, AGI and FreeSWITCH ESL to develop call control and event driven applications.

• Troubleshoot SIP signaling, RTP media, call quality, audio issues, and connectivity problems.

• Optimize system performance, scalability, security, and reliability for high call volumes.

• Deploy, monitor, and maintain VoIP services on Linux based environments.

• Collaborate with the engineering team to design new telecom features and improve existing communication infrastructure.

• Maintain technical documentation, automated tests, and deployment procedures.

Required Technical Skills

Core Technologies

• Asterisk PBX development and integration

• FreeSWITCH development and integration

• Kamailio SIP server and routing configuration

• SIP, RTP, SDP, DTMF and VoIP call flows

• SIP trunking, gateways, IVR, dialers and call routing

Programming and APIs

• Strong proficiency in at least one backend language, preferably Python, PHP, JavaScript/Node.js, or C/C++

• Asterisk ARI, AMI, AGI and FreeSWITCH ESL

• REST APIs, WebSockets and event driven programming

• SQL databases such as PostgreSQL or MySQL

Infrastructure and Networking

• Linux server administration

• TCP/IP, UDP, DNS, TLS and SRTP

• Docker and Git

• SIP debugging tools such as Wireshark, sngrep and tcpdump

• Experience with production deployments, monitoring and troubleshooting

Preferred Qualifications

• Experience integrating Kamailio with Asterisk or FreeSWITCH.

• Experience building carrier grade or multi tenant VoIP platforms.

• Familiarity with WebRTC, cloud infrastructure, high availability, and load balancing.

• Experience with call detail records (CDRs), billing systems, and telecom reporting.

• Previous experience in telecom, wholesale VoIP, SIP trunking, or contact center platforms.

What We Are Looking For

A hands on engineer who understands both the software development and telecom infrastructure sides of VoIP. The candidate should be capable of developing custom functionality, solving complex call routing issues, and building reliable communication systems that can scale with business requirements.