VoIP Software Engineer (Asterisk and FreeSWITCH)

Company Description Xinix is a global telecommunications and communications technology provider delivering reliable, scalable, and cost-effective solutions for homes, businesses, resellers, and wholesale partners worldwide. The company offers a unified platform that brings together Voice, Data, SMS, Mobile SIMs, Global Phone Numbers, Cloud Communications, SIP Trunking, Broadband, CRM Integrations, customer engagement tools, and AI-powered solutions. Xinix helps organizations simplify communication, enhance customer experiences, expand internationally, and improve operational efficiency. Its portfolio includes cloud phone systems, data connectivity, wholesale voice, phone numbers, mobile services, SMS and engagement solutions, and white-label options for resellers. With worldwide reach and a focus on intelligent, connected communication, Xinix supports businesses wherever they operate.
Role Description This is a full-time remote role for a VoIP Software Engineer specializing in Asterisk and FreeSWITCH. The role involves designing, developing, and maintaining VoIP platforms and services, including dial plans, call routing logic, SIP trunk configurations, and integration with Xinix’s cloud communication solutions. The engineer will implement new features, optimize system performance, troubleshoot complex VoIP issues, and ensure high availability, reliability, and security across the telephony infrastructure. Daily tasks include writing clean, maintainable code, configuring and tuning Asterisk/FreeSWITCH, collaborating with cross-functional teams (network, DevOps, support, product), and contributing to architecture decisions. The role also includes documenting technical solutions, participating in code reviews, and supporting continuous improvement of VoIP services and related tooling.
Qualifications

• Strong foundation in Computer Science concepts, including algorithms, data structures, networking fundamentals, and operating systems.
• Experience in Software Development and Programming, with proficiency in at least one relevant language (e.g., C, C++, Python, Java, or Go) and familiarity with software engineering best practices.
• Knowledge of Back-End Web Development, including APIs, microservices, RESTful services, and integration with cloud-based platforms.
• Solid understanding of Object-Oriented Programming (OOP) principles and ability to design modular, maintainable, and scalable systems.
• Hands-on experience with Asterisk and/or FreeSWITCH, SIP protocols, RTP, codecs, and VoIP signaling and media handling.
• Familiarity with Linux administration, shell scripting, and deployment automation (e.g., Docker, CI/CD pipelines) is highly beneficial.
• Experience with database technologies (SQL and/or NoSQL) and configuration management for telephony platforms.
• Ability to diagnose and resolve complex VoIP issues using relevant tools (logs, packet captures, monitoring systems) and to work effectively in a remote, distributed team