VoIP Software Engineer (Asterisk and FreeSWITCH Development)

We are seeking a Senior VoIP Software Engineer with extensive hands-on experience in Asterisk, FreeSWITCH, and Kamailio to join our remote engineering team. The successful candidate will play a key role in developing, enhancing, and maintaining scalable VoIP and telecom communication platforms. This position requires a deep understanding of SIP signaling, call routing, real-time communications, telephony application development, and integrating VoIP systems with backend applications and APIs. The ideal candidate will be comfortable working in a Linux environment and collaborating with cross-functional teams to deliver robust telecom solutions.

Key Responsibilities
- Develop, customize, and maintain VoIP applications using Asterisk and FreeSWITCH.
- Configure and develop SIP routing logic, registrations, proxies, and load balancing with 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.
- Utilize Asterisk ARI, AMI, AGI, and FreeSWITCH ESL for call control and event-driven application development.
- Troubleshoot SIP signaling, RTP media, call quality, audio issues, and connectivity problems.
- Optimize system performance, scalability, security, and reliability to support high call volumes.
- Deploy, monitor, and maintain VoIP services on Linux-based environments.
- Collaborate with engineering teams to design new telecom features and improve existing communication infrastructure.
- Maintain technical documentation, automated tests, and deployment procedures.

Required Qualifications
- Proven experience in Asterisk PBX development and integration.
- Strong knowledge of FreeSWITCH development and integration.
- Expertise in Kamailio SIP server configuration and routing.
- In-depth understanding of SIP, RTP, SDP, DTMF, and VoIP call flows.
- Experience with SIP trunking, gateways, IVR, dialers, and call routing.
- Proficiency in at least one backend programming language such as Python, PHP, JavaScript/Node.js, or C/C++.
- Familiarity with Asterisk ARI, AMI, AGI, and FreeSWITCH ESL interfaces.
- Experience working with REST APIs, WebSockets, and event-driven programming.
- Knowledge of SQL databases like PostgreSQL or MySQL.
- Strong Linux server administration skills.
- Understanding of TCP/IP, UDP, DNS, TLS, and SRTP protocols.
- Experience with Docker and Git version control.
- Proficiency in SIP debugging tools such as Wireshark, sngrep, and tcpdump.
- Demonstrated ability to manage production deployments, monitoring, and troubleshooting.

Preferred Qualifications and Benefits
- Experience integrating Kamailio with Asterisk or FreeSWITCH platforms.
- Background in building carrier-grade or multi-tenant VoIP systems.
- Familiarity with WebRTC, cloud infrastructure, high availability, and load balancing techniques.
- Knowledge of call detail records (CDRs), billing systems, and telecom reporting.
- Previous work experience in telecom, wholesale VoIP, SIP trunking, or contact center environments.

We are looking for a hands-on engineer who combines software development expertise with a solid understanding of telecom infrastructure. The ideal candidate will be adept at developing custom functionalities, resolving complex call routing challenges, and building reliable communication systems that scale with evolving business needs. This role offers the opportunity to work remotely with a dynamic team focused on delivering innovative VoIP solutions.