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Jatis Mobile

Software Engineer (VoIP & Asterisk)

2-4 Years
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Job Description

Job Description

  • Develop, maintain, and enhance telephony applications and services based on Asterisk and VoIP technologies.
  • Design and implement integrations between telephony platforms and internal/external applications using APIs, AMI, ARI, AGI, WebSocket, and related technologies.
  • Build backend services to support voice automation, AI Voice Bot, Call Center, and communication platforms.
  • Troubleshoot and optimize SIP signaling, RTP media flow, call routing, and telephony performance issues.
  • Collaborate with Product, Engineering, and Infrastructure teams to deliver scalable voice communication solutions.
  • Develop and maintain REST APIs, microservices, and system integrations related to telephony services.
  • Participate in system architecture discussions and contribute to technical design decisions.
  • Create and maintain technical documentation, deployment procedures, and operational guidelines.
  • Support deployment, testing, monitoring, and maintenance of production telephony environments.
  • Research and implement new technologies related to AI Voice, Speech Processing, and modern communication platforms.

Job Requirements

  • Minimum 2 years of experience as Software Engineer, Backend Engineer, VoIP Developer, Telephony Developer, or similar role.
  • Strong proficiency in at least one programming language: Golang
  • Experience developing REST APIs and backend services.
  • Experience working with Linux-based environments.
  • Understanding of software development lifecycle, version control, and deployment processes.
  • Familiar with SQL and/or NoSQL databases.
  • Understanding of SIP, RTP, VoIP, and telephony concepts.
  • Experience working with Asterisk and related technologies.
  • Able to perform troubleshooting and root cause analysis independently.
  • Good communication and problem-solving skills.

Preferred Qualifications

  • Experience integrating Asterisk using AMI, ARI, AGI, or FastAGI.
  • Experience with SIP Trunk, IP Phone, Call Center, or PBX platforms.
  • Experience building telephony, contact center, or communication platforms.
  • Familiar with WebRTC, FreeSWITCH, Kamailio, or OpenSIPS.
  • Experience integrating AI Voice Agent, Speech-to-Text (STT), Text-to-Speech (TTS), or conversational AI solutions.
  • Familiar with Docker, Kubernetes, RabbitMQ, Redis, or microservices architecture.
  • Understanding of telecommunication infrastructure and enterprise voice solutions.

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About Company

Job ID: 149346029