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Senior AI Voice & VoIP Platform Engineer (Asterisk / FreeSWITCH / OpenSIPS)

Xinix.io

Remote
Remote Senior 🇬🇧 English
Asterisk FreeSWITCH OpenSIPS SIP RTP TCP/IP UDP NAT traversal QoS Linux administration Bash Python Ansible CI/CD pipelines AI/ML voice services speech recognition IVR call analytics

Job description

About the role

We are looking for a Senior AI Voice & VoIP Platform Engineer to design, build and optimise scalable, AI‑enhanced telephony solutions for our global clients. The role is fully remote and focuses on SIP‑based communication, real‑time media handling and high‑availability voice infrastructure.

Key responsibilities

  • Design, implement and maintain Asterisk, FreeSWITCH and OpenSIPS clusters in cloud or hybrid environments.
  • Integrate AI/ML services such as speech‑to‑text, IVR and call analytics into the telephony stack.
  • Develop automation scripts and CI/CD pipelines for deployment, configuration and monitoring.
  • Troubleshoot complex SIP signalling, media, networking and performance issues.
  • Implement monitoring, alerting and incident response processes to ensure platform reliability.
  • Collaborate with software engineers, DevOps and product teams on architectural decisions and documentation.

Required profile

  • Strong hands‑on experience with Asterisk, FreeSWITCH and OpenSIPS.
  • Deep understanding of SIP, RTP, TCP/IP, UDP, NAT traversal and QoS for real‑time communications.
  • Proven track record of supporting production VoIP environments and performing root‑cause analysis.
  • Experience designing scalable, secure and highly available VoIP architectures in cloud or hybrid settings.

Required skills

  • Linux system administration
  • Bash and Python scripting
  • Ansible automation
  • CI/CD pipeline tooling
  • AI/ML voice integration (speech recognition, IVR, call analytics)

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Published 1 month ago

Expires 3 days from now

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