Job Summary
We are seeking an experienced specialist who can design, develop, and maintain Asterisk/FreePBX telephony systems for contact center and conferencing environments, while also building APIs for call routing, call bridging, conferencing control, and integrations.
This role combines VoIP engineering, contact center architecture, conferencing, and backend/API development.
Key Responsibilities / Duties
- Asterisk / FreePBX Engineering (Contact Center & Conferencing)
â Architect, deploy, and manage Asterisk/FreePBX for high-volume contact center and conferencing use cases.
â Configure
â IVRs, ACD, skill-based routing, multi-level queue structuresâ Outbound dialer workflows (predictive, preview, progressive)â Call conferencing (multi-party conferences, PIN management, moderator controls, dynamic conference rooms)â Whisper, barge, monitor, recording, quality monitoringâ Optimize call flows for low latency and high concurrency.
- API Development & Call Automation
Design, Build, And Maintain REST APIs For
â Call bridging (agentâcustomer, multi-party, warm transfers, blind transfers)â Conference creation, joining, and controls (mute/unmute, kick, lock/unlock, recording control)â Click-to-call & CRM-triggered callsâ Dynamic call routing & IVR adjustmentsâ Agent login/logout, pause/unpauseâ Call event streaming (webhooks)â Conferencing analytics (participant count, duration, event logs)
Work With Asterisk Interfaces Such As
â ARI (Asterisk REST Interface)â AMI (Asterisk Manager Interface)â AGI (Asterisk Gateway Interface)
â Integrate Telephony Systems With
â CRM platforms (Salesforce, Zoho, Freshdesk, Dynamics, HubSpot)â Conferencing portals or custom meeting management appsâ Ticketing systems, chat systems, workforce management toolsâ Reporting and analytics dashboards
â Sync conference metadata, call logs, queue stats, and recordings.
- Monitoring, Performance & Troubleshooting
â Ensure high availability, low jitter, low packet loss, and optimal VoIP performance.
â Troubleshoot SIP And RTP Issues Using
â sngrepâ tcpdumpâ Wiresharkâ Asterisk CLI logsâ Optimize codecs, transcoding performance, and server load.
â Monitor Real-time
â Queue performanceâ Conference performanceâ SLA metricsâ MOS score and call quality indicators
â Implement Security Measures For Voice And Conferencing Systems
â Fail2ban, IP filtering, SIP firewallingâ Anti-fraud and intrusion preventionâ Manage system upgrades, patches, backups, and DR plans.â Deploy HA clusters for mission-critical contact center + conferencing environments.
Required Skills & Qualifications
â 3â5+ years hands-on experience with Asterisk & FreePBX (contact center + conferencing).â Strong understanding of SIP, RTP, conferencing modules, bridge applications.â Proven experience building APIs and backend services (Node.js, Python, PHP, Go, etc.).â Hands-on experience with ARI, AMI, AGI, and dialplan logic.â Strong Linux skills (CentOS, Debian, Ubuntu).
â Experience Configuring
â Contact center queues and routingâ Conference bridgesâ Recordings, monitoring toolsâ Experience with SIP trunking and carrier integrations.
Preferred / Good-to-Have Skills
â Experience with large-scale conferencing platforms (Asterisk ConfBridge, MeetMe, or custom conferencing solutions).â Knowledge of OpenSIPS/Kamailio for scaling conferencing or call center environments.â Familiarity with WebRTC and browser-based conferencing.â Knowledge of predictive dialers like VICIdial/GoAutoDial.â Cloud deployment experience (AWS/GCP/Azure).â Familiarity with microservices and distributed architectures.
Soft Skills
â Strong analytical and debugging mindset.â Ability to document systems, flows, and API specs clearly.â Good communication and collaboration skills.â Ability to work in fast-paced contact center or conferencing environments.
