Job description for PBX Engineer (VoIP / SIP / Asterisk / Kamailio) at PT. Hama Smart Solution
## ■ Overview
We are building a large-scale, highly available telephony platform that integrates with mobile and desktop applications, supporting both call center operations and AI-driven voice communication.
In this role, you will design, develop, and operate a redundant PBX system using Asterisk and Kamailio, as well as implement call control logic in Python.
You will be responsible for ensuring high reliability (SLA 99.99%+) and scalability (10,000+ concurrent calls) in a real-time communication environment.
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## ■ Responsibilities
### 1. PBX / SIP Infrastructure Design & Development
- Design, build, and operate PBX systems using Asterisk
- Implement SIP routing using Kamailio (load balancing, failover)
- Design high availability architectures (Active-Active / Active-Standby)
- Optimize RTP/media handling for large-scale traffic
- Ensure system scalability to support 10,000+ concurrent calls
### 2. Call Control Logic Development
- Develop call control APIs using Python (e.g., FastAPI)
- Integrate with Asterisk via ARI / AMI / AGI
- Design and implement call flows (IVR, recording, transfer, queue handling)
- Handle real-time call events (webhooks, event-driven architecture)
### 3. Client Integration (Mobile / Desktop / WebRTC)
- Integrate telephony features with mobile apps (iOS / Android)
- Integrate with desktop applications (e.g., Electron-based softphones)
- Design connectivity with SIP clients and WebRTC
- Improve call quality (latency, jitter, packet loss)
### 4. System Reliability & Operations
- Maintain SLA of 99.99%+ availability
- Monitor call quality metrics (MOS, jitter, packet loss)
- Implement logging and tracing per call/session
- Troubleshoot and resolve real-time system issues
- Perform capacity planning and scaling strategies
### 5. AI & Voice Integration (Nice to Have)
- Integrate AI-based voice systems (e.g., real-time AI conversations)
- Build pipelines for call recording, transcription, and analysis
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## ■ Requirements
### Must Have
- Strong understanding of SIP / RTP / VoIP protocols
- Hands-on experience with Asterisk
- Experience with Kamailio or OpenSIPS for SIP routing
- Experience with Linux server environments
- Backend development experience in Python
- Solid understanding of networking (TCP/IP, NAT, firewall)
### Nice to Have
- Experience with WebRTC
- Experience with ARI / AMI / AGI
- Experience with FreeSWITCH
- Cloud infrastructure experience (AWS / GCP)
- Experience with Docker / Kubernetes
- Experience with real-time / low-latency systems
- Experience in call center systems or telecom platforms
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## ■ Preferred Qualifications
- Experience designing systems for 10,000+ concurrent calls
- Experience maintaining high availability systems (99.99%+ SLA)
- Experience in distributed systems and event-driven architectures
- Experience with observability (metrics, tracing, logging)
- Experience working on AI-driven voice or conversational systems
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## ■ Tech Stack (Example)
- PBX: Asterisk
- SIP Proxy: Kamailio
- Backend: Python (FastAPI)
- Infrastructure: AWS / GCP
- Client: iOS / Android / Electron
- Protocols: SIP / RTP / WebRTC
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## ■ What We Offer
- Opportunity to design and build a large-scale telephony infrastructure from scratch
- Work on cutting-edge AI + voice communication systems
- Experience in high-load, real-time distributed systems
- Collaborative and global engineering environment
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## ■ System Scale & Requirements
- Concurrent Calls: 10,000+
- SLA: 99.99%+ availability
- Use Cases:
- Call center systems
- AI-driven voice communication
- Mobile and desktop softphone integration
