Asterisk Developer

Role Summary

Own the design, development, and operations of high-availability voice platforms built on Asterisk/FreePBX/FreeSWITCH, integrating AI voice bots, audio streaming, and video live streaming. You’ll craft resilient SIP/RTP call flows, automate with AGI/ARI/AMI, and run carrier-grade services with strong observability, security, and performance.

Key Responsibilities

• Architecture & Design
• Design multi-tenant PBX/EPABX solutions with HA, SBCs, and media proxying (OpenSIPS/Kamailio).
• Define dialplan logic (inbound, outbound, IVR, queues, failover, least-cost routing).
• Plan codecs, transcoding strategy, and QoS for audio/video (SIP, RTP/RTCP, SRTP).
• Voice Bot & Realtime AI
• Build voice bots using Record(), AGI (Perl/Python), and ARI for conversational flows.
• Implement audio streaming pipelines (WebSocket/AudioSocket, µ-law/PCM) to STT/TTS engines (Vosk/Whisper/Google/AWS).
• Optimize latency, barge-in, silence detection/VAD, timeouts, and prompt/response chaining.
• Platform Implementation
• Configure and harden Asterisk, FreePBX, and FreeSWITCH (profiles, endpoints, ACLs).
• Develop AGI services (Perl/Python) and automation scripts (Bash/Node.js/PHP) for provisioning, monitoring, and reporting.
• Integrate CRMs, ticketing, and analytics; expose REST/gRPC for call control.
• Carrier & Routing
• Trunking with ITSPs (TLS/SRTP), ENUM/LCR, number normalization, CNAM/ANI.
• Failover strategies: SIP 3xx/503 handling, re-invite, RTP re-anchor, media bypass.
• Observability & Reliability
• Telephony monitoring/alerting (SIP traces, RTP stats, MOS), CDR/queue metrics.
• Set up dashboards and alerts; perform SIPp load tests; drive capacity planning.
• On-call ownership for P1/P2 incidents; root-cause analysis and post-mortems.
• Security & Compliance
• SIP authentication hardening, topology hiding, SBC policies, fraud detection.
• TLS cert lifecycle, SRTP, call recording retention, GDPR/PII handling.
• Data & Storage
• Design CDR schemas and analytics in MySQL and MongoDB.
• Optimize recording storage life-cycle: cold storage, rotation, indexing.

Required Skills & Experience

• Deep expertise with Asterisk, FreePBX, SIP/RTP, NAT traversal, and media debugging.
• Practical experience building AI voice bots with AGI (Perl & Python) and Record(), plus AMI/ARI.
• Hands-on with OpenSIPS/Kamailio as SBC/registrar/load balancer.
• Comfortable with Linux (Ubuntu/CentOS), Bash, and scripting in Perl, PHP, Python, Node.js.
• Database fluency: MySQL (schema, optimizations) and MongoDB.
• Strong troubleshooting using sngrep/rtpengine/tcpdump/pcap/Wireshark and log analysis.
• Experience with FreeSWITCH (profiles, ESL, mod_sofia/mod_verto) is a strong plus.
• Familiarity with video live streaming (WebRTC/SFU basics, TURN/STUN, codecs H.264/Opus).

Nice-to-Have

• STT/TTS engines (Vosk, Whisper, Google, AWS Polly), barge-in design.
• WebRTC gateways (Asterisk PJSIP WebSocket, FreeSWITCH mod_verto), Janus/mediasoup.
• CI/CD (Git, containers), infrastructure-as-code, Ansible.
• Message buses (RabbitMQ/Kafka) for async call events and analytics.
• Grafana/Prometheus/ELK/OpenTelemetry for end-to-end tracing.

Tools & Stack You’ll Use

Asterisk/FreePBX, FreeSWITCH, OpenSIPS/Kamailio, rtpengine/rtpproxy, PJSIP, sngrep, Wireshark/tshark, SIPp, SoX/ffmpeg, Python/Perl/PHP/Node.js, MySQL/MongoDB, Nginx, systemd, Git.

Sample Deliverables (First 90 Days)

• Production-ready IVR voice bot using AGI (Perl/Python) with STT/TTS, barge-in, and error recovery.
• Carrier-grade SBC edge with OpenSIPS/Kamailio (rate limiting, topology hiding, TLS/SRTP).
• Monitoring suite: MOS/RTP jitter loss dashboards, SIP error heatmaps, alert rules.
• LCR engine with failover and cost-based routing; CDR/QA reporting in MySQL/MongoDB.
• Recording pipeline with auto-rotation, S3/offsite archive, and searchable metadata.

Example Responsibilities by Interface

• AGI: Record(), STREAM FILE, GET DATA, custom business logic; Perl/Python modules; ASR/TTS adapters.
• AMI: originate calls, queue/status automation, click-to-call, call barge/whisper.
• ARI: dynamic call control, bridges, stasis apps, media manipulation for advanced bots.
• EPABX/Custom PBX: tenant isolation, DID mapping, call transfers/parking, paging/intercom.

KPIs

• ASR accuracy & median bot response time (P50/P95).
• Call setup success rate (CSSR), Post-Dial Delay (PDD), and MOS.
• Abandon rate, containment rate in bot flows, avg. handle time impact.
• Incident MTTR, change failure rate, and platform availability.

Qualifications

• 4–10 years in VoIP/RTC engineering with solid Asterisk/FreePBX.
• Proven projects in audio streaming and AI voice bots; exposure to video live streaming.
• Strong Linux administration and scripting discipline.
• Clear communicator, comfortable with customer-facing troubleshooting.