Asterisks Developer

CC33 Global Limited; Bengaluru; On-site; INR 1,500,000 - 2,500,000; Full time

Job summary

CC33 Global Limited seeks an Asterisk Developer with deep, hands-on expertise to design, build, and scale call-control applications on the Asterisk stack. You will own end-to-end call flows, implement ARI/AMI/AGI integrations, troubleshoot SIP and media issues, and work with our infra team to keep the platform highly available.

You should have 3+ years of production Asterisk work beyond dialplan editing, solid PJSIP knowledge, and strong scripting in Python or Bash, plus relational databases and

Qualifications

• 3+ years hands-on Asterisk in production beyond dialplan editing.
• Production experience with ARI, AMI, and AGI; know when each fits.
• Prior FastAGI servers experience.
• Strong PJSIP knowledge including endpoints and registrations.
• Proficient in Python or Bash scripting for integrations.
• Familiar with relational DBs (PostgreSQL/MySQL) and Redis.

Responsibilities

• Design, develop, and maintain Asterisk-based call-control platforms.
• Build call-control logic using ARI, AMI, and AGI; scale across nodes.
• Develop dialplans, contexts, queues, IVRs, and routing.
• Troubleshoot SIP/Media issues using traces, PCAPs, and logs.
• Integrate Asterisk with REST APIs, CRMs, and AI/voice platforms.
• Own capacity planning, failover, and deployment topology.
• Write docs for call flows, configurations, and runbooks.

Skills

Asterisk development

ARI/AMI/AGI

FastAGI

PJSIP knowledge

Python scripting

Bash scripting

Linux administration

Cloud/Kubernetes

Tools

PostgreSQL

MySQL/MariaDB

Redis

Prometheus/Grafana

Kubernetes

Job description

Job Overview

We're looking for an Asterisk Developer with deep, hands-on expertise in the Asterisk stack — not just dialplan configuration, but building and scaling call-control applications on top of it. You'll own call flows end-to-end: designing routing logic, building integrations via ARI/AMI/AGI, debugging SIP and media issues down to the packet, and working with our infrastructure team to keep the platform scalable and highly available.

Key Responsibilities

• Design, develop, and maintain Asterisk-based call-control applications and platforms.
• Build call-control logic using ARI(stasis applications, channel/bridge/playback control),AMI(event handling, originate, real-time state), andAGI/FastAGI(persistent-process AGI servers, not just per-call scripts).
• Develop and maintain dialplans, contexts, extensions, queues, IVRs, conferences (ConfBridge), and inbound/outbound routing.
• Configure and troubleshoot PJSIP endpoints, SIP trunks, DID/DDI routing, caller ID/CLI handling, and number translation.
• Diagnose SIP signalling and media problems — one-way audio, no audio, call drops, early disconnects, registration and NAT failures, codec mismatches, DTMF issues — using SIP logs,pjsipCLI, PCAPs, Wireshark, and tcpdump.
• Integrate Asterisk with REST APIs, databases, CRMs, and AI/voice platforms (ASR, TTS, LLM-driven voice agents) over WebSockets and external media channels.
• Implement call recording, CDR/CEL pipelines, and call-detail processing and reporting.
• Own capacity planning and scaling of the Asterisk layer: horizontal scaling, load distribution across nodes, registration/media distribution, failover, and graceful drain during deploys.
• Define infrastructure requirements and guide the infrastructure/DevOps team on sizing, networking, firewall/NAT, media port ranges, and deployment topology.
• Handle production incidents and lead root-cause analysis.
• Write and maintain technical documentation for call flows, configurations, and troubleshooting runbooks.

Required Skills

Asterisk (deep)

• 3+ years hands-on Asterisk in production, at a level beyond dialplan editing.
• Production experience with ARI, AMI, and AGI— clear understanding of when each is the right tool.
• Prior experience building FastAGIservers.
• Strong PJSIP knowledge (transports, endpoints, AORs, contacts, registrations).

SIP & Media

• Excellent working knowledge of SIP, SDP, RTP/RTCP, DTMF (RFC2833/SIP INFO/inband), codecs, NAT traversal, and SIP response codes.
• Proven ability to read SIP traces and PCAPs to root-cause complex signalling and media issues.
• Experience with SIP trunking, carrier integrations, DID/DDI, CLI, and number translation.

Engineering & Infrastructure

• Strong scripting/development skills in Python (or PHP/Go/Node) and Bash.
• PostgreSQL or MySQL/MariaDB, and Redis.
• Solid Linux administration, networking, firewalls, DNS, and load balancers.
• Experience designing highly available, horizontally scalable VoIP infrastructure — able to drive scaling decisions independently.
• Comfortable in cloud environments; containers/Kubernetes is a plus.

Good to Have

• Experience integrating telephony with AI voice systems — streaming audio to ASR, TTS playback, barge-in handling, latency tuning for conversational voice bots.
• Exposure to Kamailio/OpenSIPS as an edge SIP proxy in front of Asterisk.
• Monitoring and observability for VoIP (Prometheus/Grafana, Homer/sipcapture).