Location: Bengaluru (on-site / hybrid)

Experience: 3 to 7 years

Employment type: Full-time

Compensation: Competitive, benchmarked to experience.

About Us

Legodesk builds legal-powered credit and recovery infrastructure for India's banks, NBFCs, fintechs and ARCs. Our platform runs the full recovery lifecycle.

The Role The Asterisk and its configuration, the SIP trunk and DID routing, the call-control layer that drives Asterisk through ARI, the browser softphone our agents work from, and the event pipeline our answer and outcome logic depends on.

What You will Own The Asterisk platform

- Deploy, configure, harden and operate Asterisk on AWS: chan_pjsip, dialplan, transports, codecs and module configuration
- Patching, upgrades, monitoring and capacity planning. You decide when we upgrade and you carry out the upgrade
- Design redundancy and failover for a production collections floor.
- Secure the platform against toll fraud and SIP attack: firewalling, fail2ban or equivalent, credential hygiene, rate limits, and an SBC in front of the trunk if you judge it necessary
- Know the concurrent-call ceiling of what you have built, and prove it with load testing rather than optimism

Call Control over ARI

- Originate calls, bridge legs, manage conferences, start and stop recording, and hang up, all programmatically through the Asterisk REST Interface
- Consume the ARI WebSocket event stream as the source of truth for call state: ringing, answered, hangup, failure and answering-machine detection
- Make that stream reliable in production: reconnection, ordering, deduplication, missed-event reconciliation and state recovery after a drop. Our answer and outcome logic keys off these events
- Keep call control behind a clean abstraction, so campaign, agent and disposition logic sits on a stable interface rather than being rewired every time the telephony layer changes

WebRTC agent softphone





- Own the in-browser softphone for agents on Chrome: registration against chan_pjsip over WSS with DTLS-SRTP, per-agent endpoint credentials, and provisioning
- Handle the hard parts: media permissions, device selection, NAT and ICE, reconnection on network flap, and call state sync between the softphone and the console
- Diagnose and fix audio quality issues from the browser side: one-way audio, latency, jitter, packet loss and dropped legs
- Own the agent experience end to end

SIP Trunk, DIDs and caller ID

- Configure and own the SIP trunk on Asterisk: peering, registration or IP authentication, ports, codecs and inbound DID routing
- Set outbound caller ID per call through ARI so our per-DID rotation for deliverability keeps working
- Manage the DID inventory and CLI validity against what the carrier will accept
- Own the carrier relationship on the technical side. You are the person who reads the SIP trace and tells the carrier what is actually wrong, with evidence

Recording and post-call pipeline

- Control recording start and stop via ARI, and own where recordings are written, in what format, and how they reach our storage
- Guarantee that every call has a retrievable recordings, campaign and agent, with no silent gaps
- Keep the downstream transcription, AI summary and playback pipeline fed and healthy
- Own storage lifecycle, retention and access control on recordings

What we are Looking for

Must Have

- You have built and run Asterisk in production yourself: chan_pjsip,



dialplan, trunk configuration, ChanSpy and ConfBridge, plus the patching, hardening and monitoring that come with owning the box
- Hands-on production experience with ARI: originating and bridging calls programmatically and consuming the WebSocket event stream. Comparable FreeSWITCH ESL experience counts
- Real WebRTC softphone work in the browser: SIP over WSS, DTLS-SRTP, and a library such as SIP.js or JsSIP wired into a live application
- Working knowledge of SIP, RTP, codecs, NAT and ICE, and the ability to debug a call from trace to root cause
- Experience configuring a carrier SIP trunk and securing a voice platform against toll fraud
- Enough JavaScript to own the softphone in the browser, not just call an API from the backend
- Experience with an outbound dialer or contact-centre platform at real volume.
- Linux competence: systemd, Nginx, logs, performance tuning, and basic AWS (EC2, RDS, S3, VPC, security groups)
- Ownership instinct. You are comfortable being accountable for a live production voice system

Solid Plus

- You have built a browser-based agent calling experience on Asterisk that real agents used every day, not a proof of concept
- Experience in collections, debt recovery, BPO or contact-centre tech; you know what a collections floor actually looks like
- You have designed Asterisk high availability and failover for a live contact-centre floor, not just read about it
- Redis, Celery or equivalent queueing
- Speech tooling: ASR, TTS and diarization, especially for Indian languages
- Familiarity with lending or BFSI compliance obligations

How to apply

Send your CV to [email protected] In a few lines, tell us about the dialing or voice system you have worked on: how many concurrent calls, what stack, what you personally built, and what broke.

📌 Voice & Asterisk / ARI / WebRTC Engineer (Bengaluru)
🏢 Legodesk
📍 Bengaluru