About the role
About the Role
We are looking for an experienced Dialer Engineer / SIP Engineer who has deep expertise in setting up and scaling VoIP-based contact center solutions using Asterisk or similar technologies. The ideal candidate should have hands-on experience in building inbound and outbound dialer systems from the ground up, with a strong grasp of SIP protocol, cloud infrastructure, and telephony integration best practices.
Key Responsibilities
● Dialer Setup & Architecture
○ Design and implement scalable cloud-based inbound and outbound dialer architecture using Asterisk, Kamailio/OpenSIPS, and related tools.
○ Lead the end-to-end setup of dialer systems including IVR, predictive/auto/manual dialing modes, and queue management.
● SIP & Telephony
○ Deep understanding of SIP protocol, RTP, and VoIP call flows.
○ Configure and troubleshoot SIP trunks, SBCs, media gateways, and NAT traversal issues.
○ Optimize call routing, codec negotiation, DTMF handling, and failover.
● System Engineering
○ Own the cloud deployment of dialer systems (AWS, GCP, or Azure) using containerization (Docker/Kubernetes preferred).
○ Ensure high availability, performance monitoring, logging, and disaster recovery for dialer infrastructure.
○ Collaborate with network and DevOps teams to fine-tune VoIP performance and reliability.
● Compliance & Quality
○ Ensure the dialer adheres to compliance standards like DND, TRAI/DoT (India), TCPA (US), etc.
○ Implement call recording, real-time monitoring, and post-call analytics systems.
● Development & Integration
○ Work closely with backend and frontend developers to expose dialer APIs for CRM/agent dashboards.
○ Integrate with third-party telephony platforms (Tata, Airtel, Twilio, Exotel, etc.) and CRM systems.
● Troubleshooting & RCA
○ Perform in-depth debugging of call failures using PCAP traces, SIP logs, and Asterisk CLI.
○ Provide root cause analysis and implement permanent fixes for recurring issues.
Requirements
● 6+ years of hands-on experience with Asterisk, Kamailio/OpenSIPS, FreeSWITCH, or similar VoIP systems.
● Deep knowledge of SIP protocol, RTP, WebRTC, STUN/TURN, and NAT-related issues.
● Experience designing and deploying large-scale dialer systems on cloud infrastructure.
● Solid experience in Linux system administration and shell scripting.
● Familiarity with SBCs (e.g., Acme Packet, Sansay, Audiocodes) and SIP debuggers (sngrep, Wireshark).
● Exposure to call center metrics, DNC lists, retry logic, concurrency management.
● Experience with REST APIs, MySQL/Postgres, and message queues (RabbitMQ/Kafka) is a plus.
● Bonus: Experience with voice biometrics, conversational IVR, AI-based call scoring.
Preferred Qualifications
● Bachelor’s or Master’s degree in Computer Science, Electronics, or a related field.
● VoIP certifications (dCAP, CCVP, etc.) are a plus.
● Prior experience working in fintech, edtech, or BPO-focused tech environments.
Why Join Us?
● Opportunity to own and shape the telephony backbone of a high-growth product.
● Work with a team that values autonomy, innovation, and deep technical ownership.
● Be part of solving real business challenges through scalable communication infrastructure.