We are looking for an experienced VoIP Developer 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:

● 2+ 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.