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-relatedissues.
Experience designing and deploying large-scale dialer systems on cloudinfrastructure.
Solid experience in Linux system administration and shell scripting.Familiarity with SBCs (e.g., Acme Packet, Sansay, Audiocodes) and SIPdebuggers (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 callscoring.
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 UsOpportunity 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.Show more
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Skills Required
opensips, Wireshark, Nat, Sip Protocol, Linux System Administration, freeswitch, Kafka, Rtp, Rabbitmq, Shell Scripting, Mysql, Postgres, stun, Rest Apis, Webrtc, Asterisk