* Develop sales strategies and plans. * Conduct regular sales reviews with team members. * Analyze sales data using analytics tools. * Collaborate on miscellaneous industry projects. * Support in business operations, performance tracking process

Responsibilities: * Manage client relationships through regular engagement * Close sales through effective negotiation techniques * Generate leads through telecom expertise * Present sales pitches with confidence

Job description * Develop sales strategies and plans. * Conduct regular sales reviews with team members. * Analyze sales data using analytics tools. * Collaborate on miscellaneous industry projects. * Support in business operations, performance tracking process

A skilled developer to design, develop, and deploy desktop applications for both Windows and macOS platforms. You will also create reliable installers/setup packages to ensure smooth installation and execution for end-users on both operating systems. Food allowance Annual bonus Health insurance

Responsibilities: * Manage recruitment process from sourcing to offer * Ensure compliance with labor laws at all times * Coordinate onboarding activities for new hires

Responsibilities: * Lead international business growth through strategic planning & execution. * Manage global partnerships & client relationships. * Oversee voice processing operations worldwide.

Responsibilities: * Manage international client relationships * Meet revenue targets through effective sales strategies * Close deals with global organizations * Collaborate on voice overs projects worldwide

Design, develop & maintain scalable FreeSWITCH-based cloud telephony. Handle SIP, RTP, NAT, TLS, 5000+ calls, conferencing, SBCs, Kamailio, APIs. Debug via Wireshark. Integrate codecs, build HA clusters. Bonus: WebRTC, SMPP, H.323, SIPX knowledge.

Are you passionate about cloud telephony, VoIP systems, and building next-gen PBX solutions? Were looking for a skilled FreeSWITCH Developer to help us develop a robust and scalable PBX panel for enterprise-level deployments. Key Responsibilities: • Design, develop, and maintain tools supporting a scalable cloud telephony infrastructure. • Customize FreeSWITCH for audio/video conferencing, capable of 1000 to 1500 concurrent calls. • Deep understanding of SIP, RTP, RTCP, NAT traversal (TURN/STUN), and TLS encryption. • Build and deploy multiple FreeSWITCH instances behind load balancers for high availability. • Analyze media stream issues using tools like Wireshark and debug SIP signaling errors. • Work closely with frontend/mobile teams to develop API-driven PBX solutions. • Integrate codecs like PCMU, PCMA, G729, Opus and support SDP offer/answer model. • Collaborate on RTP Proxy, routed audio conferencing, and NAT traversal setups. • Familiarity with SIP proxy servers (e.g. Kamailio/OpenSER) and Session Border Controllers (SBCs). • Bonus: Knowledge of H.323, SIPX, WebRTC, and SMPP. You Should Have: • Strong hands-on experience with FreeSWITCH or other open-source telephony platforms. • Deep understanding of telecom signaling protocols and VoIP architecture. • Practical experience developing or supporting PBX platforms. • Experience with SIP debugging and high-concurrency media handling.

Job Description We are looking for a skilled and passionate FreeSWITCH & Kamailio Developer to join our on-site team in Noida Extension . The ideal candidate will help us build a scalable, secure, and high-performance PBX platform tailored for enterprise-level VoIP deployments. You will be responsible for designing, customizing, and maintaining cloud telephony systems, optimizing call flows, and handling SIP signaling at scale. This role is perfect for someone with deep technical knowledge of VoIP architecture and a hands-on approach to real-time communications. Responsibilities Design and maintain scalable cloud telephony infrastructure Customize FreeSWITCH for audio/video conferencing (4000–5000 concurrent calls) Deploy FreeSWITCH behind load balancers for high availability Debug SIP signaling and analyze RTP/media streams (Wireshark, sngrep) Integrate codecs (PCMU, PCMA, G729, Opus) and support SDP offer/answer models Develop API-integrated PBX systems in coordination with frontend/backend teams Configure RTP Proxy and handle NAT traversal (TURN/STUN) Set up and manage SIP proxy servers (Kamailio/OpenSIPS) Work on SBCs to ensure secure and reliable call routing Bonus: Experience with WebRTC, SIPX, SMPP, and H.323 Required Skills Hands-on experience with FreeSWITCH, Asterisk, or similar VoIP platforms Strong knowledge of SIP, RTP, RTCP, NAT traversal, and TLS Experience in Kamailio/OpenSIPS configuration and routing logic Proficiency in Linux system administration and shell scripting Familiarity with tools like sngrep, tcpdump, and Wireshark Ability to debug and optimize VoIP call flows and media streams Excellent troubleshooting and problem-solving skills Bachelor's degree in Computer Science, IT, or related field Experience in telecom or VoIP support environment is a plus Job Types: Full-time, Permanent Pay: ₹13,650.56 - ₹100,000.00 per month Benefits: Commuter assistance Flexible schedule Ability to commute/relocate: Noida, Uttar Pradesh: Reliably commute or planning to relocate before starting work (Preferred) Work Location: In person

Job Description We are looking for a skilled and passionate FreeSWITCH & Kamailio Developer to join our on-site team in Noida Extension . The ideal candidate will help us build a scalable, secure, and high-performance PBX platform tailored for enterprise-level VoIP deployments. You will be responsible for designing, customizing, and maintaining cloud telephony systems, optimizing call flows, and handling SIP signaling at scale. This role is perfect for someone with deep technical knowledge of VoIP architecture and a hands-on approach to real-time communications. Responsibilities Design and maintain scalable cloud telephony infrastructure Customize FreeSWITCH for audio/video conferencing (4000–5000 concurrent calls) Deploy FreeSWITCH behind load balancers for high availability Debug SIP signaling and analyze RTP/media streams (Wireshark, sngrep) Integrate codecs (PCMU, PCMA, G729, Opus) and support SDP offer/answer models Develop API-integrated PBX systems in coordination with frontend/backend teams Configure RTP Proxy and handle NAT traversal (TURN/STUN) Set up and manage SIP proxy servers (Kamailio/OpenSIPS) Work on SBCs to ensure secure and reliable call routing Bonus: Experience with WebRTC, SIPX, SMPP, and H.323 Required Skills Hands-on experience with FreeSWITCH, Asterisk, or similar VoIP platforms Strong knowledge of SIP, RTP, RTCP, NAT traversal, and TLS Experience in Kamailio/OpenSIPS configuration and routing logic Proficiency in Linux system administration and shell scripting Familiarity with tools like sngrep, tcpdump, and Wireshark Ability to debug and optimize VoIP call flows and media streams Excellent troubleshooting and problem-solving skills Bachelor's degree in Computer Science, IT, or related field Experience in telecom or VoIP support environment is a plus Job Types: Full-time, Permanent Pay: ₹13,650.56 - ₹100,000.00 per month Benefits: Commuter assistance Flexible schedule Ability to commute/relocate: Noida, Uttar Pradesh: Reliably commute or planning to relocate before starting work (Preferred) Work Location: In person

Responsibilities: * Collaborate with cross-functional teams on project delivery. * Manage customer relationships through effective communication. * Analyze sales data to optimize VoIP solutions.

Responsibilities: * Manage customer relationships through telephone support. * Generate leads, close deals, meet targets. * Collaborate with engineering team on solution design. * Present product demos and proposals.