50 Freeswitch Jobs

Title : Sr Software Engineer (FreeSwitch) Experience : 2+ years Location : Remote About The Opportunity Netrix Global is looking for a passionate Senior Software Engineer that is resourceful Freeswitch and Lua developer to join our team. If you're someone who enjoys tackling complex telephony challenges and independently exploring innovative solutions, we encourage you to apply! How You Will Make An Impact As a key member of our development team, you will play a vital role in shaping and enhancing our telephony applications. You will: Drive Innovation: Conduct independent research and development to address intricate technical problems within our telephony infrastructure, proposing and implementing cutting-edge solutions. Build and Optimize: Design, develop, and maintain robust and scalable telephony applications leveraging the power of Freeswitch and Lua. Troubleshoot and Resolve: Analyze and resolve complex technical issues within our telephony systems, ensuring seamless and reliable communication. Integrate and Extend: Integrate Freeswitch with other systems and develop custom modules and functionalities to meet specific business needs Contribute to Architecture: Participate in the design and architecture of our telephony solutions, ensuring they are efficient, secure, and future-proof. Collaborate Effectively: Work closely with other developers, system administrators, and stakeholders to deliver high-quality solutions. What You Will Bring To The Table You are a motivated and skilled developer with a strong foundation in telephony and scripting. Ideally, you will bring: Solid Experience: A minimum of 2 years of hands-on development experience with Freeswitch and Lua combined with proficiency in reading/understanding code written in C. Freeswitch Expertise: A deep understanding of Freeswitch architecture, modules, configuration, and core functionalities. Lua Proficiency: Strong scripting skills in Lua, with the ability to develop efficient and well-documented code. Telephony Fundamentals: A good grasp of VoIP protocols (SIP, RTP), codecs, and general telephony concepts. Problem-Solving Prowess: Proven ability to independently research, analyze, and solve complex technical challenges. Linux Familiarity: Comfortable working in a Linux environment. Independent Learner: A proactive attitude towards learning new technologies and staying up-to-date with industry trends. Excellent Communication: Strong written and verbal communication skills. Interested folk please share your updated CV on smahore@netrixllc.com What You Can Expect From Us We offer a competitive compensation package, comprehensive group benefits to meet the needs of you and your family, flexibility, and time off when you need it, and a casual work environment. All qualified applicants will receive consideration for employment without regard to race, color, religion, sex, sexual orientation, gender identity, national origin, disability, or veteran status. To learn more about Netrix Global please go to www.netrixglobal.com

As a Junior System Administrator (VoIP) / Telephony Engineer at Telebu, you will be responsible for developing, implementing, and supporting IP Telephony related technologies. This includes working on IP Telephony, IVR platforms, Conferencing solutions, Voice engineering integration, Voice over IP (VoIP), Session Border Controllers (SBC), Session Initiation Protocol (SIP), WebRTC, and Public Switched Telephone Network (PSTN) gateways. **Key Responsibilities:** - Develop and implement telephony networks with various components such as SIP proxies, registrar, media-servers, billing systems, etc. - Deploy SIP VOIP/PRI trunking solutions that are highly scalable, robust, high-availability (HA), and fault-tolerant telecom products/systems. - Handle administration of SIP and Media Servers, conduct Network/Protocol level debugging and testing, work on Contact center solutions, and troubleshoot and resolve complex problems. - Provide IP Telephony and VoIP Subject Matter Expertise for the company, manage Company's managed service providers, and handle 3rd party telecom carriers and providers. **Qualifications Required:** - 3 to 5 years of hands-on industry experience in telecommunications. - Strong conceptual knowledge and experience with telephony protocols like SIP, SDP, RTP, SRTP, WebRTC, and audio/video codecs. - In-depth working experience with Kamailio, Freeswitch, any of the SIP stack (Sofia, reSIProcate, PJSIP, etc.), ICE Framework (STUN/TURN), and Linux. - Proficient in writing production quality code using scripting languages like Python, Go, Erlang, etc. - Experience in using VoIP testing tools like Wireshark, VoIPMonitor, SIPp, SIPCapture, Homer, etc. **Nice to have:** - Working knowledge in any of NoSQL databases like MongoDB, Redis, Cassandra, etc. - Passion for understanding VoIP Protocol standards & related RFCs. - Experience with virtualization/container related technologies such as Xen, VMware vSphere / ESXi, Docker, Kubernetes.,

3-5 years of experience in B2B technical sales or VoIP solution engineering. Strong understanding of VoIP technologies: SIP, RTP, PBX, DID provisioning, codecs, Experience with platforms like FusionPBX, Asterisk, FreePBX, 3CX, or Freeswitch .

As a Freeswitch Engineer with OpenSIPs, you will be responsible for leveraging your 3-5 years of hands-on experience in a production environment to deploy, configure, and maintain Freeswitch and OpenSIPs systems. Your technical expertise will include a strong understanding of Freeswitch configuration, dial plans, scripting languages (e.g., Lua, XML), and custom module development. You will also need extensive experience with OpenSIPs, focusing on SIP routing, load balancing, and SIP proxy configuration. Your role will require proficiency in troubleshooting VoIP issues, such as call routing, signaling, and media path problems. You should be familiar with SIP protocols, VoIP standards (RFCs), and have experience integrating SIP-based platforms. Additionally, you will be expected to have a strong grasp of networking concepts and protocols (TCP/IP, DNS, NAT) and knowledge of network security practices to secure VoIP services and prevent toll fraud. Scripting and automation skills are essential for this position, as you will need experience with languages like Python, Perl, or Bash to automate tasks such as system backups, monitoring, and configuration changes. Your problem-solving abilities will be put to the test as you diagnose and resolve telecommunication issues efficiently and think critically to identify problems proactively. Preferred qualifications include certifications in VoIP, Freeswitch, or OpenSIPs (e.g., Freeswitch Certified Administrator, OpenSIPs Certification). If you are ready to take on this challenging role and contribute to our team, please share your resume at giri@vitelglobal.com.,

As a VoIP Developer, you are expected to have a minimum of 3 years of experience in the field. Your responsibilities will include working with open source VoIP solutions like Asterisk, FreeSWITCH, OpenSIPS, and Kamailio. You should possess a strong understanding of PSTN lines and be proficient in VoIP and SIP technologies. Your role will involve the installation and configuration of various products, so prior experience in this area is essential. Proficiency in tools such as wireshark, tcpdump, sipp, fail2ban, and IPtables is required. You must have excellent analytical and troubleshooting skills to identify and resolve issues effectively. A good grasp of databases, including Mysql and Mongo, is necessary, along with experience in Database Replication and Heartbeat configuration. Familiarity with the TCP/IP protocol stack, LAN/WAN networks, and value-added VoIP features like Voicemail, ACD, IVRS, and CTI is also expected. Knowledge of SIP/SDP protocols, Softphones, Linux/server administration, WebRTC, and Enterprise Communication Servers (PBXs) is crucial for this role. You should be adept at problem troubleshooting and debugging techniques for both hardware and software components. In summary, your expertise in VoIP technologies, database management, network protocols, and server administration will be essential for success in this role.,

You are Hiring for the position of VoIP Developer at Vitel Global Communications Pvt Ltd in Hyderabad. As a VoIP Developer, you will be responsible for developing and maintaining VoIP applications, ensuring the smooth functioning of VoIP services, and optimizing communication channels for secure and efficient operations. To excel in this role, you should have proven experience in VoIP development with expertise in VoIP protocols such as SIP, RTP, and WebRTC. Your strong programming skills in languages like Python, C/C++, Java, or Lua will be essential for creating VoIP applications. Additionally, you should be adept at working with platforms like Asterisk, FreeSWITCH, OpenSIPS, or similar systems. Your responsibilities will include configuring and troubleshooting VoIP services on Linux servers, utilizing databases like MySQL or PostgreSQL for call data storage, and implementing network configurations for optimal VoIP performance. Knowledge of cloud platforms such as AWS, Azure, or Google Cloud, as well as experience with WebRTC for browser-based communication, will be beneficial for this role. Understanding security protocols in VoIP, call routing, load balancing, and high-availability configurations are key aspects of this position. Familiarity with SBCs and their role in securing VoIP networks will be an advantage. Your excellent troubleshooting skills, ability to diagnose complex VoIP issues, and strong communication skills are essential for collaborating with the team and ensuring effective VoIP operations. To qualify for this role, you should hold a Bachelor's degree in Computer Science, Information Technology, or a related field, along with a minimum of 3 years of experience in VoIP development, integration, and troubleshooting. Previous experience in a production environment working with telephony platforms and network infrastructure will be valuable. If you meet these requirements and are passionate about VoIP development, integration, and troubleshooting, we encourage you to share your resume with us at giri@vitelglobal.com to explore this exciting opportunity further.,

Hiring FreeSWITCH Developer | South Delhi Location: Eco Space Exp: 4+ yrs (VoIP/FreeSWITCH) Full-time | Day Shift Skills: FreeSWITCH, SIP, RTP, WebRTC, Linux, SQL, Bash/Python, Telecom Protocols. 6290 291 883

About the Role: We are looking for a skilled and dedicated FreeSWITCH Engineer with hands-on experience in VoIP systems. You will play a key role in developing, configuring, and maintaining scalable and reliable FreeSWITCH-based voice infrastructures. This is a remote opportunity, giving you the flexibility to work from anywhere while collaborating with a global team. Key Responsibilities: Design, deploy, and maintain FreeSWITCH servers and related VoIP infrastructure. Troubleshoot and resolve FreeSWITCH and VoIP-related issues. Develop custom dial plans, modules, and call routing logic. Work with SIP, RTP, and related VoIP protocols. Monitor system performance and ensure high availability. Collaborate with development, network, and support teams to optimize voice systems. Document configurations, workflows, and system changes. Requirements: Minimum 3 years of hands-on experience with FreeSWITCH in a production environment. Strong understanding of VoIP technologies and SIP protocol. Experience with Linux system administration. Familiarity with scripting languages (e.g., Bash, Python, Lua). Ability to work independently in a remote setup. Strong problem-solving and analytical skills. Preferred Skills: Experience with other VoIP platforms (e.g., Asterisk, Kamailio, OpenSIPS). Knowledge of WebRTC, RTP engines, or media servers. Exposure to monitoring tools (Grafana, Prometheus, etc.). Familiarity with APIs and backend integration. Why Join Us Collaborative and supportive team environment Opportunity to work on innovative VoIP solutions at scale

Position Summary : The candidate suitable for this role of Senior Software Engineer will be responsible for leading the development and implementation of complex software solutions. This role involves a high level of technical expertise and the ability to guide and mentor junior team members. The Senior Software Engineer will collaborate with cross-functional teams to define, design, and ship new features while maintaining high standards of software quality. Key Responsibilities : - Design and develop high-volume, low-latency applications for mission-critical systems, delivering high availability and performance. - Contribute to all phases of the development lifecycle, from concept and design to testing. - Write well-designed, testable, and efficient code. - Ensure designs comply with specifications. - Prepare and produce releases of software components. - Support continuous improvement by investigating alternatives and technologies and presenting these for architectural review. Skills : - 4-6 years of experience in software development. - Solid background in GoLang. - Strong data structures and algorithms concepts. - Designing and problem-solving skills, with a strong bias for architecting for performance and scalability. - Sound knowledge of cloud services and Kubernetes. Good to have Skills : - Good Understanding SIP/RTP protocols - Hands-on experience with any of FreeSWITCH/Asterisk/OpenSIPS/Kamailio open source VoIP softwares. Qualifications : - B. Tech/M. Tech/MCA in Computer Science Benefits : - Flexible Working Hours. - Hybrid Working Style. - Personal Accidental Insurance. - Health Insurance to Self, Spouse and two kids. - 5 days working week.

As a skilled FreeSWITCH Developer with experience in OpenSIPS and/or Kamailio, you will join our core VoIP engineering team. In this hands-on role, you will be responsible for architecting, building, and maintaining robust SIP-based communication systems. If you are excited about solving complex signaling challenges and scaling VoIP infrastructure, this is the right opportunity for you. Your responsibilities will include building and maintaining scalable VoIP services using FreeSWITCH, configuring and managing SIP proxies (OpenSIPS, Kamailio) for routing, load balancing, and NAT traversal. You will also develop and fine-tune dialplans, call flows, and scripts in Lua, Python, or JavaScript. Troubleshooting signaling/media issues using tools like Wireshark, sngrep, and FreeSWITCH logs will be a part of your routine. Additionally, you will integrate VoIP systems with external APIs, databases, CRMs, and billing platforms, ensuring security, uptime, and scalability across the voice infrastructure. Documenting systems and contributing to architectural decisions will also be crucial aspects of your role. To be successful in this role, you should have at least 3 years of experience working with FreeSWITCH in production environments, a strong grasp of the SIP protocol and related RFCs, and practical experience with OpenSIPS and/or Kamailio configuration. Deep troubleshooting skills using tools like Wireshark, sngrep, and HOMER/HEPIC, as well as scripting experience (Lua, Bash, Python, JS) are required. A solid understanding of VoIP architecture, NAT traversal, and RTP handling, along with Linux systems admin experience (Ubuntu, Debian, CentOS), are also essential. Bonus points if you have experience integrating WebRTC, knowledge of rtpproxy, rtpengine, monitoring with Grafana, Prometheus, or HEPIC, working in containerized environments using Docker/Kubernetes, or experience in billing or CRM system integration. In return, you will receive a competitive salary, flexible hours with remote work options, the opportunity to have a direct impact on production systems and architecture, a chance to work with advanced VoIP technologies, and the support of smart teammates who are passionate about what they do. If you are passionate about VoIP and building high-performance SIP systems, we encourage you to apply by sending your resume and a brief cover letter to info@riopath.com.,

MetroGuild is seeking a Network Engineer with VoIP exposure who is excited to work on modern, open-source telephony technologies. You will play a central role in helping us manage and evolve our SIP-based infrastructure using platforms like FreeSWITCH, Kamailio, and related systems with mentoring and guidance provided as needed. This is a hybrid network + telephony + cloud infrastructure role. Prior deep experience in FreeSWITCH or Kamailio is not expected; what we need is strong networking fundamentals, solid Linux experience, familiarity with cloud platforms, and a proactive, learning mindset. You'll be joining a distributed team that builds and supports voice solutions for our customers across the US, EU, and India. Our stack includes Python, ElasticSearch, ReactJS, React Native, Cassandra, AWS, and modern open-source VoIP components. Your responsibilities will include configuring and maintaining SIP-based VoIP infrastructure using FreeSWITCH, Kamailio, and RTP relays, monitoring and troubleshooting SIP signaling, RTP flows, NAT traversal, and WebRTC performance, working closely with React Native and backend developers integrating SIP.js and WebRTC flows, integrating and maintaining SIP trunks (Twilio, VoIP.ms, Airtel, BSNL, etc.), managing internal office LAN/WAN, DHCP/DNS, and network security across teams, administering AWS resources like Route 53 (DNS), S3 (object storage), EC2 instances, ensuring high uptime, secure networking, and proper infrastructure documentation, and extending call routing logic and debugging live systems under production load. Minimum Qualifications: - 1-3 years of experience with FreeSWITCH (Mandatory) - 3-5 years of experience in Networking, DevOps, or Infrastructure Engineering - Strong understanding of networking protocols: TCP/IP, NAT, DNS, SIP, RTP - Hands-on experience managing AWS services, especially Route 53, S3, and EC2 - Prior experience managing or configuring internal corporate networks and routers - Linux system administration and troubleshooting proficiency - Familiar with one scripting/backend language (Python, Java, or Node.js) - Basic exposure to VoIP/SIP concepts and desire to grow into the domain - Good communication and collaboration skills - Bachelor's degree in Computer Science, IT, or a related field. Nice to Have (Optional but Valued): - Exposure to Kamailio / OpenSIPS - Familiarity with WebRTC, SIP.js, and SIP over WebSocket - Experience working with trunk providers (Twilio, Airtel, BSNL, TATA) - Experience with CI/CD, monitoring tools, and Docker-based deployment Work Location: This is an on-site role (Pune) as the telephony infrastructure and internal networking are managed locally. Occasional client site visits may be required based on project needs.,

We are seeking VoIP engineer, who can manage VoIP Switches and VoIP PBX Systems. Candidate should have a deep knowledge in the Contact Center System and Business Communication System. and capable of handling business customers as well.

We are seeking an experienced Dialer Engineer with 3-5 years of expertise in VoIP & Dialer System, focusing on constructing, customizing, and enhancing dialer platforms for large-scale contact center solutions. The perfect candidate will possess profound knowledge in Vicidial, Asterisk, FreeSWITCH, OpenSIPS, and Kamailio, coupled with practical experience in integrating CPaaS and Telephony APIs. Your responsibilities will include installing, configuring, and tailoring Vicidial / GoAutoDial / other dialer systems, managing and optimizing Asterisk / FreeSWITCH / OpenSIPS / Kamailio for call routing and SIP services, implementing and refining Predictive Dialer, Auto Dialer, IVR, and Call Routing, integrating VoIP APIs (Plivo, Twilio, Telnyx, Exotel, Ozonetel, etc.) into dialer solutions, managing SIP Trunks, DID Management, Call Queues, Failover & High Availability, optimizing dialer performance for concurrent calls, scalability, and QoS, troubleshooting and resolving VoIP/Dialer-related issues in production environments, and ensuring security, fraud prevention, and SIP firewalling in VoIP infrastructure. The ideal candidate will possess strong proficiency in Vicidial / Asterisk / FreeSWITCH, familiarity with OpenSIPS / Kamailio (SIP Proxy & Load Balancer), knowledge in SIP, RTP, WebRTC, IVR, Call Recording, Conferencing, integration experience with Plivo, Twilio, Telnyx, Exotel, Ozonetel, experience in VoIP QoS, Load Balancing, and Failover mechanisms. Additionally, experience with Call Center Analytics & Reporting, familiarity with CRM integrations for dialer systems, and exposure to Call Recording, Compliance (DNC, GDPR), and Security best practices would be advantageous. At ORI, you can expect a workplace infused with passion and happiness, an open culture with tremendous growth opportunities, fostering leadership, opportunities to work on cutting-edge technology, and the freedom to innovate with multiple technologies. If you possess a strong expertise in dialer technologies and a drive for crafting innovative contact center solutions, this role is perfect for you.,

Key Responsibilities: Design, configure, and maintain OpenSIPS as SIP proxy/registrar/load balancer. Develop and customize FreeSWITCH for call routing, IVR, conferencing, and advanced VoIP features. Implement SIP, RTP, WebRTC integrations with high availability and redundancy. Optimize system performance for low latency and high concurrent call handling . Troubleshoot signaling and media issues using SIP traces, Wireshark, sngrep, Homer, etc. Collaborate with network/security teams for NAT traversal, TLS/SRTP, and firewall policies . Work with developers for API integrations (REST, XML, ESL) and billing systems . Ensure system scalability, failover testing, and load balancing . Required Skills & Qualifications: Strong hands-on experience with OpenSIPS (or Kamailio) scripting, module configuration, clustering. Proven expertise in FreeSWITCH – dialplans, Lua/JavaScript modules, ESL, and media applications. Good understanding of SIP, RTP, STUN/TURN/ICE, ENUM, DNS-SRV . Proficiency in Linux (Debian/Ubuntu/CentOS) server administration. Experience with PostgreSQL/MySQL, Redis, RabbitMQ/Kafka is a plus. Familiarity with Ansible/Docker/Kubernetes for deployment is an added advantage. Excellent debugging and problem-solving skills.

Job Profile:- We are looking for an experienced Dialer Engineer with 3-5 Years of experience in VoIP & Dialer System , who specializes in building, customizing, and optimizing dialer platforms for large-scale contact center solutions. The ideal candidate should have strong expertise in Vicidial, Asterisk, FreeSWITCH, OpenSIPS, and Kamailio , along with hands-on experience in integrating CPaaS and Telephony APIs . Typical work week looks like:- Install, configure, and customize Vicidial / GoAutoDial / other dialer systems Manage and optimize Asterisk / FreeSWITCH / OpenSIPS / Kamailio for call routing and SIP services Implement and fine-tune Predictive Dialer, Auto Dialer, IVR, and Call Routing Integrate VoIP APIs (Plivo, Twilio, Telnyx, Exotel, Ozonetel, etc.) into dialer solutions Handle SIP Trunks, DID Management, Call Queues, Failover & High Availability Optimize dialer performance for concurrent calls, scalability, and QoS Troubleshoot and resolve VoIP/Dialer-related issues in production environments Ensure security, fraud prevention, and SIP firewalling in VoIP infrastructure Our ideal candidate should have:- Strong expertise in Vicidial / Asterisk / FreeSWITCH Experience with OpenSIPS / Kamailio (SIP Proxy & Load Balancer) Knowledge of SIP, RTP, WebRTC, IVR, Call Recording, Conferencing Integration experience with Plivo, Twilio, Telnyx, Exotel, Ozonetel Experience in VoIP QoS, Load Balancing, and Failover mechanisms Good to Have:- Experience with Call Center Analytics & Reporting Familiarity with CRM integrations for dialer systems Exposure to Call Recording, Compliance (DNC, GDPR), and Security best practices What you can expect from ORI:- Passion & happiness in the workplace with great people & open culture with amazing growth opportunities. An ecosystem where leadership is fostered builds an environment where everyone is free to take necessary actions to learn from real experiences. Chance to work on cutting-edge of technology. Freedom to pursue your ideas and innovate with multiple technologies If you have strong expertise in dialer technologies and a passion for building innovative contact center solutions, then you will love this role. Show more Show less

Role & responsibilities Must-Haves Strong VoIP Protocol Knowledge - SIP, RTP, WebRTC Hands-on VoIP Platforms - Asterisk or Freeswitch (Kamailio/OpenSIPS is a bonus) Programming in Golang (preferred) OR Ruby / C++ with willingness to learn Golang Networking Fundamentals for VoIP - NAT traversal, ICE/STUN, L2/L3 routing Cloud Experience - AWS (preferred) or GCP/Azure Linux Proficiency (deployment, debugging, and scripting) Nice-to-Haves Kamailio/OpenSIPS expertise SIP testing tools Elasticsearch + Grafana/Kibana DevOps tools (Terraform, Kubernetes, Ansible, Jenkins) Android/iOS VoIP client development

Must have experience in Freeswitch SIP and RTP Protocols Must have experience in coding language C, Python or Lua Kamailio and Opensip is a plus point. Call Hr 8016099788

As a skilled FreeSWITCH Developer with 2 to 5 years of experience, you will be responsible for designing, developing, and customizing FreeSWITCH-based voice and call center applications. Your expertise in Lua scripting and deep understanding of VoIP protocols will be essential for building scalable and high-performance voice applications and call flows. Additionally, your familiarity with open-source telephony platforms such as FusionPBX, VICIdial, or Asterisk will be a strong advantage. Your key responsibilities will include writing and maintaining Lua scripts for call routing, IVRs, conferencing, and voicemail, as well as implementing and supporting complex call flows, voice menus, and call queuing systems. You will be required to integrate with external systems using REST APIs, WebSockets, or sockets, optimize system performance, troubleshoot VoIP issues, and provide Tier 2/3 technical support. Furthermore, you will configure and manage FreeSWITCH modules, dial plans, and profiles, and collaborate closely with DevOps and QA teams to deploy and test solutions. To excel in this role, you must possess a minimum of 2 years of hands-on development experience with FreeSWITCH and demonstrate strong proficiency in Lua scripting for telephony use cases. A good understanding of SIP, RTP, WebRTC, and other VoIP standards is crucial, along with experience customizing call center platforms like FusionPBX or VICIdial. Familiarity with Linux administration, REST APIs, MySQL/PostgreSQL, and debugging call flows using SIP traces and logs will be beneficial. Preferred qualifications include experience in multi-tenant or high-availability VoIP architecture, familiarity with WebRTC and browser-based calling features, knowledge of containerization, CI/CD pipelines, and Git, as well as prior involvement in building custom call center dashboards or real-time monitoring tools. In addition to technical skills, you are expected to possess strong problem-solving and debugging abilities, work effectively both independently and within a team, communicate clearly, and document your work meticulously. Comfort with Agile/Scrum environments is desirable for this full-time position. This position offers benefits such as provided food and paid sick time, with a Monday to Friday schedule.,

Join Vonage and help us innovate cloud communications for businesses worldwide! Vonage has built its successful global Support teams on individuals with technical savviness, superior customer relationship skills, and a passion for learning. We challenge our Support Engineers to provide a customer experience that leaves our users impressed, loyal and true advocates of our company. As a Senior Support Engineer, you will provide first-class technical support to our rapidly growing strategic customer base, who rely on our real-time communication APIs and SDKs. You will be responsible for driving and managing customer-related projects, initiatives and tasks for our strategic accounts, collaborating heavily with Sales, Engineering, and the rest of the Vonage organization. What will you do Investigate, troubleshoot, diagnose and resolve technical issues related to customer API and SDK implementations Communicate effectively (both verbal and written) with our customers and internal stakeholders Be a problem solver, have a natural curiosity, and demonstrate the ability to learn rapidly Contribute to internal and external knowledge bases Collaborate with your team to identify bugs and escalate to Product/Engineering teams Communicate well with different audiences (developers, technical and non-technical users) What You Must Have 3+ years as a Support Engineer in the telecommunications or SaaS sectors Messaging technologies: SMPP, GSM, SMS Strong knowledge of RESTful APIs and the ability to understand and troubleshoot issues with cloud solutions English and Japanese language proficiency Any Of The Following Is a Plus Experience with Voice technologies: SIP, VoiceXML, CCXML, WebRTC Supporting APIs or SDKs Excellent understanding of networking: TCP/IP, UDP, most common protocols Voice software: Asterisk, Freeswitch, Kamailio, Voxeo Prophecy Theres no perfect candidate. You don&apost need all the preferred qualifications to make a valuable impact on our team. Our employees and customers come from diverse backgrounds, so if you&aposre passionate about what you could achieve at Vonage, we&aposd love to hear from you. Who We Are Vonage is a global cloud communications leader. And your talent will further help brands - such as Airbnb, Viber, WhatsApp, and Snapchat - accelerate their digital transformation through our fully programmable-based unified communications, contact center solutions, and communications APIs. Ready to innovate Then join us today. Note: The purpose of this profile is to provide a general summary of essential responsibilities for the position and is not meant as an exhaustive list. Assignments may differ for individuals within the same role based on business conditions, departmental need or geographic location. Show more Show less

Roll-Freeswitch/Asterisk Developer Exp- 2+ years Type-FTE Budget- As per company standard Location- Pune Mode- WFO Notice Period- Immediate to 90 days Max Mandatory skill set- FreeSWITCH, Kamilio, Programming & Scripting: Develop custom VoIP features and modules using any one languages like Golang, Lua, Python, C, and C++,AWS.

As a Senior Java Developer/Technical Lead for Voice & Email Platform at our growing team, you will be responsible for developing, optimizing, and maintaining Java-based applications. Your role involves working with RESTful APIs, Webhooks, and WebSocket protocols to facilitate real-time communication. Building secure and scalable backend systems to efficiently handle high-volume traffic is a key aspect of your responsibilities. Collaboration with cross-functional teams is essential to enhance performance and reliability. Additionally, you will troubleshoot issues related to support tickets and third-party integrations and optimize system performance using multithreading, caching, and database indexing techniques. To excel in this role, you should have 3 to 7 years of experience as a Senior Developer, with expertise in Java, Spring Boot, Hibernate, and Microservices. Hands-on experience with Asterisk or Freeswitch, along with a solid understanding of RESTful APIs, Webhooks, and JSON, is required. Proficiency in MySQL, PostgreSQL, or NoSQL databases (MongoDB, Redis), exposure to AWS and scalable application architecture, and familiarity with message brokers like RabbitMQ and Kafka are necessary. Knowledge of security protocols such as OAuth, JWT, and experience with Docker, Kubernetes, CI/CD pipelines, and Git (GitHub, GitLab, Bitbucket) is expected. Unit testing using JUnit, Mockito, experience in CPaaS, and understanding of chatbots, NLP, and AI-driven messaging are essential skills. For the Tech Lead role, in addition to the above requirements, you need to have a strong grasp of design patterns and best practices, proven leadership and team mentorship abilities, and experience in managing sprints and tasks via tools like JIRA. Exposure to building Voice Bots using AI/ML is considered a significant advantage. The ideal candidate should hold a Bachelors or Masters degree in Computer Science, Information Technology, or a related field. If you are passionate about building high-performance communication systems and enjoy solving real-world challenges, we encourage you to apply for this exciting opportunity with us. Contact us directly via messaging or apply now to jobs@pinnacle.in or dhiraj.rokade@pinnacle.in. Join us in revolutionizing communication platforms and making a difference in the tech industry! #JavaJobs #Hiring #TechLead #SeniorDeveloper #SpringBoot #VoicePlatform #MessagingPlatform #CPaaS #AI #Freeswitch #Asterisk #Microservices #LiveChat,

Job Title: Engineering Manager Location: Gurugram, Haryana Experience Required: 12-15 years About the Role: We are seeking an experienced Engineering Manager to lead a talented team of FreeSWITCH engineers, Python developers, and UI/UX designers in building and scaling our click-to-call marketing SaaS analytics platform for advertisers. You will drive the development of a robust, scalable, and secure platform that empowers advertisers with real-time call tracking, analytics, and campaign optimization tools. Key Responsibilities Lead and mentor a cross-functional team of FreeSWITCH engineers, Python developers, and UI/UX designers to deliver high-quality, scalable features for our SaaS platform. Oversee the end-to-end development lifecycle, from planning and architecture to deployment and maintenance, using Agile/Scrum methodologies. Ensure seamless integration of telephony systems (e.g., FreeSWITCH, VoIP) with analytics pipelines and advertiser-facing dashboards. Collaborate with product managers and stakeholders to translate advertiser needs (e.g., call attribution, ROI tracking) into technical solutions. Implement and maintain CI/CD pipelines, containerized environments, and API-driven integrations to support rapid iteration and scalability. - Uphold SaaS security best practices (e.g., SOC2, GDPR) to protect sensitive call and advertiser data. Foster a collaborative, inclusive team culture, motivating engineers to achieve technical and professional excellence. Qualifications Bachelors degree in Computer Science, Engineering, or a related field (Masters preferred). 1215 years of software engineering experience, with expertise in both front-end (e.g., React, Angular) and back-end development (e.g., Python, Node.js). Proven experience building and scaling B2B SaaS platforms, ideally in adtech, martech, or analytics. Hands-on experience with VoIP and telephony systems (e.g., FreeSWITCH, Asterisk) and real-time communication protocols (e.g., SIP, WebRTC). Strong understanding of data analytics pipelines for processing and visualizing marketing metrics (e.g., call attribution, conversion tracking). Demonstrated leadership in managing cross-functional engineering teams, with a track record of delivering complex projects on time. Proficiency in containerization (e.g., Docker, Kubernetes), microservices, and API-first architectures. Experience with CI/CD tools (e.g., Jenkins, GitLab CI) and cloud platforms (e.g., AWS, GCP, Azure). Knowledge of SaaS security, including access control, data privacy, and compliance (SOC2, GDPR). Excellent problem-solving and decision-making skills in fast-paced environments. - Strong communication skills to convey technical concepts to non-technical stakeholders, such as advertisers or marketing teams. (Optional) Experience working in dynamic markets like Dubai or the Middle East, with an understanding of regional business practices. Preferred Skills Familiarity with adtech/martech ecosystems (e.g., CRM integrations, Google Ads, or call tracking platforms). Experience mentoring remote or distributed teams to maintain productivity and collaboration. Passion for building user-centric analytics platforms with intuitive UI/UX. Why Join Us? Be part of a fast-growing SaaS company revolutionizing click-to-call marketing analytics. Lead a talented team, work on cutting-edge telephony and analytics solutions, and make a direct impact on advertisers success.

Sr. Software Engr–VoIP (3+ yrs), SIP, RTP, Asterisk/Freeswitch, Kamailio, WebRTC, AWS, Golang, REST APIs, DevOps, MySQL, Linux. Prod-grade VoIP dev exp a must. C2H via TE Infotech (Exotel), Convertible to Permanent, Loc:BLR @ ssankala@toppersedge.com

We are seeking a skilled and dedicated FreeSWITCH Engineer with hands-on experience in VoIP systems to join our team. As a FreeSWITCH Engineer, you will be instrumental in the development, configuration, and maintenance of scalable and reliable FreeSWITCH-based voice infrastructures. Your responsibilities will include designing, deploying, and maintaining FreeSWITCH servers and related VoIP infrastructure. You will troubleshoot and resolve FreeSWITCH and VoIP-related issues, develop custom dial plans, modules, and call routing logic, and work with SIP, RTP, and related VoIP protocols. Monitoring system performance, ensuring high availability, collaborating with development, network, and support teams, and documenting configurations and system changes will also be part of your role. To be successful in this position, you should have a minimum of 2 years of hands-on experience with FreeSWITCH in a production environment, a strong understanding of VoIP technologies and SIP protocol, experience with Linux system administration, and familiarity with scripting languages such as Bash, Python, and Lua. The ability to work independently in a remote setup, strong problem-solving and analytical skills are also essential. Preferred skills include experience with other VoIP platforms like Asterisk, Kamailio, OpenSIPS, knowledge of WebRTC, RTP engines, or media servers, exposure to monitoring tools like Grafana and Prometheus, familiarity with APIs and backend integration. Join us for a collaborative and supportive team environment where you will have the opportunity to work on innovative VoIP solutions at scale.,

You will be part of Telebu's Communications engineering team as a Senior System Administrator (VoIP) / Telephony Engineer (Telecommunications). Your primary responsibility will be to develop, implement, and support IP Telephony related technologies such as IP Telephony, IVR platforms, Conferencing solutions, Voice engineering integration, Voice over IP (VoIP), Session Border Controllers (SBC), Session Initiation Protocol (SIP), WebRTC, and Public Switched Telephone Network (PSTN) gateways. Your key responsibilities will include developing and implementing telephony networks with various components like SIP proxies, registrar, media-servers, billing systems, and deploying SIP VOIP/PRI trunking solutions that are highly scalable, robust, high-availability (HA), and fault-tolerant. You will also be responsible for the administration of SIP and Media Servers, network/protocol level debugging and testing, contact center solutions, and troubleshooting and resolving complex problems. Additionally, you will provide IP Telephony and VoIP Subject Matter Expertise for the company and its managed service providers, as well as manage 3rd party telecom carriers and providers. To be successful in this role, you should have at least 5 years of hands-on industry experience in telecommunications. You should possess strong conceptual knowledge and experience with telephony protocols like SIP, SDP, RTP, SRTP, WebRTC, and audio/video codecs. In-depth working experience with Kamailio, Freeswitch, any of the SIP stack (Sofia, reSIProcate, PJSIP, etc.), ICE Framework (STUN/TURN), and Linux is required. Hands-on experience in writing production quality code using scripting languages like Python, Go, Erlang, etc., is essential. Experience in using VoIP testing tools like Wireshark, VoIPMonitor, SIPp, SIPCapture, Homer, etc., will be beneficial. Nice to have skills include working knowledge of NoSQL databases like MongoDB, Redis, Cassandra, a passion for knowing everything about VoIP Protocol standards & related RFCs, and experience with virtualization/container-related technologies such as Xen, VMware vSphere/ESXi, Docker, Kubernetes.,