40 Voip Protocols Jobs

Key Responsibilities: Design, deploy, and maintain voice communication solutions using Cisco Control Hub and Webex Cloud. Manage and optimize voice and collaboration platforms including Genesys cloud & On-Prem -based contact center solutions. Working knowledge of Genesys on-prem which is installed on Windows platform, including IVR configuration and call flows. Basic understanding of Windows server and Linux CLI interface. Administer VoIP infrastructure to ensure high availability, security, and optimal performance. Troubleshoot voice-related issues and provide advanced-level support for end users. Collaborate with cross-functional teams to implement telephony-related projects and upgrades. Maintain comprehensive documentation for system configurations, processes, and troubleshooting guides. Evaluate emerging technologies to recommend improvements for enterprise communication strategies. Required Skills & Qualifications: Strong hands-on experience with Cisco Control Hub Cloud and Webex Cloud Services . Working knowledge of Genesys Cloud (formerly PureCloud) platform, including IVR configuration and call flows. Solid understanding of SIP, VoIP protocols, and real-time media troubleshooting. Familiarity with network fundamentals such as QoS for voice, firewalls, and VPNs. Proficiency in using diagnostic tools like packet captures and call analytics platforms. Experience with enterprise telephony integrations (e.g., CRM, ticketing systems). Preferred Qualifications: Cisco certifications (e.g., CCNP Collaboration , Cisco Webex Certified ). Genesys certifications or prior experience in a high-volume call center environment. Scripting or automation experience using Python or similar tools is a plus. Mandatory Skills: Network Voice and Collabration Admin . Experience: 1-3 Years .

You will be responsible for the following tasks in this role: - Creating custom telephony applications and features using Asterisks APIs, scripting languages, and modules. - Integrating Asterisk with other systems, databases, and third-party services to create comprehensive communication solutions. - Developing call routing logic, interactive voice response (IVR) systems, voicemail solutions, call forwarding, and other call handling functionalities. - Implementing and managing VoIP protocols such as SIP (Session Initiation Protocol) and RTP (Real-Time Transport Protocol). - Ensuring the reliability, performance, and security of Asterisk applications through rigorous testing and effective debugging. No additional details of the company were provided in the job description.,

Core Responsibilities: Day-to-day support and maintenance of Genesys PureCloud Solutions MACD on Genesys, including call flows, SIP trunks etc Act as L3 support during issues and drive towards resolution Work with various internal external application teams on integrating their platform with Genesys Migration and implementation of Genesys PureCloud Solution Monitor project progress and adjust as necessary to ensure successful project delivery. Creation of Documentation and Knowledge base Responsible for managing incident and service requests Manage and coordinate team activities, working closely with different support groups. Conduct regular reviews of the platform to ensure optimal performance and user satisfaction. Act as the primary point of contact for stakeholders regarding Genesys PureCloud projects. Skills and Experience: Hands on experience on troubleshooting Genesys PureCloud solutions Good knowledge of APIs and integrating with 3rd party applications. Experience in migration and implementation of Genesys PureCloud Strong understanding of business requirements and ability to convert to solutions. Knowledge of Call flows, Inbound, Outbound Campaigns, Call Recordings and Call Monitoring Good knowledge of Gateway configuration and Genesys SIP Trucking and Dial Plans. Able to troubleshoot and identify the root causes via reading logs and other Genesys tools for troubleshooting. Can explain the issue clearly to the Vendor and provide necessary information correctly if the Vendors need it Knowledge of network routing, VOIP Protocols, SIP Responses, Handshake communication etc.

Responsibilities: * Manage international sales pipeline from lead generation to closure. * Develop new international business opportunities through cold calling and relationship building.

Job Title: Customer Service Executive International Process Location: Gurugram Job Type: Full-time Shift Timings: Night Shift / Rotational Shifts (as per international time zones) Job Summary: We are looking for a dynamic and customer-focused individual to join our international customer service team. As a Customer Service Executive, you will be responsible for resolving queries from global clients via phone, email, or chat, ensuring a high level of customer satisfaction and professionalism. Key Responsibilities: Include responsibility for providing technical troubleshooting over the phone for Intermedia customers globally Specify handling of basic technical assistance calls related to telecom, VoIP, networking and communications Add the following under required competencies: Basic networking knowledge Intermediate knowledge of computers (software and hardware) Technical Skills: Computer knowledge: Intermediate or advance knowledge of computer Hardware and Software Operating Systems: Intermediate knowledge of computer OS both Windows and Mac and Mobile OS Android and iOS Networking: Intermediate knowledge of networking Key Requirements: Education: Key Requirements: Education: Under Graduation / Graduation Experience: Experience – Fresher or Experience both can apply (BPO and Service industry experience preferred) Communication Skills: Excellent verbal and written English communication is a must. Willingness to work in night shifts and rotational shifts. Good computer skills (MS Office, email, CRM tools). Ability to handle pressure and multitask effectively. Customer-focused with a problem-solving attitude.

Miratech helps visionaries change the world by providing global IT services and consulting that bridges the gap between enterprise and start-up innovation. Supporting digital transformation for some of the world's largest enterprises, Miratech partners with large and small players to remain at the forefront of technology. Upholding a culture of Relentless Performance, Miratech ensures the success of over 99% of engagements since its inception in 1989. Covering 5 continents and operating in over 25 countries, Miratech boasts nearly 1000 full-time professionals and sustains an annual growth rate exceeding 25%. Join the revolution in customer experiences with our client, a global leader in cloud contact center software. Empowering businesses worldwide to provide seamless, personalized, and joyful customer interactions through cloud innovation, we are seeking a Telecom Engineer to enhance voice call capabilities for advanced call and contact center solutions. The Voice Client team plays a pivotal role in delivering high-quality voice call capabilities, ensuring seamless connectivity, crystal-clear audio, and exceptional reliability to enhance customer engagement globally. As a Telecom Engineer, you will be responsible for designing, deploying, and maintaining our client's global voice infrastructure. Responsibilities include collaborating with customer telecom and IT teams, building and maintaining SIP Trunk connectivity, providing operational support for the telecom network, and serving as an escalation point for critical alerts. The ideal candidate will possess expertise in VoIP/SIP technologies, Ribbon (Sonus) SBCs, and hands-on experience in troubleshooting complex network issues. Qualifications for the role include 2+ years of hands-on experience with Ribbon PCX or SBC, solid knowledge of SIP, experience in telecom troubleshooting, and strong communication skills. Additional qualifications such as WebRTC and TLS knowledge are a plus. Desired qualifications encompass experience with international routing, knowledge of VoIP protocols and codecs, familiarity with network security, hands-on experience with AudioCodes SBCs, and a background in designing high-availability telecom environments. Miratech offers a Culture of Relentless Performance, a Work From Anywhere Culture, Growth Mindset opportunities, Global Impact collaborations, a Welcoming Multicultural Environment, and Social Sustainability Values focused on IT education, community empowerment, fair operating practices, environmental sustainability, and gender equality.,

Miratech helps visionaries change the world by providing global IT services and consulting that merge enterprise and start-up innovation. Supporting digital transformation for some of the world's largest enterprises, Miratech partners with large and small players to stay at the forefront of technology and create solutions that enhance business operations. With a values-driven culture of Relentless Performance, Miratech has achieved success in over 99% of engagements since its establishment in 1989. Operating across 5 continents and in over 25 countries, Miratech boasts a team of nearly 1000 full-time professionals and an annual growth rate exceeding 25%. The company invites individuals to join in revolutionizing customer experiences with a global leader in cloud contact center software, empowering businesses worldwide to deliver seamless, personalized, and joyful customer interactions through cloud innovation. The Voice Client team at Miratech focuses on delivering essential voice call capabilities for customer call/contact center solutions, emphasizing smooth and high audio quality to enhance customer experiences. The team develops native applications for Mac OS and Windows OS, as well as web-based applications utilizing the latest web technologies. By directly engaging with customers, the team addresses concerns promptly and leverages feedback to continually enhance the product for exceptional customer experiences and added business value. Responsibilities: - Contribute to the robust development of the global voice platform through comprehensive design, development, and testing processes to ensure high availability and scalability. - Write high-quality code in C++ adhering to best practices and participate in code reviews to maintain code integrity. - Enhance system capabilities for compatibility with the Google Cloud platform, troubleshoot solutions, and provide detailed Root Cause Analysis reports. - Contribute to CI/CD automation pipelines for voice services, respond promptly to production issues, and create technical documentation for system maintenance. - Collaborate with peer development teams, apply performance and scalability principles, and optimize system efficiency. Qualifications: - 5+ years of industry experience in software development and troubleshooting VOIP protocols. - Expertise in C++ programming on Windows and macOS platforms, along with frontend functionality using Javascript. - Proficiency in troubleshooting tools, audio media analysis, and debugging techniques. - Experience in Agile/Scrum methodologies, a Bachelor's degree in a relevant discipline, and excellent communication skills. - Positive, self-motivated, and enthusiastic attitude with a strong sense of ownership. Nice to have: - Experience with virtual environments like Citrix and VMware, familiarity with javascript and Node.js, and knowledge of Contact Centre, IVR, Virtual Agents, and telecommunications service providers. Miratech offers a Culture of Relentless Performance with a 99% project success rate, remote work flexibility, professional development opportunities, impactful projects for global clients, a welcoming multicultural environment, and a focus on sustainable business practices across five pillars.,

Responsibilities: * Provide technical onsite support for routing protocols, VoIP protocols, SIP trunking, IP telephony, IP PBX, IP networking, home automation, WiFi, and onsite support. Provident fund Travel allowance

You have 2+ years of experience in supporting Cisco UCCE / UCCX / PCCE solutions & 3rd party Call recording platforms. Your responsibilities will include basic Cisco ICM/CCMP/CVP/CUIC troubleshooting, MACD creation in Cisco UCCE & IPT platforms, and a clear understanding of VOIP protocols like SIP, H.323, and MGCP. It is essential that you are familiar with the ITIL process. In this role, you will need a clear understanding of Cisco Voice network deployment models and the functioning of voice network devices. Your knowledge and use of VOIP and Performance Management Tools, as well as integration experience with other tools, will be crucial. Understanding of Telecom infrastructure protocols such as ISDN and analogy signalling is also required. Your role will involve working in a 24/7 business support environment. Previous experience on Cisco Voice platforms, particularly CUCM/UCCE/UCCH/CVP platforms, will be advantageous.,

This is a remote position. As a proactive and innovative Prompt Engineer Intern, you will have the opportunity to work remotely and collaborate with our AI and engineering teams. Your primary responsibility will involve designing, testing, and refining prompts that enhance the performance of our conversational assistants. By leveraging your creativity and technical skills, you will play a vital role in improving the overall user experience by ensuring accurate, relevant, and engaging responses from our assistants. In this role, your key responsibilities will include: - Design & Experimentation: Creating and testing various prompt strategies to optimize AI assistant performance. - Collaboration: Working closely with cross-functional teams to understand requirements, gather feedback, and iterate on prompt designs. - Data Analysis: Analyzing the effectiveness of different prompts using performance metrics and user feedback. - Documentation: Maintaining detailed records of experiments, findings, and insights to drive future improvements. - Research & Innovation: Staying updated with the latest trends and research in NLP and prompt engineering to continually enhance our strategies. Requirements: - Interest in AI: Demonstrated passion for AI, machine learning, and natural language processing through academic projects, research, or personal initiatives. - Experience: Previous experience with prompt engineering or NLP projects is advantageous but not mandatory. - Work Ethic: Ability to work independently, manage time effectively, and thrive in a remote work environment. Benefits: - Remote Flexibility: Enjoy the freedom to work from anywhere with flexible hours that suit your schedule. - Hands-on Experience: Gain practical exposure to prompt engineering and contribute to cutting-edge AI assistant development. - Mentorship & Learning: Collaborate with experienced professionals in AI and engineering, fostering opportunities for growth and learning. - Networking: Engage with industry experts and participate in innovative projects with a direct impact on improving AI technologies. - Potential for Full-Time Role: Exceptional interns may be considered for full-time positions upon successful completion of the internship. Required Skills and Experience: - Proven experience with Asterisk PBX installation, configuration, and customization. - Strong knowledge of VoIP protocols such as SIP, RTP, IAX, and telephony standards. - Proficiency in developing and debugging custom dial plans and call flows. - Experience with integrating Asterisk with databases (e.g., MongoDB, PostgreSQL). - Familiarity with scripting languages like Python or PHP for telephony applications. - Knowledge of QoS, echo cancellation, and other VoIP performance tuning techniques. - Basic understanding of network configurations, including firewalls, NAT, and VPNs. Preferred Skills: - Experience with FreePBX or other Asterisk GUIs. - Hands-on experience with Linux/Unix environments for server administration. - Familiarity with cloud telephony platforms (e.g., Twilio, Plivo) and APIs. - Exposure to AI voice systems or IVR integration with conversational AI platforms. - Knowledge of WebRTC and real-time communications tools. - Certifications in VoIP or telecommunication systems (e.g., Digium, CCNA Voice).,

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

As a Technical Support Lead at Vitel Global Communications Pvt Ltd in Hyderabad, you will be responsible for overseeing a team of technical support engineers to ensure the delivery of high-quality VoIP services and maintain customer satisfaction. Your main responsibilities will include team leadership, technical support, client interaction, process improvement, and documentation. In terms of team leadership, you will lead, mentor, and manage a team of technical support engineers, providing guidance and support to ensure high-quality performance. You will also be required to provide technical guidance and escalate complex technical issues to the appropriate teams when necessary. As the primary point of contact for escalated technical issues, you will interact with clients to ensure timely resolution and minimal disruption to operations. Additionally, you will be responsible for identifying opportunities to enhance internal processes that promote best practices and improve overall performance and organizational efficiency. Maintaining comprehensive documentation of network configurations, procedures, and troubleshooting guides will also be a key part of your role. The ideal candidate for this position should have 5 to 9 years of experience in technical support, with at least 2 years in a leadership position. A Bachelor's degree in Telecommunications, Computer Science, Information Technology, or a related field is required. Required skills include a strong knowledge of VoIP protocols and technologies, experience with VoIP hardware and software, familiarity with networking concepts and technologies, excellent troubleshooting and problem-solving skills, strong communication skills, and the ability to work both independently and as part of a team. A customer-focused mindset with a dedication to providing high-quality support is also essential. This position requires working night shifts from 7:30 PM to 4:30 AM IST at the office in Hyderabad, Telangana. If you are passionate about leading a team, providing technical support, and ensuring customer satisfaction in the VoIP industry, this role could be a perfect fit for you.,

You are a skilled Voice / VoIP Engineer with over 3 years of experience in SIP, VoIP protocols, and IP telephony. Your main responsibilities will include designing, implementing, and troubleshooting VoIP networks to ensure optimal voice quality and system performance. Your key responsibilities will involve configuring and managing IP-PBX, SIP trunking, and VoIP systems, resolving issues related to ASR, ACD, PDD, CLI, DTMF, codecs, and voice quality, performing packet trace (PCAP) and CDR analysis for diagnostics, handling fraud monitoring, SIP Trunk down, and voice escalations, as well as providing support to US clients with 24/7 availability. To excel in this role, you must possess a strong knowledge of TDM, VoIP, SIP, and network security, along with hands-on experience in voice gateways and cloud PBX. Additionally, excellent communication skills are a must-have for effective interaction with clients and team members.,

Join Miratech in revolutionizing customer experiences with our client, a global leader in cloud contact center software. Together, we bring the power of cloud innovation to enterprises worldwide, empowering businesses to deliver seamless, personalized, and joyful customer interactions. The Voice Client team at Miratech plays a critical role in delivering high-quality voice call capabilities for advanced call and contact center solutions. Our team is dedicated to providing seamless connectivity, crystal-clear audio, and superior reliability to enhance customer engagement. Through our work, we directly influence the experience of millions of users globally, making voice communication smarter, faster, and more resilient. As a Senior Telecom Engineer at Miratech, you will play a vital role in delivering high-quality voice call capabilities for advanced call and contact center solutions. In this position, you will serve as the technical expert responsible for designing, deploying, and maintaining our client's global voice infrastructure. Your expertise in VoIP/SIP technologies, telecom carrier connectivity, and Ribbon (Sonus) SBCs will be crucial in troubleshooting complex network issues and ensuring seamless voice communication for users worldwide. Responsibilities: - Collaborate with customer telecom and IT teams to deploy customized solutions and troubleshoot issues. - Build and maintain SIP Trunk connectivity with customers and carriers, including interop sessions and activations. - Provide operational support for the telecom network, analyze incidents, and implement preventive measures. - Serve as an escalation point for critical alerts from SBCs and deliver root-cause analyses for outages. - Manage telecom service providers and vendors, and oversee hardware/software deployments and new service rollouts. - Develop testing plans, create technical documentation, and maintain SOPs for recurring tasks. - Mentor junior engineers in troubleshooting and managing complex service issues. - Understand product capabilities, limitations, and contribute to continuous improvements. Qualifications: - 5+ years of telecom engineer experience with VoIP/SIP voice applications. - Strong knowledge of voice/data communications (SIP, TCP/IP, MPLS), VoIP protocols (H.248, G.711, G.729, WebRTC), and security (TLS, IPSEC, ACLs). - High level knowledge of VoIP principles, protocols and CODECs such as H.248, SIP, G.711, G.729, WebRTC, MPLS, VPN, UDP, RTP, MTP etc. - Experience with international routing (ITFS, iDID) and telephony design for high availability (99.99% SLA). - Hands-on experience with Softswitches, SBCs (Sonus/Ribbon, AudioCodes), SIP proxies, and media servers (AudioCodes IPM-6310, FreeSWITCH). - Skilled in Wireshark, Empirix, and RTP stream analysis (MOS, Jitter, Latency, Packet Loss). - Ability to design, troubleshoot, and maintain complex global voice networks. - Strong organizational skills and experience implementing telecom architecture changes in lab and production. Join us at Miratech and be part of an unstoppable technology development team with a 99% project success rate and over 30% year-over-year revenue growth. Enjoy the flexibility of remote work, professional development opportunities, impactful projects for top global clients, and a welcoming multicultural environment focused on sustainability and inclusion.,

You should have a minimum of 3+ years of experience in the field of Voice over IP (VoIP) and related technologies. Your hands-on experience should include working with major platforms such as Cisco CUCM, Unity, UCCX, Expressway, Avaya Aura, Microsoft Teams Voice, and Ribbon / SBCs. It is essential to have a strong knowledge of VoIP protocols like SIP, H.323, and RTP. Additionally, familiarity with QoS, VLANs, and WAN optimization for voice communication is necessary. You should also possess an understanding of network protocols and troubleshooting tools such as Wireshark and TCPDump. Having relevant certifications like Cisco CCNP Collaboration, Avaya ACIS / ACSS, or Microsoft Teams Voice Engineer Associate would be preferred. In this role, your responsibilities will involve configuring, managing, and troubleshooting various VoIP systems and platforms. You will be expected to work with different network protocols and ensure the quality and reliability of voice communication over networks. The ideal candidate for this position would have at least 2 years of practical experience in Voice over IP (VoIP), Wireshark, and TCPDump, and a minimum of 3 years of experience working with Cisco Unified Communications Manager (CUCM). This is a full-time, permanent position that requires in-person work. If you meet the experience and skill requirements outlined above and are ready to take on the challenges of managing and optimizing voice communication systems, we encourage you to apply for this role.,

Miratech is a global IT services and consulting company that partners with both enterprise and start-up innovators to support digital transformation for some of the world's largest enterprises. By staying at the forefront of technology and creating innovative solutions, we help our clients enhance their businesses. With a global presence across 5 continents and operations in over 25 countries, Miratech consists of nearly 1000 full-time professionals and maintains an annual growth rate exceeding 25%. Join us in revolutionizing customer experiences with our client, a global leader in cloud contact center software. Our team at Miratech brings the power of cloud innovation to enterprises worldwide, empowering businesses to deliver seamless, personalized, and joyful customer interactions. The Voice Client team at Miratech plays a crucial role in delivering high-quality voice call capabilities for advanced call and contact center solutions. We are committed to providing seamless connectivity, crystal-clear audio, and superior reliability to enhance customer engagement. Our work directly impacts the experience of millions of users globally, making voice communication smarter, faster, and more resilient. As a Senior Telecom Engineer at Miratech, you will be a key player in delivering high-quality voice call capabilities for advanced call and contact center solutions. In this role, you will serve as the technical expert responsible for designing, deploying, and maintaining our client's global voice infrastructure. Your responsibilities will include collaborating with customer telecom and IT teams to deploy customized solutions, building and maintaining SIP Trunk connectivity with customers and carriers, providing operational support for the telecom network, serving as an escalation point for critical alerts, managing telecom service providers and vendors, developing testing plans, creating technical documentation, and mentoring junior engineers. To qualify for this role, you should have at least 5 years of experience as a telecom engineer with expertise in VoIP/SIP voice applications, strong knowledge of voice/data communications, VoIP protocols, security, international routing, telephony design for high availability, hands-on experience with various technologies, and the ability to design, troubleshoot, and maintain complex global voice networks. At Miratech, you will be part of a culture of Relentless Performance, have the opportunity to work remotely, benefit from professional development opportunities, collaborate on impactful projects, thrive in a multicultural environment, and contribute to social sustainability values focused on IT education, community empowerment, fair operating practices, environmental sustainability, and gender equality.,

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.,

RESPONSIBILITIES (INCLUDES TASKS AND AUTHORITIES): Responsibilities: Responsible to lead A team for Contact Centre with project delivery and solar-wind (total observability software of IT assets) projects. Handle Level 2 and L3 troubleshooting for Contact Centers, , VoIP Protocols, Omni-Channel Interactions, QA and QM Modules. Handle SolarWinds implementation and support Pre-sales support on above solution areas as well as preparing SOW Oversee IVR Builds, Complex Integrations, API builds, and CRM Integrations. Provide problem resolution, Root Cause Analysis, and Action Plans. Requirements: Minimum 5 years of experience in a similar role within an SI Company. Strong knowledge of ICT Infrastructure solutions, Telco, and Network Layer. Experience with Contact Centers technologies, networking and voice Ability to analyze problems and provide robust solutions. Strong understanding of monitoring networks and Contact Center Applications, Excellent communication, and organizational skills. Relevant industry certifications (Contact Centers, Microsoft Technologies, Network,). TECHNICAL SKILLS /COMPETENCIES: MANDATORY Technical and business communication technical skill in Network, Voice and Contact centre People management and team leadership SOFT SKILLS: MANDATORY Good Communication ability to communicate and articulate the message effectively and engage the person/s well during the conversation. Customer service and relationship management. Problem Solving ability to understand & diagnose the problem and propose solutions in the best interest of the organization & employees. Passion for Results Demonstrate the sense of urgency to deliver results Show more Show less

Miratech helps visionaries change the world by providing global IT services and consulting, combining enterprise and start-up innovation. Supporting digital transformation for leading enterprises, Miratech partners with large and small players to remain at the forefront of technology. With a values-driven culture focused on Relentless Performance, over 99% of Miratech's engagements have successfully met or exceeded objectives since 1989. Covering 5 continents and operating in 25+ countries, Miratech boasts nearly 1000 full-time professionals and an annual growth rate exceeding 25%. Join the team in revolutionizing customer experiences with a global leader in cloud contact center software, enabling seamless, personalized, and joyful interactions for businesses worldwide. The Voice Client team is dedicated to delivering essential voice call capabilities for customer call/contact center solutions, prioritizing high audio quality for an enhanced customer experience. Developing applications for Mac OS, Windows OS, and web-based platforms using cutting-edge technologies, the team interacts directly with customers to improve products continually. **Responsibilities:** - Contribute to the robust development of the global voice platform through design, development, and testing, ensuring high availability and scalability. - Write high-quality C++ code following best practices and engage in code reviews to maintain integrity and quality. - Enhance system capabilities for compatibility with the Google Cloud platform. - Troubleshoot solutions, identify root causes, and provide detailed analysis reports. - Contribute to CI/CD automation pipelines for customer service builds, deployments, and upgrades. - Respond promptly to production issues and collaborate to restore services efficiently. - Create technical documentation to facilitate system maintenance and troubleshooting. - Communicate effectively with peer development teams to foster collaboration and knowledge sharing. - Apply performance and scalability principles in software development to optimize system efficiency. **Qualifications:** - 5+ years of industry experience in software development and troubleshooting VOIP protocols (SIP, RTP, RTCP, WebRTC). - Expertise in C++ programming on Windows and macOS platforms. - Hands-on experience in frontend functionality using JavaScript. - Proficiency in troubleshooting tools, audio media analysis, and debugging techniques. - Experience with Agile/Scrum methodologies. - Bachelor's degree in a relevant discipline. - Positive, self-motivated attitude with strong communication skills. **Nice to have:** - Experience with virtual environments like Citrix and VMware. - Familiarity with JavaScript, Node.js. - Experience with Contact Centre, IVR, Virtual Agents, and telecommunications service providers. **We offer:** - Culture of Relentless Performance with a 99% project success rate and significant revenue growth. - Work From Anywhere Culture providing flexibility with remote work. - Growth Mindset with professional development opportunities. - Global Impact working on impactful projects for top clients. - Welcoming Multicultural Environment fostering inclusivity and support. - Social Sustainability Values focusing on IT education, community empowerment, fair practices, environmental sustainability, and gender equality.,

As a Telecom Infrastructure Engineer at Ubona Technologies in Bangalore, you will be part of a pioneering SaaS startup specializing in full-stack customer conversation solutions for large enterprises. Ubona Technologies aims to revolutionize consumer communication through cutting-edge AI/ML technologies, enhancing efficiency and reducing friction in enterprise-customer interactions. Our state-of-the-art, cloud-based solutions power millions of interactions daily for leading banks, insurance companies, NBFCs, e-commerce ventures, and telecom operators. In this role, you will work on cutting-edge telephony and VoIP solutions, focusing on SIP servers, dialer systems, and optimizing VoIP performance. We are seeking a skilled SIP/Dialer Software Engineer with a minimum of 2 years of hands-on experience in configuring and managing telephony systems like Asterisk and Kamailio. Your responsibilities will include optimizing VoIP performance, troubleshooting system issues, and ensuring seamless communication for call center operations. Key Responsibilities: - SIP Server Management: Install, configure, and manage Asterisk or similar technology, Kamailio for seamless SIP trunking. - Dialer Setup & Optimization: Configure and maintain predictive, progressive, manual, and preview dialers. - Asterisk/Freeswitch Expertise: Provide high-level support and troubleshooting to ensure system stability. - VoIP Performance Optimization: Improve RTP/SIP signaling, reduce latency, and enhance call quality. - Business Systems Integration: Connect dialers with CRMs, databases, and third-party platforms. - Security & Compliance: Keep systems updated with security patches and regulatory standards. - Collaboration & Support: Work closely with network engineers and developers to resolve issues. Skills required for this role include SIP Server Expertise, Dialer Systems Knowledge, VoIP Protocols understanding, Analytical & Problem-Solving Skills, Network Knowledge, and Excellent Communication. Preferred qualifications include a Bachelor's degree in Computer Science, Telecommunications, or a related field, experience with cloud-based/on-prem telephony solutions, knowledge of high-availability and clustering for telephony platforms, familiarity with compliance standards, and experience with load testing and scalability for VoIP systems. At Ubona Technologies, we offer a collaborative, inclusive, and dynamic work environment where you will have the opportunity to shape innovative customer communication solutions and grow professionally. If you are passionate about telephony technologies and have a strong understanding of SIP protocols, we encourage you to apply and be part of our innovative team.,

Key Responsibilities: Design, implement, and maintain VoIP and telephony solutions across enterprise environments. Manage and troubleshoot unified communication platforms including Cisco, Avaya, or Microsoft Teams. Configure and support SIP, H.323, and MGCP protocols. Collaborate with IT teams to integrate telephony with office automation and business applications. Monitor system performance and ensure high availability and quality of service. Provide technical support and training to end-users and internal teams. Required Skills: Strong knowledge of VoIP protocols (SIP, H.323, RTP) Experience with Cisco Unified Communications , Avaya , or Microsoft Teams Familiarity with PBX systems , call routing , and dial plan configuration Understanding of networking fundamentals (TCP/IP, DNS, DHCP) Experience with office automation tools and IT service management Preferred Skills: Certifications like CCNA Voice , CCNP Collaboration , or Avaya Certified Specialist Experience with cloud-based telephony solutions (e.g., Zoom Phone, RingCentral) Knowledge of ITIL processes and ticketing systems (e.g., ServiceNow)

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

4 - 6 years of relevant experience supporting Information Security architecture and technologies in large enterprise networks. Architecture of large-scale network and application environments (including Internet-facing) Familiar with cloud infrastructure and related technologies Solid understanding of the TCP/IP protocol suite and related concepts and technologies and IPv6 Experience with router and multilayer switch technologies and products Experience with stateful firewall technologies and products, NextGen firewalls (e.g. Checkpoint, Juniper, Cisco, Palo Alto) and firewall management tools Experience with key security tools such as Antivirus, File Integrity Monitoring, Intrusion Detection/Prevention, Data Loss Prevention, Web Application Firewalls, Web Proxies, Web Content Filtering; DDoS defence. Solid experience with system engineering, installation and administration of *NIX-based and Windows-based operating systems, system hardening and troubleshooting Ability to support Incident Response processes Familiarity with major application protocols (HTTP, HTTPS, SMB, FTP/SFTP, SMTP, NTP, SNMP, VoIP protocols, etc.) Strong understanding of authentication and encryption technologies Primary Skills Experience with stateful firewall technologies and products, NextGen firewalls (e.g. Checkpoint, Juniper, Cisco, Palo Alto) and firewall management tools Experience with key security tools such as Antivirus, File Integrity Monitoring, Intrusion Detection/Prevention, Data Loss Prevention, Web Application Firewalls, Web Proxies, Web Content Filtering; DDoS defence. Solid experience with system engineering, installation and administration of *NIX-based and Windows-based operating systems, system hardening and troubleshooting Familiarity with major application protocols (HTTP, HTTPS, SMB, FTP/SFTP, SMTP, NTP, SNMP, VoIP protocols, etc.) Strong understanding of authentication and encryption technologies

As an Application Developer, you will be responsible for creating custom telephony applications and features using Asterisk's APIs, scripting languages, and modules. Your role will involve system integration by connecting Asterisk with other systems, databases, and third-party services to develop comprehensive communication solutions. You will be developing call routing logic, interactive voice response (IVR) systems, voicemail solutions, call forwarding, and other call handling functionalities to optimize communication processes. Additionally, you will be implementing and managing VoIP protocols such as SIP (Session Initiation Protocol) and RTP (Real-Time Transport Protocol) to ensure seamless communication. Your responsibilities will also include testing and debugging to guarantee the reliability, performance, and security of Asterisk applications. Through rigorous testing and effective debugging practices, you will contribute to the stability and efficiency of the telephony applications developed. Overall, your expertise in application development, system integration, call routing, protocol implementation, and testing will play a vital role in enhancing communication capabilities and delivering high-quality solutions to clients.,