Top FreeSWITCH Alternatives

3CX is a versatile, software-driven IP PBX built on open standards that delivers comprehensive Unified Communications straight out of the box. It caters to businesses of all sizes and across various sectors, offering a full suite of features that range from mobility and presence indicators to sophisticated call center functionalities, all at a significantly reduced cost. This platform is particularly well-suited for remote work environments, featuring mobile applications, video conferencing capabilities, and live chat integration for websites. Installation, management, and maintenance of the PBX are straightforward, allowing users to easily handle it themselves on their own hardware, a dedicated server, or via a cloud environment. Alternatively, users can opt for 3CX's hosting services to simplify their setup even further. With its user-friendly design and extensive feature set, 3CX empowers companies to enhance their communication strategies effectively.

PBXware stands out as the original and most prominent IP PBX Professional Open Standards Turnkey Telephony Platform available today. Since its inception in 2004, PBXware has been delivering adaptable, dependable, and scalable Next Generation Communication Systems (NGCS) alongside VoIP solutions tailored for small and medium-sized businesses (SMBs), large enterprises, Internet Telephony Service Providers (ITSPs), call centers, and various governmental agencies across the globe. This achievement is realized by integrating the finest elements of cutting-edge technologies. The Bicom Systems softswitch is offered in several editions, including Business, Call Center, and Multi-Tenant, with each edition designed to support specific features that enhance performance, reliability, and expandability, ensuring users get the most out of their systems. This versatility makes PBXware a preferred choice for organizations seeking robust telephony solutions.

Yeastar P-Series Phone System is a business communication solution that offers companies of all sizes with a complete package for calls, video, messaging and integrations, out of the box. With inbuilt visual call management, integrated video conferencing, advanced contact center features, and ready-made SMS, WhatsApp, Microsoft Teams, CRMs, and more platform integrations, it boosts user experience at all levels and provides everything across desktop, mobile, and browser with simple user apps.

FusionPBX acts as a multifaceted solution suitable for both single and multi-tenant PBX configurations, functioning effectively as a carrier-grade switch, call center system, fax server, VoIP server, voicemail server, conference server, and voice application server, among various other functions. In contrast, FreeSWITCH™ is recognized for its flexibility and scalability, operating seamlessly across multiple threads and platforms. It offers a rich set of features, such as unlimited extensions, voicemail-to-email, music on hold, call parking, call center functions, call queues, and phone provisioning, making it ideal for businesses of varying sizes. By providing enterprise-grade phone system capabilities, it effectively addresses the diverse requirements of small, medium, and large organizations alike. Considerable resources have been dedicated to bolstering security, addressing technical issues, and enhancing the codebase. In addition, there are continuous improvements underway in anticipation of a forthcoming release, with the master branch exhibiting advancements that surpass the current version in multiple areas. As the team diligently finalizes critical updates, excitement builds regarding the forthcoming enhancements, which promise to elevate user experience. This dedication to ongoing enhancement is a testament to the platform's commitment to staying competitive and relevant amid the fast-paced changes in the telecommunication industry, ensuring it meets the ever-evolving demands of its users.

Kamailio® is a powerful open-source SIP server that emerged from the earlier versions of OpenSER and SER, and is released under GPLv2+, designed to efficiently handle thousands of call setups per second. This versatile platform is ideal for building large-scale VoIP systems, facilitating real-time communication, managing presence, supporting WebRTC, enabling instant messaging, and accommodating numerous other applications. Moreover, Kamailio excels in scaling SIP-to-PSTN gateways, PBX systems, and media servers, such as Asterisk™, FreeSWITCH™, or SEMS, with remarkable ease. Its rich feature set encompasses support for asynchronous protocols like TCP, UDP, and SCTP, ensuring secure communication through TLS for voice, video, and text, in addition to compatibility with WebSockets for WebRTC. The server also fully supports both IPv4 and IPv6, offers straightforward instant messaging and presence functionalities via an integrated XCAP server and MSRP relay, and facilitates asynchronous operations effortlessly. Additionally, it includes IMS extensions for VoLTE, ENUM capabilities, along with features for DID and least cost routing, load balancing, and routing failover to enhance reliability. Kamailio's comprehensive suite of accounting, authentication, and authorization features ensures robust security, while it seamlessly integrates with a variety of backend systems such as MySQL, Postgres, Oracle, Radius, LDAP, Redis, Cassandra, MongoDB, and Memcached. With all these advanced capabilities, Kamailio stands out as a highly adaptable solution for contemporary communication infrastructures, making it a preferred choice among developers and organizations alike.

Asterisk is a dynamic open-source framework that facilitates the development of a variety of communication applications. It enables a standard computer to function as a powerful communications server. Asterisk is integral to IP PBX systems, VoIP gateways, conference servers, and many other customized solutions. Its user base includes small businesses, large corporations, call centers, telecom operators, and government entities around the world. Provided free of charge, Asterisk is backed by Sangoma's support. Currently, there are over a million Asterisk-powered communication systems in operation across more than 170 countries. Notably, virtually all companies within the Fortune 1000 list leverage Asterisk in some form. Frequently adopted by system integrators and developers, Asterisk can serve as the basis for a complete business phone system or be used to enhance an existing setup or link various systems. Whether you aim to design a bespoke system using Asterisk or are interested in acquiring an effective and affordable turnkey solution, there are numerous options to explore that can help you find the perfect match for your requirements. As it adapts to technological advancements and user needs, Asterisk remains a key player in the evolving landscape of modern communication solutions. Its ongoing development ensures users continue to benefit from innovative features and improved performance.

VitalPBX is a comprehensive PBX platform built on Asterisk and designed to operate on Centos 7. This powerful and adaptable platform enables users to effortlessly manage their PBX systems with ease. With an intuitive user interface complemented by helpful tooltips, navigating through the system is straightforward. It is essential for telephony systems to remain vigilant about potential security vulnerabilities and implement measures to mitigate these risks. VitalPBX provides multiple layers of defense to safeguard against cyber threats in the realm of VoIP communications. Moreover, its security features ensure that users can maintain a reliable and secure communication environment.

ICTPBX is a powerful open-source PBX solution tailored for Internet Service Providers (ISPs), Internet Telephony Service Providers (ITSPs), and various service-oriented businesses. At its core, it utilizes FreeSWITCH 1.10 for media processing, combined with FusionPBX 6.6 for configuration management, all accessible through the ICTCore REST API and managed via an user-friendly Angular 13 web interface. Unlike hosted PBX alternatives, ICTPBX runs entirely on your infrastructure, providing complete autonomy over your data, client accounts, and billing systems. Its flexible architecture supports scaling from a handful of extensions to thousands of users without the burden of per-user licensing fees, making it an economical choice for many organizations. This platform is engineered for multi-tenant configurations, guaranteeing that each tenant enjoys their own distinct PBX environment, complete with custom branding, preset limits, and role-specific access controls for improved security and personalization. Notable features comprise support for SIP extensions, ring groups, call queues (ACD), and interactive voice response (IVR) systems, in addition to voicemail, conferencing capabilities, time-based conditions, and intricate call routing. It also facilitates SIP trunks (gateways), inbound Direct Inward Dialing (DID) routing, and call blocking for enhanced operational efficiency. Furthermore, ICTPBX can handle multiple tenants with unique branding and quota management, while seamlessly integrating voice, fax (T.38/FoIP), SMS (SMPP), and email functions into a unified system for efficient communication. This all-encompassing solution effectively addresses the varied demands of contemporary telecommunications, making it a valuable asset for any service provider.

Introducing a revolutionary TelePhony Platform designed specifically for individual entrepreneurs, wholesale service providers, and residential VoIP operators! This platform is available as an open-source solution, enabling anyone to embark on their telephony journey without incurring any initial expenses thanks to ASTPP. Renowned as one of the foremost VoIP Billing Software solutions in the market, ASTPP focuses on improving user experience by providing a holistic telephony solution. This open-source platform has advanced significantly beyond merely being a billing tool, evolving into a comprehensive “Smart TelePhony Platform.” The latest version of ASTPP is equipped with sophisticated functional modules that enhance operational efficiency, thereby eliminating the need for multiple applications to manage a VoIP business. It features integrated hosted IP PBX capabilities, Class 4/5 Softswitch functions, and a powerful invoicing and billing system, all driven by Smart Technology. Users benefit from the ability to manage both origination and termination rates independently, optimizing their profits through least cost routing and cost-based routing techniques. This groundbreaking solution promises to transform the VoIP landscape for both emerging and established enterprises, making telephony more accessible than ever before. With its user-friendly interface and comprehensive features, entrepreneurs can confidently navigate the complexities of the VoIP market.

Cut down on the costs tied to your business communication system by choosing a pay-as-you-go model. Our vast international network provides direct routes that allow you to take advantage of discounts based on volume and usage. With our premium SIP and IAX services, you can enjoy features such as elastic trunking, instant provisioning, direct interconnections, and secure connections. No matter if your workforce includes 25 or 200 agents, we possess the knowledge to meet all your SIP enterprise needs for call termination and origination. CALLR proficiently handles the encoding and decoding of signals, supporting all major codecs in both directions to ensure high-quality communication. Our services integrate seamlessly with leading VoIP platforms like Asterisk, Freeswitch, Vicidial, and FreePBX. Whether your requirements are for outbound or inbound calling, we are ready to address all your telecommunication necessities. Our advanced routing system allows us to present competitive pricing that can greatly reduce your call expenses. Furthermore, our SIP and IAX network has a global reach, guaranteeing dependable service with local interconnection through strategically placed data centers aligned with your network's footprint. Elevate your communication experience with our all-inclusive solutions designed specifically for your business requirements, enabling you to operate more efficiently and effectively.

ALARIS LABS focuses on software development, delivering a wide array of products specifically designed for the voice wholesale and SMS hubbing industries. Voice carriers can leverage ALARIS inVOICE, a sophisticated billing and routing platform that simplifies partner management by managing interconnections, rates, billing, bilateral agreements, DID management, alerting, and analytics, all while enabling real-time routing through both manual and dynamic methods, such as least-cost routing and quality-driven approaches across multiple softswitches and SBCs including Sonus, Sansay, Dialogic, Genband, Mera MVTS, Huawei, VOS, Freeswitch, and others, utilizing the SIP 300/302 API. Moreover, the Alaris SMS Platform efficiently organizes message traffic with a strong framework for overseeing both one-way and two-way A2P and P2P communications across a diverse set of protocols and OTT APIs like SMPP, CMPP, HTTP, SIGTRAN, UCP, ParlayX, RCS, WhatsApp, Viber, and Telegram. This robust message-switching capability, which can handle up to 25,000 SMS messages per second, is complemented by a wide selection of business features tailored to meet various operational requirements. Consequently, ALARIS LABS equips telecommunications companies with a potent suite of tools that can dramatically improve both efficiency and profitability in their operations, ultimately positioning them for success in a competitive market.

The word "Free" in FreePBX embodies the concept of Freedom, underscoring that FreePBX is the foremost open-source IP PBX solution worldwide, which enables users to design a tailored phone system that aligns with their unique needs. Delve into the possibilities of open-source communication and investigate the wide array of features offered by FreePBX that assist in building your perfect phone system. By incorporating commercial modules from Sangoma and a variety of other enhancements, you can transform your FreePBX setup into a robust Unified Communications platform. Remember to consult the Knowledge Base on the FreePBX Wiki for further information! Alternatively, seeking support from FreePBX experts can aid you with installation, updates, troubleshooting, and additional services. FreePBX is accessible for free download and usage, and its strength is bolstered by a global community of developers committed to ensuring its high compatibility and flexibility, which guarantees it encompasses all necessary functionalities required to establish a scalable business phone system for any budget. This community-driven approach not only promotes innovation but also ensures ongoing advancements, solidifying FreePBX as a dependable option for diverse communication requirements. With its extensive resources and community support, users can continually enhance their experience and capabilities.

The *astTECS IPPBX system serves as the perfect solution for businesses of all sizes, ensuring a comprehensive set of high-end features at a reasonable cost while holding TEC certification from the Government of India. This innovative PBX solution empowers organizations to be adaptable, forward-thinking, and competitive, all while streamlining telecom costs. Additionally, the *astTECS Call Center Dialer stands out as an omnichannel contact center solution, featuring a sophisticated real-time dashboard to monitor agents effectively. With the incorporation of advanced analytics and AI-driven capabilities, this dialer significantly enhances both agent productivity and customer interactions. Furthermore, *astTECS offers a robust call recording solution tailored for your business needs, with an intuitive voice recorder that boasts a range of advanced functionalities. Seamless integration with various PBX systems is a key benefit, as it supports ISDN/PRI, IP, analog lines, and mobile networks, ensuring comprehensive recording of all incoming and outgoing calls. This all-in-one solution not only optimizes communication but also facilitates better decision-making through detailed call insights.

CompletePBX, Xorcom's core telephone system solution, serves as a Virtual IP PBX (Private Branch Exchange) or Phone System Appliance, integrating Asterisk and Linux to ensure robust stability and high operational performance. Developed with user-friendliness, advanced functionalities, and comprehensive security in mind, CompletePBX is designed to meet the diverse needs of modern communication. You can access a fully functional evaluation of the CompletePBX 5 virtual machine at no cost! Additionally, free online certification training for CompletePBX is offered, providing a foundational understanding of the system that can significantly streamline your learning process. With our assistance, you can expedite your journey to mastering this powerful phone system.

Thirdlane is a comprehensive multi-tenant communication platform that combines VoIP calling, unified messaging, video conferencing into powerful solution. Designed for both enterprises, MSPs, telecom providers and resellers, it offers unmatched flexibility, cost-effectiveness, and scalability. Highlights: - PBX with advanced call routing, voicemail, and conferencing. - IVR, call queues, recording, and analytics. - CRM integrations with Click-to-Call and caller ID screen pops. - Thirdlane Connect desktop, web, and mobile apps for collaboration. - Flexible choice: cloud-hosted or on-premises. - White-label Who it’s for: - MSPs and UCaaS providers launching branded communications. - Enterprises seeking a customizable phone system with CRM integration. - Telecom providers and resellers looking for modern PBX offerings. Thirdlane delivers a complete, future-proof phone system for organizations that want to unify communications and scale without compromise.

The Taridium IPBX represents a cutting-edge VoIP PBX solution that replaces traditional hardware systems with a software-based alternative. It functions effortlessly on standard server setups or through cloud services while avoiding the burden of additional licensing fees. An End User Panel is available, allowing users to manage their extensions effectively, including actions like handling calls through an operator console, checking voicemail, configuring call forwarding, and setting up spontaneous phone conferences, among other functionalities. Users can tailor their extension settings to forward calls during specified times by linking their iCalendar feed or by selecting preferred caller IDs from a provided list. Additionally, each user receives a meet-me style conference room, which helps in eliminating scheduling issues and allows for real-time management of conference calls, with handy options to mute or remove participants as necessary. Inviting additional participants is made easy; one simple click prompts the system to dial them directly, sidestepping the complexities of tedious login steps. This innovative system transforms the user experience by providing intuitive controls and unmatched flexibility, ultimately boosting communication efficiency and collaboration. Moreover, its ability to integrate with existing tools and workflows further enhances its appeal to businesses looking for seamless connectivity.

Dial-Office is a flexible and all-encompassing IP PBX software designed for small and medium-sized enterprises, as well as start-ups, that wish to harness the benefits of VoIP technology. By employing the SIP standard architecture, this IP PBX solution provides an array of advanced telephony features that were previously available only in costly traditional circuit-switched corporate phone systems. This reliable and cost-effective unified communications tool ensures that small to mid-sized businesses can operate effectively over a well-established Local Area Network. Recognized for its excellence, this VoIP solution offers outstanding voice clarity along with various business management tools that aim to enhance productivity, lower communication costs, and elevate customer service standards. Moreover, the platform includes features tailored to the specific demands of contemporary businesses, making it a practical option for improving communication efficiency and overall collaboration. Ultimately, Dial-Office empowers organizations to adapt and thrive in an increasingly digital world.

BillCall is a comprehensive telecom billing solution that supports both prepaid and postpaid roaming, along with voice and SMS interconnect services, rate and policy management, as well as managing deals and disputes, and order processing. Additionally, it provides a full suite of reconciliation and settlement functionalities for mediation, termination, roaming, transit traffic, and origination. This system is designed to improve current infrastructure and seamlessly integrate with numerous external software applications, including accounting systems, CRM tools, and telecommunication switches. Furthermore, BillCall is equipped to adapt to the evolving needs of telecom operators, ensuring they remain competitive in a dynamic market.

Vodia is a PBX solution that operates on both Windows and Linux platforms, delivering a comprehensive suite of advanced functionalities typically associated with robust PBX systems, such as call queues, auto attendants, and call forwarding capabilities. Additionally, Vodia stands out as a leader in Multi-Tenancy, allowing users to deploy it on a dedicated virtual machine in the cloud or on-site while managing numerous fully qualified domain names for different clients. You can experience Vodia by signing up for a free demo license, valid for 60 days, to explore its features firsthand. This trial is an excellent opportunity to see how Vodia can cater to your specific telecommunication needs.

Established in January 2015, sipXcom was created as a result of a split from the sipXecs project, driven by the eZuce, Inc. development team. The challenges faced by the sipXecs and SIPfoundry communities in broadening their developer pool due to restrictive contributor agreements led to the inception of sipXcom, which embraces an open contribution model. The sipXcom source code is distributed under the copyleft-friendly AGLP v3 (Affero General Public License), which offers developers increased flexibility. Currently, SWARM is the project name for the next version of sipXcom and sipXecs, with a production release anticipated in early 2017. This forthcoming platform will be built on a microservices architecture, aimed at improving scalability, reliability, and configurability compared to the existing sipX offerings. SWARM is designed to function seamlessly across various computing environments, such as dedicated, virtual, or cloud servers, while also supporting hybrid configurations that integrate on-premise and cloud solutions. With these innovative features, sipXcom aspires to enhance user experience and operational capabilities in the dynamic realm of communication technology. As the development progresses, the team is committed to addressing the evolving needs of users in this space.

Issabel is a notable open-source unified communications platform that combines a variety of IP communication services into a single, cohesive system, successfully addressing the shortcomings of conventional PBX systems by providing an array of functionalities such as call center operations, email integration, management reporting, fax services, video conferencing, audio recordings, and CRM connections. The call center feature is designed to be both powerful and flexible, ensuring that users can efficiently automate and manage their contact centers while skillfully managing both inbound and outbound calling campaigns through an intuitive agent interface and call management tools. Moreover, Issabel offers simplified PBX administration for service providers, enabling them to oversee multiple clients from one instance, thereby reducing the need for separate servers for each client. In addition to its core features, Issabel supports various add-ons that integrate smoothly with the platform, further enhancing telephony interactions and improving customer service by automating support tasks. This adaptability and comprehensive feature set make Issabel an excellent option for organizations aiming to optimize their communication processes while ensuring flexibility in managing their needs. As a result, businesses can achieve greater operational efficiency and improved client engagement through the use of this versatile platform.

HoduPBX stands out as an exceptional choice for enhancing business communication. Its distinctive user interface, combined with an optimal user experience, ensures that businesses can interact efficiently with their clients and stakeholders. This software excels in providing seamless connectivity, whether in a traditional office setting or remote locations, thereby facilitating smooth communication. Additionally, HoduPBX enhances agents' productivity through its multitasking features and streamlined operations. As a robust IP PBX solution, it can handle over 1000 concurrent calls and accommodates up to 10,000 extensions, making it an ideal option for diverse business needs. The HoduPBX IP PBX Software is equipped with a range of advanced functionalities, including Add-on Modules, a Reseller Module, SIP Trunking, Queue Pro, Voice Transcription, and Webphone capabilities. Moreover, it supports white label options across various platforms such as iOS, Android, Chrome Extension, Windows, Mac, Linux, and the ZohoPhonebridge Plugin, ensuring versatility and adaptability for all types of users. This comprehensive feature set positions HoduPBX as a leading solution in the realm of business communication technologies.

Our extensive white-label VoIP solution, designed specifically for service providers, offers seamless integration with third-party systems to enable off-net calling and SMS services. We provide the essential tools that boost efficiency, ensuring uninterrupted and smooth business communications. Instead of depending on cumbersome equipment typically found in telecom closets, our VoIP technology leverages your Internet connection for optimal usage. All staff members can take advantage of a shared virtual PBX, which unifies communication under one cohesive phone system. The VoipSwitch platform is meticulously developed for service providers aiming to introduce innovative applications and services, while its versatile capabilities ensure high-definition audio and video communications. Additionally, users can chat and initiate group discussions with up to 100 participants, promoting teamwork and collaboration effectively. To make calls to non-VoipSwitch RCS landlines or mobile numbers, you can easily purchase one of our minute plans. Within the VoipSwitch RCS ecosystem, your phone number acts as your unique identifier, allowing for automatic recognition of contacts who are already users, which simplifies the connection process. You can enjoy complimentary audio and video calls with other VoipSwitch RCS users, further enhancing your communication experience. Thus, we are revolutionizing connectivity and collaboration for businesses in today’s dynamic environment, ensuring they remain ahead in their communication capabilities.

InstaVoice is transforming the voice communication landscape with its cutting-edge mobile services offered to carriers across 46 countries worldwide. Developed through a unique fusion of creativity and expertise by the original pioneers of Voice SMS, the InstaVoice app is designed to provide a reliable and powerful voicemail experience for users everywhere. Enjoy the ease of accessing, responding to, and managing an unlimited number of voicemails through a user-friendly chat-like interface. Say farewell to annoying ‘Your-voicemail-is-full’ alerts, as unlimited storage allows you to access all your past voicemails, no matter if you switch carriers. Moreover, you will appreciate the free Missed Call Alerts linked to each of your numbers, even when your phone is turned off or out of reach. All your missed calls will be systematically organized in the “Missed Calls” section, enabling you to return calls or respond effortlessly with a voice or text message within the app. With InstaVoice, maintaining your connections and managing your communications has reached unparalleled levels of convenience. Embrace this innovative service and enjoy seamless communication like never before.

R.E. Communications specializes in an IP-PBX system powered by Asterisk that is compatible with standard SIP phones and offers an extensive array of features for settings ranging from a single phone to as many as 4000. These systems also support SIP trunking, enabling users to replicate the capabilities of a hosted solution while achieving considerable cost reductions per seat. For example, a configuration intended for 10 phones can function effectively with just 4 to 6 SIP trunks, which leads to significant savings compared to the costs associated with a hosted solution for an equivalent number of seats. This adaptability and efficiency make R.E. Communications’ solutions highly attractive to businesses seeking to enhance their communication frameworks while optimizing their budgets. Ultimately, such cost-effective and scalable options empower organizations to tailor their communication needs to their specific circumstances.

MikoPBX provides a robust phone system renowned for its high efficiency, remarkable reliability, and a wide array of features. Users can choose to implement it on-site via hardware setup or utilize it on various cloud services, including Google Cloud, Amazon Web Services, and Microsoft Azure. A standout aspect of MikoPBX is that it is offered as a free telephone solution, delivering all vital business functions without any associated costs. The user-friendly web interface is packed with functionalities such as call recording, conference calling, voicemail, call transfers, call pickup, call queues, an interactive voice response (IVR) setup, and customizable rules for inbound and outbound calls. Moreover, users can manage holiday schedules and set preferences for night-time calls effortlessly. The system's ability to incorporate globally developed plugins and modules further enhances its adaptability to meet specific business needs. A significant advantage of MikoPBX is its lack of user licensing fees, making it an affordable option for companies aiming to grow without the burden of additional costs for new personnel or call recording features. This capacity for scalable growth ensures that organizations can enhance their communication capabilities while keeping their infrastructure strong. Ultimately, MikoPBX stands out as an exemplary solution for businesses seeking both efficiency and cost-effectiveness in their communication systems.

thinQ offers an enhanced solution for powering voice and SMS/MMS messaging applications in the cloud. With our streamlined thinQ io portal and user-friendly APIs, you can get started swiftly and efficiently. We simplify the integration of your 40 top-tier carriers with platforms like Twilio, Nexmo, Plivo, and SignalWire, in addition to enterprise contact centers and telecom VoIP service providers. Our comprehensive APIs enable you to effortlessly manage both inbound and outbound voice calls, text messaging, and various other functionalities within your current technology setup. Additionally, you can connect with current SIP providers to enhance call quality and delivery. The fully integrated thinQ.io Web platform facilitates the purchasing and porting of phone numbers, the creation of route profiles and trunks, and the activation of text messaging, along with other essential features. By incorporating call-specific analytics, SIP ladders, and our award-winning support team, you can discover a superior communication method that helps mitigate issues, reroute around outages, and establish custom routes and blocks tailored to your needs. This level of flexibility and support distinguishes thinQ as a leader in cloud communications.

NextGenSwitch is a programmable API-driven communication platform designed to revolutionize voice and message communication solutions for businesses of all sizes. This versatile platform integrates a comprehensive suite of communication tools, including PBX, Call Center, and Call Campaign solutions, into a unified and scalable platform. Designed to streamline operations and enhance customer engagement, NextGenSwitch offers seamless CRM integration for better data management and workflow efficiency. Additionally, it includes a Click-to-Call feature that simplifies outbound calling by allowing users to initiate calls directly from their CRM, website, or other digital platforms. PBX Features Call Routing & IVR: Intelligent call routing and customizable Interactive Voice Response (IVR) systems. Voicemail & Call Recording: Ensure no communication is missed with voicemail and secure call recording capabilities. Extensions & Hotdesk: Support for multiple extensions, hotdesks. Integration: Seamless integration with CRM and other third-party tools. Call Center Features Automatic Call Distribution (ACD): Efficiently route calls to the best available agents. Real-Time Monitoring & Analytics: Track agent performance and call statistics live. Voice API Feature Twilio alike API with xml markup language to send and recieve calls vial REST API.

The Huawei CloudFabric solution aims to establish a streamlined, intelligent, and extensive network for cloud data centers that telecommunications providers can deploy in diverse settings, such as private and public clouds, telecom environments, and hosting services, thereby supporting the digital evolution of enterprise clients. At the heart of this innovative solution lies an open architecture tailored for cloud services, which includes the cutting-edge CloudEngine series data center switches and the Agile Controller. This comprehensive series features the acclaimed CloudEngine 12800 series, recognized as the most advanced core switches available, alongside the durable CloudEngine 8800, 7800, 6800, and 5800 fixed switches, in addition to the CloudEngine 1800V virtual switches. Furthermore, the CloudFabric solution integrates essential components like transport, routing, security, and network management tools, leveraging Software-Defined Networking (SDN) technology to construct state-of-the-art cloud data center networks. This all-encompassing strategy not only boosts operational efficiency but also empowers carriers to enhance their service offerings to enterprise clients in a rapidly digitizing landscape, ultimately driving better customer satisfaction and fostering innovation.

Our telecom expense management and connectivity solutions enable businesses to connect and manage all devices and networks through a unified platform. The process of handling telecom costs can often feel overwhelming, but we alleviate that burden without imposing any financial strain on your organization. By adopting our managed mobility services, we optimize operations, cut costs, and free up your internal teams to focus on key strategic goals. We provide an integrated approach by merging both Mobile Network Operators (MNOs) and Mobile Virtual Network Operators (MVNOs), offering reliable single and multi-carrier solutions that ensure maximum uptime. With Advantix SmartSIM, you can effortlessly connect SIM-compatible devices to a variety of cellular networks thanks to our cutting-edge carrier switching technology. This innovation removes the hassle of choosing a specific carrier and alleviates concerns about signal quality based on location. Furthermore, our Telecom Lifecycle Management (TLM) platform integrates smoothly with carrier systems using their APIs, promoting a cohesive telecom management experience. This deep integration not only allows for immediate adjustments but also provides valuable insights that significantly boost operational efficiency and agility. In essence, our services are designed to simplify telecom management while enhancing your organization’s overall performance.