Alternatives to SIP Server

3CX is an open-standards, software-based IP PBX that provides complete Unified Communications right out of the box. 3CX is suitable for all industries and sizes. It can provide everything you need, from mobility and status to advanced call center features and more, at a fraction the cost. 3CX is ideal for remote companies and includes mobile apps, video conference, and website live chat. The installation, management, and maintenance of the PBX are so simple that you can easily manage it yourself, on your own appliance, server, or on a cloud-based platform. You can also choose to have it hosted on 3CX for an easy solution.

Our proprietary technology has been providing telecom services since more than 20 years. We are experts in Hosted IP-PBX and SIP Trunks, Call Center Software, Wholesale, Data Services, etc. Our exceptional service, rates, and products meet the FCC standards for STIR/SHAKEN network compliance. The XCast Unified Communications is Scalable and International with unlimited, free, planet-wide, extension-to-extension, all with a web-based Administrator Control Panel (ACP) to help you manage all your features. Promotional plans: Hosted PBX Basic - $9.95/month Sip Trunking Basic - $4.95/month

PBXware is the first and most established IP PBX Professional Open Standards Turnkey Telephony Platform. PBXware has been providing flexible, reliable and scalable Next Generation Communication Systems (NGCS) and VoIP solutions to small and medium-sized businesses (SMBs), enterprises and Internet Telephony Service Providers, (ITSPs), call centers and governments around the world since 2004. This is done by combining the best of the latest technologies. Bicom Systems softswitch is available in the Business, Call Center, and Multi-Tenant Editions. Each edition supports specific features that maximize performance, reliability, expandability, and expandability.

In today's fast-paced environment, the ability for professionals to effectively communicate with colleagues and clients is more crucial than ever. Our telephony solutions are flexible enough to cater to diverse requirements, whether you are in a traditional office setting or working remotely. From fixed and mobile telephony to trunking and contact centers, our business telephony offerings are increasingly versatile. Sewan's solutions are customizable to meet a wide range of demands. The advanced voice technologies integrated into our products ensure exceptional call clarity and unwavering reliability. We understand that business needs can evolve swiftly, allowing for quick adjustments such as adding or removing lines and importing contacts in mere moments. No matter the type of business telephony you opt for, our solutions aim to be as economical as possible, eliminating unnecessary features. Additionally, our offerings are tailored for the global market, ensuring that each package can be modified according to specific requirements. This comprehensive approach helps businesses of all sizes maintain efficient communication channels.

TekSIP for Windows is a SIP Registrar as well as a SIP Proxy. TekSIP supports UDP (IPv4 & IPv6), TCP (IPv4 & IPv6), TLS, and WebSocket transports. Only commercial editions support Secure WebSocket and TLS. TekSIP has been tested on Microsoft Windows Vista and Windows 7/8/10/11 servers. TekSIP is a signaling service for WebRTC based SIP-based phones. TekSIP is compliant with RFC 3261 and RFC 3263. It also complies to RFC 3581, RFC 3891, RFC 3311 and RFC 3481. It supports NAT traversal, ENUM. Please refer to Readme for technical details on NAT traversal. You can choose the IP address that will be listened to and alternative SIP endpoints when making outgoing calls. You can also log the session details in a log and monitor registrations and active sessions in real-time. Windows Performance Monitor (Perfmon.exe) allows you to monitor the number of active sessions and registrations. TekSIP redirects calls to an alternate endpoint if the called endpoint is not available (Offline, busy ...). GUI can be used to terminate an active SIP session.

AbacusCloud is a robust platform designed for VoIP, unified communications, call centers, voice recording, and SIP trunking, specifically catering to medium and large enterprises. It offers impressive scalability, reliability, and geographic redundancy to meet the evolving demands of businesses. Collaborating with top-tier service providers, data centers, Tier 1 carriers, and high-quality SIP endpoints, AbacusCloud stands out in the telecommunications landscape. The platform supports an extensive range of desk phones, softphones, and mobile devices, providing users with advanced voice, video, and wireless capabilities that can be accessed from anywhere, at any time, on any device. AbacusCloud simplifies the VoIP migration process significantly, ensuring seamless connectivity for multiple offices and mobile employees in a cost-effective and incremental manner. In addition, AbacusGroup enhances the experience with a comprehensive suite of omnichannel services, offering a cohesive user experience across different devices, making it perfectly suited for today’s workforce that prioritizes cloud and mobile solutions. This adaptability ensures that organizations can respond swiftly to changing communication needs, thereby enhancing overall productivity.

Sangoma Wholesale Carrier Services, previously known as VI Communication Services, has been tailored to enable Internet telephony service providers (ITSPs) and managed service providers (MSPs) to deliver the high-quality services their clients desire seamlessly. Companies can utilize our communication APIs to integrate voice and messaging capabilities into their applications, facilitating effortless interaction with their customers. As an ITSP, MSP, or value-added reseller (VAR), you can enhance the offerings for your current retail SIP trunking clientele by providing additional phone numbers and VoIP services that may not be accessible within your current network, thereby generating new revenue opportunities. Alternatively, if you're just beginning your journey, you have the option to launch a new retail SIP trunking service and incorporate the features you wish to provide to your clients, including messaging, conference bridging, and a variety of other functionalities to enhance their experience. This flexibility allows businesses to stay competitive in a rapidly evolving market.

Voxtelesys has been designated a preferred carrier for several PBX and Dialer vendors. With an in-house team of software and hardware engineers, we support the products we sell. Every day, we are custom-coding new applications to improve connectivity to Voxtelesys' services. Voxtelesys will make your decision easy once you see how they support customers. We answer the phones and take care of business. Voxtelesys offers scalable IP Communications services to businesses and contact centers of all sizes. Our High Capacity SIP Trunk with Next Generation Routing supports contact centers with 4 to 4,000 lines. Our carrier-grade SIP termination service is fully redundant with 99.999% (five nine) reliability and offers unparalleled integration power with your network infrastructure. Our SIP service is based on the G711/ULAW/ALAW codecs and delivers the highest call quality.

Unified communications and collaboration for enterprises can seamlessly connect individuals anytime, anywhere, and across various devices through Cisco's comprehensive collaboration framework designed for voice and video interactions, messaging, and mobility solutions. The Cisco Unified Communications Manager (Unified CM) delivers dependable, secure, scalable, and manageable call control along with session management capabilities. By streamlining your communications infrastructure, Cisco Unified Communications Manager empowers your workforce and teams to engage effortlessly. This solution encompasses a range of features including IP telephony, high-definition video conferencing, unified messaging, as well as Instant Messaging and Presence capabilities. Redefine your work environments to draw in and retain top talent from any location, equipping them with the Cisco Unified Communications Manager and the essential tools they need to achieve success. With robust features tailored to accommodate mobile and remote employees, whether you are a local family-owned business or a global enterprise, selecting a solution that adapts to your evolving needs is essential for ongoing growth and innovation. Ultimately, implementing Cisco’s unified communications can significantly enhance collaboration and productivity across your organization.

SmartSIP® is a software solution designed for on-premise use that offers DTMF suppression and masking, allowing contact centers to securely collect sensitive cardholder information during live calls. Customers can input their confidential data using their touch-tone phone while remaining connected to a contact center representative. Through DTMF suppression and masking, the tones generated by the customer are substituted with flat tones that cannot be reverted to their original forms. The agent's interface displays a masked version of the customer's input. By effectively removing the need for verbal communication of sensitive details, SmartSIP ensures that full call recordings and comprehensive screen captures are maintained. Acting as a SIP proxy between the Session Border Controller (SBC or Media Gateway) and the voice platform, SmartSIP processes the DTMF data without interfering with the audio connection. This innovative tool is specifically designed for VoIP telephony within contact centers, making it an essential asset for maintaining data security during customer interactions. Moreover, its implementation can significantly enhance compliance with various regulatory standards related to data protection.

Voice communication plays a crucial role in the success of any omnichannel business and communications strategy, yet it is frequently overlooked. Brightlink Voice serves as a comprehensive solution for all voice-related needs, delivering dependable global connectivity that facilitates calls for both enterprises and communication providers through a top-tier IP-based network. Our Enterprise Network Services feature straightforward pricing, granting you access to all essential voice communication functionalities within a single, unified offering. Additionally, our CPaaS APIs enable the creation of tailored voice integrations that seamlessly fit into your business applications and processes. Beyond our carrier services, we offer a solution specifically designed to enhance customer engagement for both enterprise and SMB clients. Brightlink Network Services consolidates vital voice communication capabilities into one all-inclusive package, ensuring that you have everything you need at your fingertips. This bundled access not only simplifies your voice communications but also empowers your business to thrive in an increasingly interconnected world.

Phonology's Enterprise VoIP solutions are tailored for small to medium-sized businesses and larger organizations, enabling effective connectivity, communication, and collaboration across multiple channels while delivering exceptional quality and flexibility in VoIP services. Their robust Hosted PBX offering for cloud phone solutions allows users to leverage the advantages of a large enterprise PBX system through virtual extensions, seamlessly integrating with preferred third-party applications to enhance telecommunication capabilities. Unified Communications consolidates various communication methods into a single platform, simplifying the collaboration process for users. If you're concerned about handling physical and wired PRI, BRI, or PSTN lines, consider Phonology's SIP Trunks, which allow for efficient and stress-free management of your business communications. This comprehensive approach ensures that businesses can focus more on their core objectives while enjoying streamlined communication solutions.

FusionPBX serves as a versatile solution for both single and domain-based multi-tenant PBX setups, functioning as a carrier-grade switch, call center system, fax server, VoIP server, voicemail server, conference server, and voice application server, among other roles. Meanwhile, FreeSWITCH™ stands out as a highly adaptable and scalable communication platform that operates across multiple threads and platforms. It boasts an impressive array of features, including unlimited extensions, voicemail-to-email, music on hold, call parking, call center capabilities, call queues, and phone provisioning, catering to businesses of all sizes. By delivering corporate-level phone system functionalities, it meets the diverse needs of small, medium, and large enterprises alike. Significant efforts have been invested in enhancing security, resolving issues, and refining the codebase. Furthermore, ongoing advancements are being made in preparation for an upcoming release, with notable progress achieved; the master branch has outpaced the current version in several respects. As the team works meticulously to finalize the essential updates, anticipation builds for the enhancements that will accompany this release. The commitment to continuous improvement ensures that the platform will remain competitive and relevant in the rapidly evolving telecommunication landscape.

VQmon is an embedded library for real-time media streams like VoIP, IP Videoconferencing, and Video Streaming. VQmon provides real-time analysis of IP-based media streams, providing detailed analytics and QoE scores. Library APIs and standardized payloads for RFC 635 (SIP PUBLISH), and RFC 3611 are both available to get metrics. VQmon has been integrated into over 500 million IP phones, media gateways and test tools. VQmon is highly efficient, scalable, and uses a fraction of a microprocessor to monitor a single voice stream or video stream. It also processes large volumes of streams in multicore servers at more than 2 million packets per core. VQmon can be used on a variety of operating systems and processor platforms.

DID Global offers high-quality, efficient premium IP telephony for business and personal communication that allows you to scale quickly and save advertising budgets while increasing sales. We offer our clients attractive rates. Our working principles are based on the common phrase "price equals quality". You can save 60% on your advertising budget if you use local operators. If you call a prestigious local number, your clients will be more trusting. We only use high-quality, stable communication channels. You can switch to a different provider immediately if you lose your connection thanks to channel redundancy. This allows you to communicate with your customers while your business runs smoothly. We offer the best conditions to our customers in over 150 countries.

Issabel stands out as an open-source unified communications platform that merges various IP communication services into one cohesive system, effectively overcoming the limitations of traditional PBX systems by offering features like call center functionalities, email integration, management reporting, fax capabilities, video conferencing, audio recording, and CRM integrations. Its call center module is not only robust and flexible but also user-friendly, enabling efficient automation and management of contact centers while adeptly handling both inbound and outbound call campaigns through a straightforward agent interface and call management tools. Additionally, Issabel simplifies the administration of PBX solutions for service providers, allowing them to manage multiple clients from a single instance, thus alleviating the need for individual servers for each client. To further expand its capabilities, Issabel accommodates various add-ons that are designed to integrate seamlessly with the platform, enhancing features that interact with telephony and improve customer service while automating support-related actions. This adaptability makes Issabel a highly effective choice for organizations looking to streamline their communication processes.

VOS3000 is tailored for operations at the carrier level, offering a comprehensive suite that includes Account Management, Exchange Rate Management, Rate Management, Package Management, Cards Management, Gateway Management, Phone Management, Softswitch Management, IVR Management, System Management, User Management, Data Query, and a Web Self-Service System. Additionally, it features supplementary modules such as the Global Card Business System, which can support up to 10 million cards, and the Extreme Media Proxy, enabling you to create a robust, dependable, and high-performance operating environment. This softswitch is compatible with SIP, H323, and SIP-H323 protocols, boasting an impressive capacity of up to 10,000 concurrent calls. By enhancing processing performance and reducing CPU usage, it ensures efficient operations. Furthermore, the H323 Add Signaling option can be selected for both the Caller (Calling Party Number / Source Address) and Callee (Called Party Number / Destination), while also incorporating a process number that functions as a local phone capability for gateway mapping, thereby increasing overall functionality and versatility.

The on-premises business phone solution that provides all the essential tools for your organization to enhance customer interactions. At the core of Mitel's premier on-site communication offerings lies MiVoice Business, which includes MiCollab, MiTeam Meetings, and MiContact Center Business, allowing employees to reap the benefits of unified, scalable, and mobile communication seamlessly. This solution equips your business with everything necessary to elevate communication standards through effective mobility, team collaboration, video conferencing, and outstanding customer experience management. MiVoice Business is tailored to accommodate businesses ranging from 5 to 130,000 users, whether operating from a single location or across multiple sites worldwide. Furthermore, the software installed on Mitel Collaboration Service Appliances can run on standard x-86 servers within a virtualized setup, streamlining operations by reducing the need for multiple appliances—an especially valuable feature for clients opting for a centralized unified communications deployment. With MiVoice Business, organizations can ensure that their communication infrastructure is both flexible and robust, fostering growth and innovation.

Upon logging into the portal, you can view your data in a visual format that makes it easy to analyze. Regardless of size or age, all information is automatically organized to generate instant charts, reports, statistics, and evaluations. The platform leverages the latest technology, including SIP Trunks in the backend, alongside multiple systems to guarantee a seamless experience without call drops. Its impressive scalability allows for flexibility and offers substantial bandwidth to handle any unexpected surges in call volume. You can enjoy unlimited parallel calls with no channel restrictions, and every plan includes a comprehensive set of features. Our transparent pay-as-you-go billing model empowers you to select your usage and manage costs effectively. Take command of your company's telephony experience with the ability to record all inbound and outbound calls for compliance, auditing, training, or enhancing customer interactions. Call records are readily accessible at any time through the dashboard, while prompt alerts can be sent via SMS or email, or you can view notifications directly within the dashboard. Additionally, this user-friendly interface ensures that you can stay informed and responsive to your communication needs without any hassle.

GAO Research Inc. serves as a comprehensive software solution provider specializing in embedded modem software, fax modem solutions, speech codecs, and telephony software tailored for electronics, telecommunications, and semiconductors in embedded engineering. The company delivers an array of services including VoIP and FoIP solutions, alongside V.92 and V.90 softmodems with fallback capabilities, Super G3 and G3 fax software, as well as advanced speech and video codecs, all compatible with the widest variety of DSPs and microprocessors in the industry. Beyond offering high-quality software, GAO is committed to providing exceptional technical support, deep platform knowledge, and cutting-edge testing facilities. Furthermore, GAO equips system designers and engineers with a comprehensive VoIP packaged solution that encompasses modules for both fax and data over IP, ultimately streamlining development timelines and expediting market entry. A typical IP Telephony System consists of essential components such as User Interface, Voice Processing, Network Protocols, Network Management, and System Services, all of which contribute to efficient system performance. This holistic approach ensures that clients receive not only innovative software solutions but also the necessary tools to enhance their operational capabilities.

Our carrier-grade Session Border Controller (SBC) and cloud telephony solutions are designed for high performance and scalability, catering to the needs of communication service providers and SMS aggregators with our mobile OTT solution and A2P SMS platform. We also provide secure communication platforms that enhance real-time business collaboration for enterprises. Our comprehensive white-label offerings enable you to deliver cloud telephony services under your own brand identity. With a targeted focus on the IP-based communication sector, we offer a diverse range of products that encompass everything from backbone infrastructure to supporting middleware components. We simplify the process for termination clients by converting their rate plan formats into a native format efficiently. Our system allows for effortless uploading of VoIP rate plans, which are then transformed into the appropriate templates automatically. Additionally, all templates undergo automatic conversion to ensure they align with the native rate plan format seamlessly, and our technology is designed to prevent call drops by ensuring that user account balances do not fall into the negative. This suite of features positions us as a leader in the communication technology space, providing robust solutions tailored to modern business needs.

Welcome to TENIOS, the cloud communications provider of the Apifonica Group. TENIOS is German-based and specializes in cutting-edge AI voicebots and telephony solutions for businesses. Their mission in short words: Bringing Conversational AI to the world. Their passion for automation unites a team of experts in Cloud Technology, Telephony, and AI to help businesses automate communication and related business workflows. TENIOS Voicebots handle outbound and inbound calls, call back leads, pre-qualifiy them, instantly update CRM data, and create reports to scale customer communications. The comprehensive telecom-platform offers services such as virtual phone numbers, intelligent call routing, interactive voice response (IVR) systems, SMS, RCS, and a robust Voice API for seamless integration of voice applications. With over two decades of experience and hosting in Germany, TENIOS ensures reliable and scalable communication solutions tailored to meet diverse business needs.

Alohaa serves as a comprehensive cloud communication platform, delivering a range of omni-channel products including virtual phone numbers, IVR systems, cloud telephony, WhatsApp API integration, bulk SMS services, and voice broadcasting aimed at businesses. Thanks to its cloud-based framework, Alohaa enables businesses to achieve both flexibility and scalability in response to their evolving needs. The platform optimizes call routing, automates customer engagements, and significantly improves communication efficiency. Catering to a variety of industries, Alohaa provides customized solutions tailored for customer service, sales, and support functions, ensuring that each sector's unique requirements are met. Its intuitive interface, combined with its advanced capabilities, positions Alohaa as a top choice for companies in search of dependable cloud communication solutions, ultimately facilitating enhanced connectivity and operational success.

thinQ provides a "better" way to power voice and SMS/MMS messaging applications in the cloud. Get up and running quickly with our integrated thinQ io portal and APIs. We make it easy to bring your 40 high-quality carriers to Twilio and Nexmo, Plivo and SignalWire, as well as enterprise contact centers and telecom VoIP providers. Our APIs allow you to quickly add and control inbound and outbound voice, texting, and other features to your existing tech stack. You can also connect to existing SIP providers to improve call delivery and quality. Our fully integrated thinQ.io Web platform makes it easy to purchase and port phone numbers, create route profiles, trunks, enable text messaging, as well as create route profiles, trunks, and other functions. You'll find a "better" way to communicate if you add call-specific analytics, SIP ladders, and our five-star support team to mitigate, route around outages, create customized routes and blocks, etc.

Establish a confidential environment for your team's interactions by utilizing Zangi for Business applications, which offer secure messaging, calling, file sharing, and conferencing tailored for professional use. These applications effectively distinguish between your work and personal communications, ensuring that your conversations remain private. With a foundation of Military-Grade Encryption and a strict ZERO data collection policy, Zangi guarantees that your information is never stored on any external servers or cloud services. Experience seamless connectivity coupled with heightened privacy as you take command of your corporate communications. This innovative platform allows you to create a personalized communication app for both mobile and desktop, eliminating the need to rely on third-party solutions. Take control of your data and enhance your business interactions by developing a custom social networking platform that reflects your brand identity, complete with unique features tailored to your needs. Furthermore, you can expand your communication capabilities by integrating your existing phone system with this secure business communication app, paving the way for enhanced collaboration and efficiency.

Unified Office, Inc. stands out as a premier supplier of SDN-based Hybrid Cloud solutions, seamlessly integrating on-premise, private cloud, and public cloud technologies to offer managed Voice-over-IP and Unified Communications services tailored for Small and Medium-sized Businesses (SMBs). Our Total Connect Now℠ service architecture is meticulously engineered from the ground up to ensure exceptional quality of experience and high availability, utilizing cutting-edge extensible business VoIP communications technology alongside cloud-based infrastructure to boost productivity for SMB workforces. This innovative design stems from our team's extensive experience accumulated over decades in both the telecommunications service provider sector and the large enterprise data networking field. As a distinguished business VoIP provider, we have not only crafted protocols and developed scalable products but have also integrated, deployed, and supported carrier-grade equipment across many of the largest converged networks globally. Central to our mission are the principles of quality, reliability, innovation, and customer support, which guide our operations and commitment to excellence. We continuously strive to adapt to the evolving needs of our clients, ensuring they stay ahead in a competitive landscape.

Go Integrator Cara serves as a comprehensive communications client tailored for the Cisco BroadWorks hosted telephony system. With a host of features designed for quick access to vital information, it seamlessly integrates with contact-focused business systems and CRM applications, boasting a standardized set of integration tools compatible with over 300 widely used applications across various sectors. This functionality leads to substantial productivity improvements, such as instantly identifying incoming callers and displaying their details, or effortlessly dialing their numbers with a simple click. By utilizing Go Integrator Cara, customers can save both valuable time and resources. The application includes a built-in SIP Softphone that can be set up and activated through a standard BroadWorks device profile. Users can easily add records to an integrated CRM directly from the client’s call history or the active call preview window, and it automatically or manually generates an activity record for each received call within the CRM. Additionally, this integration enhances overall communication efficiency by streamlining workflows and minimizing the time spent on administrative tasks.

From emerging startups to major global corporations, we possess the in-house knowledge and proficiency to create a solution that perfectly aligns with your specific needs. Whether you need a robust SIP trunking solution for your existing phone system or are in search of a cloud-based managed hosted PBX solution, we offer a variety of flexible products that can be customized to fit your requirements. At Orbtalk, we recognize that to thrive in today’s competitive landscape, your business must be capable of swiftly adapting to sudden spikes in demand. This understanding is why our services are crafted with flexibility and responsiveness as key priorities. Additionally, with our servers strategically located in various data centers worldwide, you can trust that you are receiving a service that is not only well-architected but also boasts exceptional high availability and reliability. Our commitment to innovation ensures that your communication solutions will evolve alongside your business.

If you seek top-notch voice, messaging, and additional features, it’s essential to select an API provider that operates its own nationwide network. By cutting out intermediaries, you can reduce expenses, enhance your control, and create the application that your users truly desire. While you may already have experience in this field, a reliable partner is crucial to help you achieve the growth you aspire to. Whether you opt to utilize your own carrier or develop a completely new solution, Bandwidth equips you with the necessary tools to unlock your full potential. Serving as a versatile communications platform, Bandwidth offers extensive flexibility for your needs. You require seamless integrations, robust enterprise-level APIs, and intelligent tools to effectively manage your communication system. We prioritize openness in the areas that matter most to you, providing you with options and control over your operations. Our extensive global network ensures direct connections where it counts, delivering unparalleled reliability. Ultimately, this approach fosters stronger relationships with your customers while allowing your business to scale seamlessly in any direction you choose. Additionally, Bandwidth's commitment to innovation ensures that you are always equipped with the latest advancements in communication technology.

R.E. Communications provides an IP-PBX system powered by Asterisk, compatible with all standard SIP phones and equipped with high-level features suitable for environments ranging from just 1 phone to as many as 4000. Additionally, these systems facilitate SIP trunking, allowing users to mimic the functionalities of a hosted solution while significantly reducing costs per seat. For instance, a system designed for 10 phones can efficiently operate with only 4 to 6 SIP trunks, resulting in notable savings compared to the expense of a hosted solution for the same number of seats. This versatility makes R.E. Communications’ offerings particularly appealing for businesses looking to optimize their communication infrastructure while managing expenses effectively.

SmartCallMonitor seamlessly integrates telephone system capabilities into your TAPI-compatible devices, turning your phones and computers into a cutting-edge, efficient CTI solution. By utilizing a SIP account, you can take advantage of the numerous benefits associated with VoIP through SmartCallMonitor. This powerful solution provides you with a modern telephony system. If your server features a TAPI interface and you wish to extend this functionality to additional computers, simply install the software on all connected devices. The server will handle data retrieval for calls, enabling centralized call management without requiring a TAPI interface on every individual computer. This means that a sales or customer support representative can efficiently transfer a call to a colleague identified by SmartCallMonitor as the designated account manager, using data from the database. The innovative solution facilitates communication between computers linked to a TAPI device, enhancing overall operational efficiency. Overall, SmartCallMonitor revolutionizes how telephony systems operate within a connected environment.

Our comprehensive portfolio of unified communication and collaboration (UCC), solutions will help you grow your business, increase productivity, and maximize efficiency all at once. They seamlessly integrate with existing systems and transform your user experience. Our unified communication solutions make it easy to connect, communicate, and collaborate with customers and employees worldwide via video, web, or audio conferencing. With Tata Communications' unified communication services, you can dramatically improve customer experience and loyalty. Unifying conferencing and leveraging Global SIP Trunk for IP telephony, and improving efficiency across your business will help you to improve productivity and reduce telephony costs.

SpeechSoft Inc. specializes in the creation and distribution of telephony automation software, alongside offering hardware and comprehensive solutions that seamlessly integrate with a variety of telephony systems, including Voice over Internet Protocol (VoIP), T1, Digital PBX, and Analog formats. Their CALLMaster software consolidates six robust telephony capabilities into one comprehensive package, delivering an integrated solution suitable for diverse business operations. The telephony products from SpeechSoft are designed to be scalable and built on an open, non-proprietary framework, allowing companies to craft cost-effective solutions tailored to their specific requirements and financial constraints. Established in 1987, SpeechSoft, Inc. has leveraged its extensive experience in the industry to furnish economical telephony solutions across multiple sectors. By consistently collaborating with top-tier partners such as Microsoft and Dialogic, SpeechSoft remains committed to providing innovative, state-of-the-art solutions that meet the evolving needs of its clients. With a focus on fostering long-term relationships, the company aims to adapt its offerings to future advancements in telephony technology.

Wavetel was founded in 2010 and has since grown to be a major provider of telecom services in the UK, and around the globe. Wavetel began as a wholesale operator that offered Voice, SMS, and Data services to other operators. Wavetel's B2B services were launched in 2013 by the company. These include VoIP telephony services and Connectivity, Virtual Numbers. SIP Trunks. Microsoft Teams Direct Routing. Mobile, SMS, Call Center Solutions are available to its valued clients worldwide. Our voice messaging solutions have evolved over the years to be an enterprise solution. It can integrate with third-party CRMs and business apps through our programmable APIs.

Asterisk is a versatile open-source framework designed for creating various communication applications. It transforms a standard computer into a robust communications server. Asterisk serves as the backbone for IP PBX systems, VoIP gateways, conference servers, and numerous other tailored solutions. Its users range from small enterprises to large corporations, call centers, telecom carriers, and government bodies across the globe. Available at no cost, Asterisk thrives under the sponsorship of Sangoma. Currently, over a million Asterisk-based communication systems operate in more than 170 nations. Remarkably, nearly all companies listed in the Fortune 1000 utilize Asterisk in some capacity. Most frequently implemented by system integrators and developers, Asterisk can serve as the foundation for a comprehensive business phone system or can be employed to augment an existing setup or connect disparate systems. Whether you're interested in crafting a personalized system with Asterisk or purchasing an efficient, affordable turnkey solution, explore the options available to determine the best fit for your needs. With its flexibility and wide-ranging applications, Asterisk continues to evolve, ensuring it meets the demands of modern communication.

One call can connect your entire business. Fusion Connect is your Managed Services Provider, providing business internet solutions for your SDWAN, business VoIP (UCaaS), internet accessibility, secure networking, and other requirements. Fusion Connect's Business VoIP includes Internet-based faxing and softphones. Compatible with Windows and Mac as well as other devices.

Founded in January 2015, sipXcom emerged from a fork of the sipXecs project, initiated by the eZuce, Inc. development team. The sipXecs and SIPfoundry communities were struggling to expand their developer base due to a limiting contributor agreement, prompting the creation of sipXcom, which features an open contribution model. The source code for sipXcom is licensed under the copyleft-friendly AGLP v3 (Affero General Public License), ensuring greater flexibility for developers. Currently under development, SWARM is the codename for the upcoming iteration of the sipXcom and sipXecs projects, with a production launch expected in early 2017. This new platform will adopt a microservices-based architecture, enhancing scalability, reliability, and configurability beyond what sipX currently offers. SWARM is engineered to be compatible with any computing environment, including dedicated, virtual, or cloud servers, and also allows for hybrid setups that merge on-premise and cloud solutions. With these advancements, sipXcom aims to significantly improve user experience and functionality in the ever-evolving communication landscape.

We provide a versatile solution tailored to meet your business telephony requirements. STARFACE delivers an ideal communication system, whether you prefer hardware or software options, all sourced from a single provider and proudly “Made in Germany.” The events of 2020 have highlighted the necessity of maintaining connections with both colleagues and clients, regardless of their location—be it in the office, at home, or traveling. Thanks to STARFACE UCC Clients, staying connected has never been more straightforward, with features that include business telephony, convenient video conferencing, instant messaging, and file sharing. Effective communication is the cornerstone of any successful business, and its importance has only intensified in today’s world. You can also access our telephony system seamlessly from your Android device or iPhone, allowing you to make calls using your landline number directly from your mobile phone and utilize key functionalities just as easily as you would through the main user interface. This level of flexibility ensures that you can remain productive and engaged, no matter where your work takes you.

Experience a swift return on investment through enhanced cost savings, operational efficiencies, and elevated customer service. Our Call Management System (CMS4) equips organizations with the tools to manage their telephony expenses effectively, offering in-depth performance analytics. Regardless of whether you operate a small system or a vast, globally clustered network, CMS4 supplies the essential information you need in your preferred format. The system can be deployed as an On-Premises, Cloud, or fully managed hosted service solution. CMS4 enables you to analyze traffic at each gateway or trunk group, generating grade of service reports that reveal whether your capacity aligns with demand. By leveraging this data, organizations can ensure their systems operate efficiently to accommodate present needs while also anticipating future demands. This adaptability not only optimizes resources but also positions businesses for sustained growth.

Quality Voice & Data (QVD) routes calls directly to key carriers, guaranteeing that all calls are authenticated and delivered with complete SHAKEN attestation. Their Trusted Call Completion suite merges superior caller ID verification with immediate monitoring and remedial actions to avert call blocking or spam labeling. To further enhance outbound communication, QVD offers a Branded Calling ID that showcases business names, logos, and call purposes, resulting in an impressive improvement in answer rates by more than 48%. In addition to this, QVD utilizes proprietary technology, sophisticated data models, and user-friendly dashboards to optimize call delivery processes. With a rich history spanning over three decades, numerous patents, and the processing of billions of calls, QVD's offerings encompass local caller ID numbers, elastic SIP trunking, real-time monitoring, caller ID spam remediation, and cloud-based VoIP phone systems. Their holistic strategy not only ensures effective communication but also helps businesses sustain a reputable phone presence in an increasingly competitive market. This commitment to quality service sets QVD apart as a leader in the telecommunications industry.

PBXMate is specifically designed for providers of conferencing solutions, developers and operators of VoIP technologies, as well as organizations aiming to eliminate echo and noise during their phone conversations. This software solution operates independently of hardware modifications, making it user-friendly and efficient. Featuring advanced algorithms for noise reduction and server-side echo cancellation, PBXMate actively monitors network quality, offering immediate visual notifications if call quality deteriorates. SoliCall’s PBXMate seamlessly integrates with any network utilizing the standard SIP interface. Thanks to its plug-and-play technology, users can initiate noise reduction within minutes without needing to alter existing hardware setups. PBXMate also provides real-time statistics for ongoing calls and aggregated data for all processed calls, allowing users to observe the enhancements made in call quality. This insight not only aids in appreciating the positive impact of PBXMate but also empowers users with better control over various call quality parameters. Overall, PBXMate enhances the communication experience by ensuring clearer conversations and reducing disruptive noise.

The Taridium IPBX is an advanced software-based VoIP PBX solution that effectively takes the place of conventional hardware PBX systems. It operates seamlessly on standard server infrastructure or via cloud services without necessitating additional licensing costs. Users benefit from an End User Panel that empowers them to manage various aspects of their extensions, including call operations through an operator console, accessing voicemail, setting up call forwarding, and initiating impromptu telephone conferences among other features. You can customize your extension to forward calls only during specific hours by integrating your iCalendar feed or by selecting from a list of preferred caller IDs. Each user is provided with a meet-me style conference room, eliminating scheduling conflicts and enabling live management of conference calls, including options to mute or remove participants as needed. Inviting users is straightforward, requiring just a click to have the system initiate a call to them, thus avoiding the hassle of complicated login procedures. This system redefines user experience by offering intuitive controls and flexibility that enhance communication efficiency.

Our voice platform offers a comprehensive array of products and services designed to automate various business processes and fully capitalize on the benefits of cloud telephony. Experience the capabilities of cloud telephony and access all your communication solutions from a single source. Allow your customers to reach you effortlessly while expanding your business on a global scale. YakoVoice boasts a robust cloud infrastructure distributed across multiple regions to guarantee high availability and reliability. The system is meticulously crafted and rigorously tested to ensure optimal performance. In the event of an unexpected outage in any region, our resilient cloud infrastructure is crucial for effective failover management for each toll-free number, maintaining seamless communication for your business and customers alike.

Sabienzia revolutionizes telephony by transforming it into intelligent communication, enabling businesses to make data-driven decisions based on metrics, context, and meaning, all while streamlining workflows through automation. With over ten years of experience in offering cloud telephony solutions, Sabienzia enhances customer service for established firms in the travel, automotive, and e-commerce industries around the globe, utilizing a completely GDPR-compliant infrastructure that is entirely hosted in Europe, ensuring the highest level of PCI security. In just a few minutes, you can establish a fully customizable cloud-based customer service system to effectively engage with your international audience. Our goal is to create a centralized platform that empowers your users to deliver a more personalized, meaningful, and empathetic experience to customers from any location. You can initiate the cloud contact center solution for your business in just one minute without any installations required. By leveraging the appropriate channels and metrics, you can maintain a closer connection with your customers, remote agents, and employees, fostering a more collaborative environment. This seamless integration allows for improved communication and service efficiency, ultimately leading to enhanced customer satisfaction.

RingLeader is revolutionizing communication for businesses and teams by introducing innovative technology known as "Crowdvoicing," enabling small enterprises and teams to effortlessly implement cloud-based phone services for durations as short as one day. Our SIP Trunking solution seamlessly integrates with any PBX hardware, making it ideal for larger organizations; in fact, many of our SIP clients can utilize our trunks without the need for extra hardware. As one of the rare NATIVE SIP providers globally, we believe that the best solutions are often the most straightforward ones. RingLeader empowers your business to communicate more intelligently, quickly, and efficiently—whether from a traditional office or a home setting—offering adaptable solutions tailored to the needs of small, medium, or large teams. Our services are backed by dedicated support from real people who answer calls personally. Ultimately, RingLeader provides the quickest and easiest method for organizations to enhance their communication strategies, fostering better connections with both colleagues and customers, regardless of their location. By prioritizing user experience and accessibility, we ensure that businesses can thrive in their communication efforts.