Alternatives to R.E. Communications IP-PBX Premise based

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.

Why struggle with large providers like Twilio, where response times to support requests can feel endless? At Signalmash, we’re more than just another CPaaS provider. Our commitment is to deliver top-tier CPaaS support. With us, your developers gain access to shared Slack channels with our team for quick assistance, and you’re always welcome to reach out directly to our CEO. Our real-time, high-quality support means faster development and empowers you to offer outstanding service to your customers. Our SMS services include: - SMS API - SMS CPaaS - SMS UCaaS - SMS no-code sending platform - 10DLC campaign support - Short code SMS - Toll-free SMS Voice services available: - Contact center telecoms - Voice termination - Voice origination - Local numbers - Short code numbers - Toll-free numbers - SIP Trunking Our No-code telecoms solutions: - UCaaS for SMS - CCaaS for call management - AI-driven solutions Signalmash – get unmatched support every step of the way. Schedule a call with us today for expert guidance!

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.

A versatile platform and applications tailored for MSPs and UCaaS Providers. Elevate your business with contemporary communication and phone systems, enhancing customer experience without significant upfront expenses. Chosen by a multitude of global customers, UCaaS providers, and resellers, Thirdlane is the preferred solution for those seeking a competitive advantage. Whether for businesses of any scale, government agencies, or call centers, Thirdlane's reliable and user-friendly multichannel communication solutions boost productivity, customer engagement, sales, and cost efficiency. Discover how Thirdlane can bolster your organization; request a quote today. In addition to the telephony features of the Thirdlane Multi Tenant PBX or Thirdlane Business Phone System, customers gain entry to the Thirdlane Connect application (desktop&mobile), covering chat, voice and video calls, conferencing, screen sharing, along with integration capabilities with CRM and diverse applications.

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.

Kamailio® is an open-source SIP server, which evolved from the earlier OpenSER and SER, and is distributed under GPLv2+, capable of managing thousands of call setups each second. This robust platform can be utilized for constructing extensive systems for VoIP, real-time communication, presence, WebRTC, instant messaging, and various other applications. Additionally, Kamailio is well-suited for scaling SIP-to-PSTN gateways, PBX systems, or media servers such as Asterisk™, FreeSWITCH™, or SEMS with ease. Its extensive feature set includes support for asynchronous TCP, UDP, and SCTP, secure communication through TLS for voice, video, and text, as well as WebSocket compatibility for WebRTC. The server supports both IPv4 and IPv6, offers simple instant messaging and presence capabilities with an integrated XCAP server and MSRP relay, and enables asynchronous operations. Furthermore, it includes IMS extensions for VoLTE, ENUM functionality, DID and least cost routing, load balancing, and routing failover. Kamailio also provides comprehensive accounting, authentication, and authorization features, while supporting a variety of backend systems, including MySQL, Postgres, Oracle, Radius, LDAP, Redis, Cassandra, MongoDB, and Memcached. All these capabilities make Kamailio a versatile choice for modern communication infrastructures.

Virtual IP PBX (Private Branch Exchange), or Phone System Appliance – CompletePBX is the heart and soul of Xorcom's hardware and software telephone systems. It's built on top of Asterisk and Linux platforms and is designed for stability and high performance. CompletePBX was built from the ground up to offer ease-of-use, advanced features, and security at all levels. Get a fully functional, free CompletePBX 5 virtual machine evaluation now! CompletePBX certification training is available online for free. It will guide you through the basics of the system. You can save a lot of time and effort if you do it yourself. We can help you get there quicker.

VitalPBX is a unified communications PBX platform based on Asterisk (Centos 7). VitalPBX is a robust, scalable platform that will allow you to manage your PBX in a simple and intuitive manner. VitalPBX has a simple user interface and tooltips throughout the system. Telephony systems should be aware of security risks and take steps to avoid them. VitalPBX offers a variety of levels of protection against cyber threats in VoIP.

Customers can now seamlessly operate from any location thanks to a comprehensive mobility solution that includes essential features for critical operations. Our services allow you to reduce costs through integrated redundancy and the rapid setup of SIP trunks, DIDs, and toll-free numbers. Pulsar360 Corporation’s Voice over IP (VoIP) offerings ensure superior voice communication that can be utilized over cost-effective broadband connections, whether from other providers or through our own managed carrier services. By consolidating their communication needs into a single network connection for both phone and high-speed internet services, clients not only save money but also enhance their productivity. Having dedicated over a decade to refining our VoIP platform, we have developed one of the most advanced and highly automated systems available in North America. This significant progress enables us to provide immediate setup of SIP trunks, hosted and hybrid VoIP PBXs, local and international DIDs, toll-free numbers, and sophisticated management features. With such innovations, businesses can experience a transformation in how they communicate and operate on a daily basis.

Enterprise interaction can be understood as the integration of collaboration and communication efforts. We facilitate secure enterprise interaction through your own private or hybrid cloud environment. The Universal Cloud Communications Stack (UCCS) introduces an innovative approach to utilize big data architectures, allowing for real-time communications and collaboration, which enhances the scalability of transaction-oriented services on a worldwide basis. Key immediate advantages include rapid deployment and automated management that utilize standard infrastructure frameworks. Furthermore, by utilizing virtualized, secure, and scalable containers, we provide hosted and managed communication and collaboration services with great ease, paving the way for new business models for Managed Service Providers (MSPs) and enabling enterprises to achieve significant cost reductions. In fact, communication systems offer the most substantial opportunity for cost savings across all areas of IT. The synergy of contemporary software architecture and data-driven distributed infrastructure presents a powerful opportunity to revolutionize the industry while meeting evolving business needs.

This PBX software platform is designed for easy installation and configuration, and it is fully compatible with our telephony hardware. It caters to small and medium businesses (SMBs) that desire a high-quality phone system typically associated with larger corporations, but without the hefty cost! OpenPBX leverages the user-friendly nature and efficiency of desktop PCs, seamlessly integrating them with traditional telephone handsets to provide advanced features and complete control over your communication system. Whether you need to connect to primary rate (E1/T1) trunk lines or standard POTS lines, Voicetronix offers the appropriate line interface hardware. Customers can take advantage of the OpenPBX Do-it-Yourself (DIY) option, allowing them to construct their own phone system using affordable PC servers running Linux and standard analog telephones. By configuring and maintaining their own phone system, users can eliminate the dependence on a technician for setup and support. Additionally, with a web browser, users can effortlessly navigate the system to select and listen to their voicemail messages, making communication management more convenient and efficient than ever before.

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.

The term "Free" in FreePBX signifies Freedom, highlighting that FreePBX, the leading open-source IP PBX solution globally, empowers users to create a customized phone system that meets their specific requirements. Explore the potential of open-source communication and examine the extensive range of features that FreePBX provides to help you construct your ideal phone system. By integrating commercial modules from Sangoma and various other add-ons, you can elevate your FreePBX installation into a powerful Unified Communications platform. Don't forget to check the Knowledge Base on the FreePBX Wiki for additional resources! Alternatively, you can reach out to FreePBX professionals for assistance with installation, updates, troubleshooting, and more. Available for free download and use, FreePBX's strength is supported by a worldwide community of developers dedicated to maintaining its high compatibility and customizable nature, ensuring it delivers all the essential functions needed to create a scalable business phone system that accommodates any budget. This collaborative effort fosters innovation and continuous improvement, making FreePBX a reliable choice for communication needs.

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.

HoduPBX is the best solution for business communication. HoduPBX's unique user interface and optimal user experience allow businesses to communicate efficiently with their end users. This software provides unparalleled connectivity in an office environment and remote locations, making communication seamless. HoduPBX empowers agents' work management with multitasking capabilities and efficiency. Our IP PBX software is a great choice for business communications. It supports 1000+ concurrent calls and has 10,000 extensions. HoduPBX IP PBX Software offers various sophisticated features such as Add-on Modules, Reseller Module, SIP Trunking, Queue Pro, Voice Transcription, and Webphone. It also offers white label features across IOS, Android, Chrome Extension, Windows, Mac, Linux, and ZohoPhonebridge Plugin.

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.

Dial-Office is an adaptable and comprehensive IP PBX software crafted specifically for small and medium-sized businesses as well as start-ups eager to leverage the advantages of VoIP technology. Utilizing the SIP standard framework, this IP PBX software offers a variety of sophisticated phone functionalities that were once limited to expensive, traditional circuit-switched enterprise telephone systems. The Dial-Office IP PBX software ensures dependable and affordable unified communications for small to mid-sized enterprises over a well-established Local Area Network. This award-winning VoIP solution guarantees users exceptional voice clarity and a range of business management tools aimed at boosting productivity, reducing communication expenses, and improving customer service. In addition to these benefits, the platform boasts features that cater to the specific needs of modern businesses, making it a versatile choice for enhancing communication efficiency.

Yeastar K2 IPPBX is available as both hardware and software. It supports 2000 users, 500 concurrent conversations, and 400 VoIP trunks across multiple locations. This gives medium and large enterprises full flexibility and capability. Yeastar K2IP PBX is designed to meet the communication needs of today and tomorrow. It offers enterprise-level features and exceptional reliability. It also allows remote working with next-level Linkus UC Softphone.

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.

Everything essential for engaging with the FreeSWITCH platform is provided here. FreeSWITCH serves as a Software Defined Telecom Stack, facilitating the shift from traditional telecom switches to a flexible software solution that can operate on any standard hardware, whether it’s a Raspberry PI or a powerful multi-core server, thus unleashing the telecommunications capabilities of various devices. When paired with our cloud platform, SignalWire, FreeSWITCH can connect seamlessly to external networks and adapt to any scale of operation. To simplify your initiation, a pre-configured virtual machine come complete with comprehensive written and video tutorials, allowing you to get started quickly—just install VirtualBox and you will be set up in no time. For those with advanced programming skills, there are detailed instructions available to compile FreeSWITCH from source or explore other manual installation methods. Additionally, if you have a Debian instance available, you can find thorough guidance on how to install our packages from the provided link, ensuring versatility in your approach to utilizing FreeSWITCH. This platform truly empowers both beginners and seasoned developers alike, making telecommunications more accessible than ever.

MikoPBX offers a comprehensive phone solution characterized by high performance, exceptional stability, and extensive features. It can be deployed on-premises through hardware installation or hosted on various cloud platforms like Google Cloud, Amazon Web Services, and Microsoft Azure. Notably, MikoPBX is a free telephone system that provides all essential business functionalities at no cost. Users will appreciate the intuitive web interface, which includes features such as call recording, conference calling, voicemail, call transfers, call pickup, call queues, an interactive voice response (IVR) menu, as well as customizable inbound and outbound call rules. Additionally, the system includes options for managing holiday schedules and night-time call settings. MikoPBX supports the integration of plugins and modules developed globally, allowing for enhanced functionality tailored to specific needs. Importantly, MikoPBX does not impose user licensing fees, making it an economical choice for businesses looking to expand without incurring extra costs for new staff or call recording capabilities. This flexibility in scaling ensures that businesses can grow seamlessly while maintaining a robust communication infrastructure.

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.

*astTECS IPPBX system is the ideal solution for any business, large or small. *astTECS PBX has all the top-end features at an affordable price and is TEC certified by DoT (Govt of India). *astTECS IPPBX solution allows your business to be flexible, innovative, competitive, and future-ready while reducing telecom expenses. *astTECS Call Center Dialer is an omnichannel contact centre solution that offers advanced features and a real-time dashboard for agent monitoring. Advanced analytics and AI-based call center dialer will improve agent productivity and customer engagement. *astTECS provides a complete call recording solution for your company. *astTECS voice recorder is easy to use and offers advanced features. Integration with any PBX is seamless. Supports ISDN/PRI, IP, analog lines and mobile to record all incoming and outgoing calls.

Vodia is a software-based PBX that runs on Windows and Linux. It offers all the advanced features you would expect from a full-featured PBX, including call queues, auto attendants and call forwarding. Vodia is also the pioneer in Multi-Tenancy. You can host Vodia on a dedicated VM( Virtual Machine) in the cloud, premises, and maintain multiple FQDNs for each customer. Sign up for a hosted demo license code for 60 days.

Answer360 provides a comprehensive array of high-quality telecommunication solutions tailored for small to medium enterprises, educational organizations, and mid-sized companies. Our offerings include Hosted Voice over IP (VoIP) services, unified communications, and cloud-based voice software that are all designed to be effective, trustworthy, user-friendly, and cost-effective. We integrate your phone system and services into one cohesive package! Answer360 is at the forefront of enhancing operational communication in today’s intricate business landscape. Our mission is straightforward: to assist businesses in making informed decisions and managing their critical communication processes effectively. Through our innovative approaches, Answer360 fosters connectivity while reducing expenses and enhancing productivity, all supported by our reliable cloud storage solution. Built to be scalable and future-proof, Answer360 adapts its services to meet your specific business requirements, ensuring a perfect fit for your unique needs. With our commitment to innovation and efficiency, we aim to transform the way organizations communicate and operate.

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.

To accelerate your journey toward success, it's essential to have a comprehensive phone system that operates seamlessly, no matter your location, and accommodates the applications and devices you depend on daily. Our cloud-based PBX (private branch exchange) solution meets these requirements and offers numerous advantages typically associated with conventional on-site PBX systems, eliminating the need for extra hardware and physical storage. Comcast Business provides a robust array of hosted voice services tailored for businesses of all sizes, all backed by the largest private voice-over-IP (VoIP) network nationwide. Compared to traditional PBX systems, VoiceEdge delivers superior performance, reduced expenses, easier management, and enhanced scalability. Additionally, our Business VoiceEdge Desktop App now features integrated Unified Communications tools that your business relies on for effective connectivity. This includes a softphone, easily accessible voicemail, screen sharing capabilities, and Chrome™ integration, all conveniently located on your desktop, ensuring you stay connected and productive. With these tools, your communication process becomes smoother and more efficient, empowering your team to reach new heights.

Our award-winning Avaya IP Office, accessible worldwide, provides a comprehensive unified communications platform that is both implemented and managed on-site. To enhance the interactions and experiences of both your customers and employees, delve into the extensive capabilities available with IP Office. You can analyze customer calls effectively by utilizing features such as call recording, tracking, and reporting. With the Avaya Call Reporting functionality integrated into IP Office, you can capture and document incoming calls seamlessly. Furthermore, you have the option to incorporate real-time call center agent seats and dashboards to optimize call management processes. When you pair Avaya IP Office with a subscription to Avaya Spaces, you unlock a versatile application that encompasses voice, video, messaging, conferencing, and calendar functionalities, all of which are accessible from any device and location. This integration not only streamlines communication but also enhances overall productivity across your organization.

OpenSIPS is an open-source SIP proxy/server designed for handling voice, video, instant messaging, presence, and various other SIP extensions. This versatile signaling SIP server caters to carriers, telecom operators, and ITSPs, providing solutions for diverse applications such as Class 4 and Class 5 residential platforms, trunking and wholesale services, enterprise and virtual PBX systems, session border controllers, application servers, load balancers, IMS platforms, call centers, and much more. Its high throughput capability supports tens of thousands of calls per second and millions of simultaneous connections, making it a preferred choice for various SIP scenarios and services. OpenSIPS excels due to its flexible routing and integration options, which allow for custom routing logic through routing scripts and numerous interfacing APIs. With over 120 modules available, it facilitates effective application development for SIP handling, backend operations, and routing logic. Additionally, the 3.2 release of OpenSIPS underwent a thorough security audit, enhancing its safety and reliability for users across the board. This robust framework not only meets current demands but also continuously evolves to address future challenges in the telecommunications landscape.

Yeastar P-Series PBX Systems is more than a PBX. It is a converged system designed for SMEs with higher expectations in business communications. The P-Series is a "PBX-Plus-More” solution that unifies voice, video, applications and collaboration. This allows SMEs to achieve more productivity, fill the UC gap and complete more complex tasks.

Zero consolidates all your enterprise communication needs—voice, video, and messaging—into a singular, mobile-friendly cloud platform tailored for today’s workforce. With Zero's comprehensive solution, you can effortlessly integrate all business communications. Features include multi-level auto attendants, ring groups, designated business hours, voicemail, and additional functionalities. Enjoy instant provisioning and the activation of phone numbers spanning over 50 countries. Reduce costs on mobile and internal calls within your organization by utilizing our complimentary VoIP applications. Organize conference calls accommodating as many as 40 participants and facilitate video meetings with up to 6 attendees. Enhance team interactions with secure messaging options for both individual and group chats. Other features include simultaneous ringing, call transfers, three-way calls, call holding, and recording capabilities. You can make and receive calls from your desktop, mobile device, or landline while maintaining your business phone number. Additionally, you can access your voicemail, enterprise messaging system, and conferencing services from virtually anywhere, ensuring seamless communication for your team. This all-in-one solution is designed to empower your workforce and streamline your communication processes effectively.

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.

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.

Dialexia operates as a worldwide VoIP software provider focused on creating IP-PBX solutions. Our primary clientele includes VoIP service providers, telecom operators, value-added resellers (VAR), and competitive local exchange carriers (CLEC). The hosted PBX systems we offer are crafted to minimize capital expenditures, lower ongoing costs, and improve management efficiency. With Dialexia's IP-PBX offerings, small and medium-sized businesses can enjoy access to cutting-edge unified communications features without breaking the bank. Our comprehensive enterprise IP-PBX Appliances provide a plug-and-play solution for SMBs in search of cost-effective VoIP options. Furthermore, Dialexia’s HERO hosted PBX solution serves as an exceptional communications platform tailored for telecom operators and service providers eager to implement a scalable and budget-friendly hosted PBX service for their clientele. By choosing Dialexia, businesses can leverage the latest technology to enhance their communication strategies effectively.

InGenius Connector Enterprise seamlessly integrates existing telephone systems into top CRMs using an enterprise-proven solution. Upland InGenius adapts to specific contact center requirements by using innovative computer telephony Integration (CTI). InGenius is backed by 20 years of experience in enterprise telephony and superior customer service. It helps contact centers improve productivity and the customer experience. InGenius Connector Enterprise provides contact centers with features such as screen pop, click to-dial, automated call logging, and call reports. Agents can use the solution to efficiently serve customers, and they can see a complete view of interactions for intelligent decision making. Supported CRMs: Microsoft Dynamics 365, Salesforce, ServiceNow Supported Phone Systems: Asterisk, Avaya, Cisco, Genesys, Mitel

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.

As a leading provider of Business Phone solutions in North America, we take great pride in delivering top-tier VoIP phone systems to small and medium-sized enterprises across Canada and the United States, ensuring they receive a dependable and secure service supported by our approachable and experienced customer service team. Additionally, we stand out as one of the select few VoIP solution providers in the region that safeguard client data through geo-redundant data centers located in both the U.S. and Canada, utilizing advanced security measures to ensure the utmost reliability and protection for our customers. Our commitment to excellence and security sets us apart in a competitive industry, allowing businesses to focus on their growth while we handle their communication needs.

IQ Telecom offers tailored software and solutions designed to meet the unique needs of various businesses. Their extensive range of services, including VoIP, IVR, Dialer, SIP Termination, and IP PBX Phone Systems, are not only highly scalable but also available in hosted (virtual) and on-premise formats, allowing clients to significantly reduce costs and save valuable time, ultimately providing them with a competitive advantage in their respective industries. Furthermore, this adaptability ensures that businesses can grow seamlessly while leveraging advanced technology to enhance their operations.

Since 2003, we have been dedicated to assisting small and medium-sized enterprises in their growth journeys. Our offerings include a variety of cost-effective phone systems equipped with a multitude of features, high-speed internet services, and comprehensive IT and cybersecurity solutions. In recent years, VoIP (Voice over Internet Protocol) technology has progressed, facilitating a seamless transition of voice services to the ‘cloud.’ The term cloud generally describes services hosted in data centers that are accessed via the internet. Cloud telephony specifically pertains to voice services, focusing on replacing conventional telephone equipment like PBXs with services provided by third parties. By implementing intelligent call management and analyzing behavioral patterns, businesses can significantly enhance customer satisfaction and make more informed decisions. This advancement allows smaller companies to effectively compete with larger corporations by expanding their geographic reach and improving initial customer interactions. Additionally, users can remotely access their systems from any location, ensuring that they maintain the full functionality of their office phones while on the go. Embracing these technologies can dramatically elevate a business's operational efficiency and customer engagement.

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.

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.

Brekeke SIP Server, which adheres to the widely recognized Session Initiation Protocol (SIP) standards, delivers a dependable and scalable platform tailored for telecommunications carriers, communication service providers, integrators, and manufacturers of SIP-based products. In accordance with the SIP baseline specification RFC 3261, Brekeke SIP Server encompasses the roles of a SIP registrar server, SIP redirect server, and SIP proxy server. Functioning as a stateful proxy, it efficiently manages session statuses, ensuring optimal session control processing. It is utilized by faculty, students, and staff within degree-granting academic institutions for a variety of purposes, including the evaluation of the Brekeke SIP Server Standard Edition. Additionally, the server supports SIP Trunk Services and can be integrated into smaller SIP communication systems, making it an excellent choice for research and development as well as product demonstrations. With its unmatched reliability and strong performance, Brekeke SIP Server stands out as a premier solution in the SIP server market.

Explore innovative, feature-rich solutions tailored for your business needs. Elevate your operations and provide an exceptional customer experience with Cloudli's advanced mobile-first business communications technology. By adopting our Cloud Business Phone solution, you can enhance your business efficiency, modernize your communication channels — encompassing voice, text, and shared phone lines — all while reducing costs. Boost your connectivity to maintain a competitive edge in your industry. Transform your phone and fax systems digitally with Cloudli, ensuring top-notch continuity, exceptional customer support, and high service quality. Our goal is to empower businesses of any size to communicate effectively — wherever and whenever — without sacrificing security, reliability, or efficiency. For over twenty years, we have pioneered the development of business communication solutions that specifically target common challenges faced by organizations. The results speak for themselves, showcasing the positive impact of our services.

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.