Top OpenSIPS Alternatives

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

Why endure the slow response times of major providers like Twilio when you can experience prompt assistance with Signalmash? At Signalmash, we differentiate ourselves from typical CPaaS providers by prioritizing exceptional support. Our developers benefit from direct access to shared Slack channels with our team, ensuring they receive immediate help whenever needed, and our CEO is always approachable for any inquiries. With our commitment to delivering high-quality, real-time support, we facilitate quicker development cycles, allowing you to provide remarkable service to your customers. We offer a range of SMS services, including: - SMS API - SMS CPaaS - SMS UCaaS - SMS no-code sending platform - 10DLC campaign support - Short code SMS - Toll-free SMS Our voice services encompass: - Contact center telecoms - Voice termination - Voice origination - Local numbers - Short code numbers - Toll-free numbers - SIP Trunking Additionally, we provide no-code telecom solutions such as: - UCaaS for SMS - CCaaS for call management - AI-driven solutions Choose Signalmash for unparalleled support throughout your journey. Contact us today to schedule a call and receive expert assistance tailored to your needs!

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

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

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

All the necessary resources for interacting with the FreeSWITCH platform are available right here. FreeSWITCH acts as a Software Defined Telecom Stack, enabling a transition from conventional telecom switches to a versatile software-based solution that can run on any standard hardware, from a Raspberry PI to a robust multi-core server, thus amplifying the telecommunications functionalities of various devices. When integrated with our cloud platform, SignalWire, FreeSWITCH can effortlessly connect to external networks and scale operations as needed. To ease your onboarding process, a pre-configured virtual machine is offered, equipped with extensive written and video tutorials, so you can quickly get started with just a VirtualBox installation. For those who are more technically inclined, comprehensive instructions are provided for compiling FreeSWITCH from source or exploring alternative manual installation techniques. Furthermore, if you have access to a Debian instance, you will find detailed guidance on installing our packages from the provided link, allowing for a flexible approach to implementing FreeSWITCH. This platform truly empowers both novices and experienced developers, making telecommunications more approachable than ever while fostering innovation and creativity in the field. Whether you’re looking to build a simple communication tool or a complex telecom solution, FreeSWITCH has the capabilities to meet your needs.

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

This PBX software platform is crafted for straightforward installation and configuration, ensuring full compatibility with our telephony hardware. It is specifically tailored to meet the needs of small and medium businesses (SMBs) that aspire to utilize a premium phone system akin to those found in larger enterprises, all while avoiding excessive expenses. OpenPBX harnesses the accessibility and effectiveness of desktop computers, effortlessly merging them with conventional telephone handsets to offer sophisticated features and complete oversight of your communication system. Whether you're looking to connect to primary rate (E1/T1) trunk lines or regular POTS lines, Voicetronix supplies the necessary line interface hardware. Customers have the option to utilize the OpenPBX Do-it-Yourself (DIY) feature, enabling them to build their own phone systems with cost-effective PC servers operating on Linux and regular analog telephones. By taking charge of the setup and maintenance of their phone systems, users can free themselves from reliance on technicians for installation and support. Furthermore, with a simple web browser interface, users can easily manage their voicemail messages, enhancing the overall experience of communication management. This flexibility empowers users to tailor their phone systems to their specific preferences and requirements.

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

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

The Yeastar K2 IPPBX can be accessed in both hardware and software formats, accommodating up to 2000 users, managing 500 simultaneous calls, and supporting 400 VoIP trunks across various sites. This flexibility makes it an excellent choice for medium to large businesses that require robust communication solutions. Designed to cater to the evolving demands of modern communication, the Yeastar K2 IPPBX provides advanced enterprise features along with remarkable reliability. Additionally, it facilitates remote work environments through the innovative Linkus UC Softphone, ensuring seamless connectivity for users regardless of their location. By incorporating this technology, companies can enhance their operational efficiency and improve collaboration among team members.

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

A dynamic platform and specialized applications designed specifically for MSPs and UCaaS Providers. Transform your business with modern communication and telephony systems that improve customer interactions while minimizing upfront investments. Widely adopted by numerous international clients, UCaaS providers, and resellers, Thirdlane stands out as the go-to solution for those aiming to gain a competitive edge. Whether serving enterprises of various sizes, governmental organizations, or call centers, Thirdlane's dependable and intuitive multichannel communication solutions enhance productivity, foster customer engagement, drive sales, and improve cost-effectiveness. Explore how Thirdlane can strengthen your organization by requesting a quote today. Additionally, alongside the telephony features offered by the Thirdlane Multi Tenant PBX or Thirdlane Business Phone System, customers also enjoy access to the Thirdlane Connect application (available on both desktop and mobile), which encompasses chat, voice and video calls, conferencing, screen sharing, and seamless integration with CRM systems and various applications, providing a comprehensive communication experience. This robust array of features is designed to meet the evolving needs of today's businesses.

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

Customers can now effortlessly engage in operations from virtually any location due to a robust mobility solution that encompasses vital features for essential tasks. Our offerings facilitate cost reduction through integrated redundancy, enabling swift setup of SIP trunks, direct inward dialing (DIDs), and toll-free numbers. Pulsar360 Corporation’s Voice over IP (VoIP) services guarantee exceptional voice communication that can be leveraged over economical broadband connections, whether sourced from other providers or through our own managed carrier services. By unifying their communication requirements into a single network connection for both telephony and high-speed internet, clients not only cut costs but also boost their productivity levels. With more than ten years dedicated to enhancing our VoIP platform, we have crafted one of the most sophisticated and highly automated systems available in North America. This substantial advancement allows us to offer immediate provisioning of SIP trunks, hosted and hybrid VoIP PBXs, local and international DIDs, toll-free numbers, and advanced management capabilities. Such innovations empower businesses to experience a significant transformation in their communication and operational processes on a day-to-day basis, leading to increased efficiency and streamlined workflows. Ultimately, this comprehensive approach positions organizations to thrive in a competitive landscape.

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

The globally recognized Avaya IP Office offers a robust unified communications platform that is both deployed and supervised on-site. To elevate the engagement and experiences of your customers and employees alike, explore the wide range of functions provided by IP Office. You can effectively assess customer interactions by taking advantage of features like call recording, monitoring, and detailed reporting. With the inclusion of Avaya Call Reporting within IP Office, capturing and documenting incoming communications becomes a seamless task. Additionally, you have the flexibility to add real-time call center agent workstations and dashboards, which significantly enhances the efficiency of call management processes. When Avaya IP Office is combined with a subscription to Avaya Spaces, you gain access to a multifunctional application that integrates voice, video, messaging, conferencing, and calendar services, all available from any device and location. This powerful integration not only simplifies communication but also drives productivity improvements throughout your organization, ensuring that everyone can collaborate effectively no matter where they are. Ultimately, leveraging these tools can lead to a more responsive and agile business environment.

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

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

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

Enterprise interaction encompasses the seamless integration of communication and collaboration efforts. We enable secure enterprise interaction within your own private or hybrid cloud framework. The Universal Cloud Communications Stack (UCCS) presents a groundbreaking method to leverage big data architectures, facilitating real-time communication and collaboration that boosts the scalability of transaction-driven services globally. Immediate benefits include swift deployment and automated management utilizing standard infrastructure frameworks. Moreover, by employing virtualized, secure, and scalable containers, we deliver hosted and managed communication and collaboration services effortlessly, which opens doors to innovative business models for Managed Service Providers (MSPs) and helps enterprises realize significant cost savings. In fact, the potential for cost reduction through communication systems is among the most considerable across the entire IT spectrum. The combination of modern software architecture with data-centric distributed infrastructure offers a remarkable chance to transform the industry while addressing the dynamic needs of businesses. This evolution not only enhances operational efficiency but also positions enterprises to thrive in an increasingly competitive landscape.

The Yeastar P-Series PBX Systems transcends traditional PBX functionality, offering a comprehensive solution tailored for small and medium-sized enterprises that demand excellence in business communication. As a "PBX-Plus-More" offering, it integrates voice, video, applications, and collaborative tools, enabling SMEs to enhance productivity significantly, bridge the gaps in unified communications, and tackle more intricate tasks with ease. This innovative system empowers businesses to streamline their operations and improve overall efficiency.

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

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

IdentityServer is a flexible framework that adheres to industry standards for OpenID Connect and OAuth 2.0, specifically designed for ASP.NET Core, which enables complete oversight of the user interface, overall user experience, business logic, and data management practices. It has been officially certified by the OpenID Foundation, featuring a comprehensive range of protocols and standards established by both the OpenID Foundation and IETF working groups. The framework supports a multitude of hosting options, making it suitable for deployment on-premises, in cloud settings, behind VPNs, and across various platforms including Windows, Linux, Docker, and Kubernetes. Additionally, Duende Software enriches this offering with complementary products like IdentityServer for Redistribution and a Backend for Frontend (BFF) security framework. To ensure developers can effectively leverage these tools, a wealth of documentation and training materials is readily available. Our dedication to maintaining full compliance with relevant standards is a fundamental priority, allowing us to offer extensive access to all components of the OAuth and OpenID Connect protocol family. Moreover, this comprehensive approach equips organizations with the ability to establish secure authentication and authorization mechanisms tailored to their unique requirements, fostering an environment of trust and security. Therefore, IdentityServer not only simplifies the implementation of authentication protocols but also enhances overall operational efficiency for businesses.

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

Established in 2010, Wavetel has emerged as a significant player in the telecommunications sector both in the UK and internationally. Initially starting as a wholesale provider, the company specialized in delivering Voice, SMS, and Data services to various operators. In 2013, Wavetel expanded its offerings by launching B2B services that encompass VoIP telephony, connectivity options, virtual numbers, SIP trunks, and Microsoft Teams Direct Routing. Additionally, the company provides mobile, SMS, and call center solutions to its esteemed clientele across the globe. Over time, Wavetel's voice messaging solutions have transformed into a comprehensive enterprise offering, designed to seamlessly integrate with third-party CRM systems and other business applications via customizable APIs. This adaptability ensures that Wavetel remains at the forefront of technological advancements in the telecom industry.

Transforming communication and collaboration through digital means is crucial for contemporary businesses. AudioCodes has introduced the Software-Defined Voice Network (SDvN) by leveraging Software-Defined Networking (SDN) principles, offering a versatile solution specifically designed for large enterprises embarking on their digital transformation journey. This cutting-edge framework enhances voice network management and optimizes call routing efficiency by dismantling communication barriers, integrating seamlessly with SIP trunk services, and utilizing the corporate IP network for enhanced call handling. The solution features AudioCodes' session border controllers (SBCs) along with an array of tools for global call routing, policy management, user oversight, and voice network administration, which together create a unified network that effortlessly connects both on-premises and cloud-based communication systems. Additionally, SDvN empowers organizations to upgrade their communication infrastructures without the need to discard existing platforms, thereby facilitating a more seamless transition into the evolving landscape of business communications. This adaptability not only aids in future-proofing their communication strategies but also supports improved collaboration across teams and departments.

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

Discover innovative telecommunications software paired with a powerful communication network aimed at boosting your business performance. Our hosted solutions are crafted to improve both your outbound and inbound calling processes while also reducing your expenses. We provide enterprise-grade SIP Trunking at attractive wholesale rates, ensuring you have unlimited capacity for high-quality, high-volume VoIP services. By using our system, you can significantly decrease your office telephony costs and enhance your operations with features such as custom call routing, call recording, and detailed reporting. Our network is specifically designed to handle exceptionally high call volumes without compromising on call quality or reliability. For businesses looking to expand their client outreach, Blue Telecoms offers advanced functionalities including top-notch answer machine detection, outbound number rotation, and effective lead recycling tactics. We partner exclusively with tier 1 VoIP carriers and infrastructure, ensuring you receive the highest level of service and dependability. Furthermore, our diverse carrier relationships guarantee that your essential projects proceed smoothly, regardless of any challenges that may arise. With Blue Telecoms, elevate your communication capabilities and maximize your business potential like never before. Experience the difference in service quality and efficiency today.

Arch functions as an advanced gateway that protects, supervises, and customizes the performance of AI agents by fluidly connecting with your APIs. Utilizing Envoy Proxy, Arch guarantees secure data handling, smart traffic management, comprehensive monitoring, and smooth integration with backend systems, all while maintaining a separation from business logic. Its architecture operates externally, accommodating a range of programming languages, which facilitates quick deployments and seamless updates. Designed with cutting-edge sub-billion parameter Large Language Models (LLMs), Arch excels in carrying out critical prompt-related tasks, such as personalizing APIs through function invocation, applying prompt safeguards to reduce harmful content or circumventing attempts, and identifying shifts in intent to enhance both retrieval accuracy and response times. By expanding Envoy's cluster subsystem, Arch effectively oversees upstream connections to LLMs, promoting the development of powerful AI applications. In addition, it serves as a front-end gateway for AI applications, offering essential features like TLS termination, rate limiting, and prompt-based routing. These robust functionalities establish Arch as a vital resource for developers who aspire to improve the effectiveness and security of their AI-enhanced solutions, while also delivering a smooth user experience. Moreover, Arch's flexibility and adaptability ensure it can evolve alongside the rapidly changing landscape of AI technology.

A2Billing operates as both a class 4 and class 5 softswitch, integrating inline billing functionalities and designed to provide VoIP solutions for residential, commercial, and wholesale sectors, while also offering features like calling cards, call-back services, and telephone number reselling, all backed by our dedicated customer support team. This open-source software works seamlessly with Asterisk, enabling thorough telecom customer management that includes interfaces for admin, agents, customers, and online registration, in addition to delivering dynamic inline rating and real-time billing for calls and services. Its adaptability and robust support render it a crucial asset in the telecommunications field, helping businesses streamline their operations and enhance customer satisfaction. Moreover, the platform's continuous updates ensure that users benefit from the latest technological advancements in the industry.

Sangoma Wholesale Carrier Services, formerly known as VI Communication Services, has been specifically designed to help Internet telephony service providers (ITSPs) and managed service providers (MSPs) effortlessly deliver the high-caliber services their customers expect. Businesses can take advantage of our communication APIs to weave voice and messaging features into their applications, enabling smooth communication with their client base. By leveraging our services, ITSPs, MSPs, or value-added resellers (VARs) can enrich their current retail SIP trunking offerings by adding extra phone numbers and VoIP functionalities that might not be part of their existing systems, thus unlocking new revenue streams. If you are just starting out, there is also the possibility to establish a new retail SIP trunking service, allowing you to incorporate desired features for your clients, such as messaging, conference bridging, and a range of other services to improve their overall experience. This adaptability not only empowers businesses to meet the demands of their clientele but also positions them to thrive in a constantly shifting telecommunications landscape. Moreover, the ability to customize and expand services as needed fosters long-term growth and customer loyalty.

Telecom providers, along with Unified Communications and OTT services, have the capability to set up, introduce, and profit from a variety of communication services over IP networks. These offerings encompass Unified Communications, Hosted IP PBX / IP Centrex, and SIP Trunking, in addition to various management tools such as DID Management and calling card services. Important components of this system include VoLTE and VoWiFi, which enhance communication efficiency. Among the critical functional modules are Class 4 and Class 5 SIP softswitches, designed for both residential and enterprise telephony solutions, as well as options for convergent online billing, service provisioning, and charging features that can be tailored to user needs. This flexibility ensures that service providers can adapt to the evolving demands of their customers while maximizing their revenue potential.

Toll-free and local numbers are offered in over 160 nations, allowing you to forward incoming calls from any phone, PBX, or SIP system. Our extensive global network is designed with multiple layers of redundancy to ensure reliability. You can utilize any of your business numbers as Caller ID for international outbound calls, and displaying a local caller ID can significantly enhance your success rates. To facilitate outbound calling, simply connect our SIP trunks to your existing phone system. Implementing smart calling routing features like auto-attendant/IVR, time-based routing, and simultaneous ringback tones is straightforward. You also have the flexibility to customize greetings, record calls, manage voicemail, and access call recording functionalities. Adjustments to virtual numbers, calling plans, and advanced features can be done effortlessly, while you can also download call detail records and invoices. Technical support is available around the clock, ensuring assistance whenever you need it. For personalized help, reach out to your dedicated account manager. Additionally, our user-friendly interface allows you to manage all aspects of your communication seamlessly, enhancing your overall experience.

The D-TAC "HD" series billing platform presents a holistic post-paid CDR billing solution that integrates effortlessly with various technologies such as traditional switches, IP switches, gateways, proxy servers, soft switches, and border session controllers. Designed to provide exceptional billing services for both VoIP and TDM-based CDR networks, this powerful platform is tailored to meet the specific needs of telecom companies. D-HD empowers telecom enterprises with unique billing and automation functions that streamline operational processes. Each department within your organization can access personalized screens, functionalities, and tools to boost overall efficiency. The system encompasses a wide range of resources, from customer support systems to rate management tools, all structured to ensure smooth task execution. For users requiring advanced CDR management functionalities, the D-TAC HD Billing System stands out with its invoicing options, cash receipt tracking, carrier cost analysis, ASR reporting, and detailed statistical insights into route or trunk group performance. The core of our offering includes robust billing and authentication services, featuring extensive AAA RADIUS capabilities alongside additional AARA functionalities, ensuring a comprehensive approach to telecom billing. Moreover, this adaptable system is designed to respond to the changing needs of the industry, establishing itself as a dependable choice for future expansion and innovation. By investing in the D-TAC HD Billing System, businesses can confidently navigate the complexities of telecom billing while positioning themselves for sustained growth in a competitive market.

VoIP Route's hosted phone service leverages state-of-the-art cloud technology to improve your business connectivity while removing the significant costs tied to traditional premises-based systems, all without compromising on quality. Effortlessly connect and collaborate through phones, smart devices, computers, or laptops. As a leading SIP trunk service provider, VoIP Route is equipped to serve both IP-PBX systems and analog/digital telephone adapters or gateways. Our dedication to providing exceptional service at competitive rates has earned the confidence of both our partners and customers, bolstered by unmatched support. Residential VoIP harnesses innovative technology to guarantee optimal performance while greatly lowering expenses. By substituting outdated phone lines with digital data, you can make calls using the Internet, rendering residential VoIP far more economical than conventional phone services. This shift not only reduces monthly costs but also enhances the overall communication experience for users, enabling them to stay connected more effectively than ever before. Ultimately, embracing VoIP technology can transform how you manage your communication needs.

Azure Route Server enables the automatic sharing of routing information between network devices and Azure virtual networks. By integrating your network devices with Azure ExpressRoute and VPN gateways, you can effortlessly obtain updated routing information from Azure Route Server, which removes the necessity for direct communication with each separate network. This capability supports dynamic routing between your network devices and Azure gateways, allowing organizations to move beyond the constraints of static routing. With the provision of Border Gateway Protocol (BGP) endpoints, Azure Route Server utilizes a widely accepted routing protocol to facilitate effective route exchanges. Its seamless compatibility with both established and new setups accommodates diverse network configurations, including hub and spoke, full mesh, or flat virtual networks. This adaptability not only aids in optimizing network architecture but also ensures that organizations can implement efficient routing strategies tailored to their specific needs. Consequently, Azure Route Server becomes a vital tool for enhancing network performance and reliability.

Infinity Telecom functions as a licensed telecommunications provider in the Czech Republic, actively involved in the global exchange of traffic with partners for both premium and wholesale termination services. Headquartered in Prague, this group of telecom operators offers a comprehensive array of services that encompass data storage and transmission, traditional telephony, IP solutions, Internet access, content delivery, and colocation across various regions. In addition, we excel in global voice services, providing SIP trunking and DID numbers, while also offering IP transit and hosting options that include server rentals, colocation, and hardware hosting. Moreover, we cater to local telecommunications needs in Czechia and Slovakia, with a focus on Internet access and office communication solutions. Our strategically positioned Points of Presence (PoPs) in key cities like Prague, Bratislava, Amsterdam, Mumbai, and Barranquilla significantly enhance our capacity to meet the needs of clients around the globe. This vast network not only strengthens our service delivery but also ensures that we can effectively address a wide range of communication requirements for our diverse clientele. Our commitment to innovation and client satisfaction drives us to continuously expand our offerings and improve our infrastructure.

Cool Telecom provides a cloud-based telephony solution designed to meet the needs of businesses, regardless of their size, featuring a flexible interface that seamlessly integrates CRM and Helpdesk features to enhance sales and improve communication for support teams. Clients can easily obtain phone numbers from more than 50 countries and scale their workforce based on seasonal fluctuations, while taking advantage of detailed insights through real-time analytics. The platform's accessibility on any device enables users to make and receive calls from almost any location, ensuring connectivity at all times. With Cool Telecom's multifunctional app, users can effectively manage call routing according to various factors such as IVR selections, agent skills, time zones, and more, all through an intuitive dashboard. Moreover, the platform offers sophisticated, real-time analytics on agent and team performance, allowing businesses to monitor activity via a live feed and enabling data cross-referencing with current CRM systems for enhanced insights and opportunities for improvement. Such capabilities not only keep businesses connected but also foster ongoing advancements in customer service and overall operational effectiveness, creating a robust environment for growth and satisfaction.

Inertia.js functions as a framework that enables developers to build modern single-page applications by leveraging established server-side routing and controllers. It streamlines the process of developing fully client-rendered SPAs, alleviating many of the complexities that typically arise with such applications. By removing the necessity for client-side routing or separate APIs, Inertia.js allows developers to continue utilizing their familiar controllers and page views without interruption. Acting as a conduit between traditional server-side frameworks and contemporary frontend technologies, it offers official client-side adapters for React, Vue, and Svelte, as well as server-side adapters for Laravel, Rails, and Phoenix. This innovative approach not only permits the crafting of modern SPAs but also maximizes the use of existing server-side practices. Importantly, Inertia is not meant to supplant existing server-side or client-side frameworks; rather, it is intended to enhance them. You can envision Inertia as a vital link that integrates the two domains of web development. Through its implementation of adapters, Inertia simplifies workflows and improves the overall developer experience. By utilizing this framework, developers gain the advantage of both worlds, allowing them to work with increased efficiency and a sense of familiarity in their processes. Ultimately, Inertia.js represents a forward-thinking solution that bridges the gap between traditional and modern web development techniques.

The Ingate SIParator® is a highly efficient and flexible Enterprise Session Border Controller (E-SBC) that provides secure and seamless SIP connectivity, facilitating the integration of PBXs and Unified Communications (UC) systems with SIP trunking providers. By optimizing SIP trunking operations, the SIParator connects remote UC endpoints, aggregates SIP trunks, and distributes sessions across multiple locations and service delivery points. This solution is extensively employed to secure Real-Time communications, guarantee SIP interoperability, and offer a wide array of connectivity options. Its robust design ensures compatibility with all existing network infrastructures, featuring a standard SIP proxy and a SIP registrar. Additionally, it supports Network Address Translation (NAT) and Port Address Translation (PAT), along with Transport Layer Security (TLS) and Secure Real-Time Transport Protocol (SRTP) to encrypt both SIP signaling and media streams. By tackling prevalent security issues in enterprise VoIP, the SIParator provides organizations with a secure and reliable communication landscape. Its adaptability and dependability not only enhance communication frameworks but also position it as a top choice for businesses aiming to improve their communication systems. Ultimately, the SIParator stands as a critical asset for organizations seeking to modernize and secure their communication infrastructures.

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

The JeraSoft Billing Platform is a flexible billing system tailored for mobile and business telecommunications, aimed at organizations such as telephony service providers, internet service providers, and competitive local exchange carriers. This platform enables users to optimize the billing processes across various services, resulting in improved efficiency and precision. Moreover, it automates essential functions like routing, rating, and mediation, while managing a substantial subscriber base with ease. By utilizing the features of the JeraSoft Billing Platform, companies can markedly enhance their operational processes and the quality of service they offer. Ultimately, this tool not only simplifies complex billing tasks but also empowers businesses to scale their operations more effectively.

Squid operates as a caching proxy for web traffic, supporting a variety of protocols including HTTP, HTTPS, and FTP. By storing frequently accessed web pages, it greatly reduces bandwidth consumption and improves response times for users. Its advanced access control mechanisms allow Squid to function as an effective server accelerator. Furthermore, it is versatile, working on multiple operating systems, including Windows, and is distributed under the GNU GPL license. A multitude of Internet service providers around the world depend on Squid to enhance user access to the web. By optimizing the data exchange between clients and servers, Squid not only enhances performance but also conserves bandwidth through the retention of commonly visited content. Additionally, it possesses the ability to direct content requests using various routing techniques, which facilitates the creation of cache server hierarchies that optimize network performance. Many online services leverage Squid to significantly refine their content delivery systems. Ultimately, the integration of Squid can lead to a considerable decrease in server load and an acceleration in the speed at which users receive content. Its capacity to efficiently manage web traffic positions it as an essential asset for both service providers and website administrators striving for improved user experiences. Overall, Squid’s versatility and efficiency make it an indispensable component in the realm of internet infrastructure.

Kuma is an open-source control plane specifically designed for service mesh, offering key functionalities such as security, observability, and routing capabilities. Built on the Envoy proxy, it acts as a modern control plane for microservices and service mesh, supporting both Kubernetes and virtual machine environments, which allows for the management of multiple meshes within a single cluster. Its architecture comes equipped with built-in support for L4 and L7 policies that promote zero trust security, enhance traffic reliability, and simplify observability and routing. The installation of Kuma is remarkably easy, typically requiring just three straightforward steps. With the integration of the Envoy proxy, Kuma provides user-friendly policies that significantly improve service connectivity, ensuring secure and observable interactions among applications, services, and databases. This powerful solution enables the establishment of contemporary service and application connectivity across various platforms and cloud environments. Furthermore, Kuma adeptly supports modern Kubernetes configurations alongside virtual machine workloads within the same cluster, offering strong multi-cloud and multi-cluster connectivity to cater to the comprehensive needs of organizations. By implementing Kuma, teams can not only simplify their service management processes but also enhance their overall operational effectiveness, leading to better agility and responsiveness in their development cycles. The benefits of adopting Kuma extend beyond mere connectivity, fostering innovation and collaboration across different teams and projects.

Unlock your favorite websites for free and circumvent government restrictions effortlessly with the Proxynel proxy app, a swift solution for accessing blocked content—simply launch the app and you'll be able to browse nearly any restricted site in your area. This proxy browser comes equipped with integrated US proxy servers, ensuring that your online privacy remains intact; being an Android application, it channels your web traffic through these US servers, leaving no digital footprint and providing a reliable way to avoid government surveillance. Moreover, you can say goodbye to the frustration of Google Captcha verifications while searching, as the Proxynel app automatically generates a fresh IP address with each request, which prevents Google from recognizing repetitive anonymous traffic. With its smooth functionality and superior security features, this application is undoubtedly a revolutionary tool for internet users seeking freedom online! Overall, Proxynel enhances your browsing experience while protecting your identity in an increasingly monitored digital landscape.

ProxySQL is built on a complex multi-core architecture that adeptly manages hundreds of thousands of concurrent connections, efficiently distributing these across multiple servers. It provides advanced sharding options that can be tailored by user, schema, or table, thanks to its intricate query rule engine and customizable plugins. The development team benefits from not having to modify queries produced by Object-Relational Mappers (ORMs) or pre-packaged applications, as ProxySQL's adaptive query rewriting capability can modify SQL statements dynamically as required. The term "battle-tested" only begins to describe its robustness; ProxySQL has demonstrated its reliability under the toughest circumstances. Prioritizing performance, the metrics available clearly illustrate its capabilities. As a high-performance, open-source, and highly available SQL proxy for both MySQL and PostgreSQL, ProxySQL acts as an essential bridge between database clients and servers. This comprehensive suite of features is crafted to simplify and enhance database operations, ultimately empowering organizations to leverage their database infrastructure more effectively. Furthermore, the innovative nature of ProxySQL ensures that organizations can reach unmatched levels of efficiency and dependability in managing their database operations.

CenterPoint Securities offers a comprehensive suite of products, services, and partnerships aimed at facilitating the enhancement and oversight of your trading activities. As a direct market access stock broker, the firm empowers its clients by granting them greater control over their order flow. The DAS Trader Pro platform supports direct access trading with advanced order types, real-time charting features, and multi-account management capabilities, ensuring a seamless trading experience. In addition, the Sterling Trader® Pro platform acts as an adaptable front-end trading tool, boasting a distinctive array of functionalities tailored to meet diverse trading requirements. By collaborating with clearing firms known for their impressive easy-to-borrow lists and exceptional locate providers, CenterPoint has established itself as a formidable choice for traders who employ short-selling strategies. The routing options provided are unparalleled, and continuous improvements based on client insights help introduce new features that cater to evolving needs. Retail clients benefit from institutional-grade routing alternatives, equipping them with essential tools to enhance their trading performance. This dedication to innovation, coupled with a strong focus on customer satisfaction, distinguishes CenterPoint Securities in the competitive realm of trading services, reinforcing its position as a leader in the industry. The ongoing evolution of their platforms ensures that clients remain well-prepared for the dynamic nature of the market.

IBM Sterling Secure Proxy acts as a protective layer for your dependable network by preventing direct links between external partners and internal servers. This solution significantly boosts the security of your network and data transfers at the perimeter, allowing for secure business-to-business transactions and managed file transfer (MFT) sharing. Functioning as a proxy located in a demilitarized zone (DMZ), Secure Proxy utilizes multifactor authentication, SSL session terminations, the mitigation of inbound firewall vulnerabilities, protocol inspection, and other measures to protect your trusted environment. Furthermore, it provides guidelines for effective firewall management aimed at enhancing perimeter security, which results in improved overall protection for your data and secure zones. The implementation of multifactor authentication is a prerequisite for accessing backend systems, ensuring a tighter grip on connection validation and control. In addition, its proxy capabilities and features are designed to enhance the security of file transfers at the network's edge. With sophisticated routing capabilities, it simplifies infrastructure updates, ultimately saving time and reducing the risks associated with system transitions. Overall, IBM Sterling Secure Proxy offers a holistic approach to network security, reinforcing the integrity and continuity of your business operations while adapting to the ever-evolving landscape of cyber threats. This makes it an essential asset for organizations striving to maintain secure and efficient communication.