Best PBX Software in India

3CX is a software-based, open standards IP PBX that offers complete Unified Communications, out of the box. Suitable for any business size or industry 3CX can accommodate every need; from mobility and status to advanced contact center features and more, at a fraction of the cost. Ideal for companies working remotely, 3CX includes mobile apps, video conferencing and website live chat. Installation, management and maintenance of the PBX is so easy that you can effortlessly manage it yourself, whether on an appliance or server at your premise on Windows, Linux or in the cloud. Alternatively you can choose to have it hosted by 3CX for a completely hassle-free solution.

PBXware is the world’s first and most mature IP PBX Professional Open Standards Turnkey Telephony Platform. Since 2004, PBXware has deployed flexible, reliable, and scalable Next Generation Communication Systems and VoIP solutions to Small and Medium sized Businesses (SMBs), enterprises, Internet Telephony Service Providers (ITSPs), Call Centers and governments worldwide by combining the most advanced of the latest technologies. Bicom Systems’ softswitch is offered in the Business, Call Center and Multi-Tenant Editions, each supporting specific features maximizing performance, reliability, and expandability.

Thirdlane delivers a versatile platform and flexible applications tailored to managed and cloud service providers seeking competitive differentiation. Their robust yet affordable communications tools allow both small businesses and global enterprises to streamline interactions and enhance customer experience without large upfront costs. The Thirdlane communications platform is a one-stop-shop solution for MSPs, UCaaS providers, and resellers. Its advanced voice, collaboration, and video conferencing tools are designed to be fully customizable and seamlessly integrated with partners’ own branding. Main products: Thirdlane Multi Tenant PBX, Thirdlane Business Phone System and he Thirdlane Connect which provides users with chat, voice and video calls and conferencing, SMS, screen sharing, CRM integration in modern web browsers, on mobile phones, Windows, Linux and Mac desktops., these solutions empower businesses to boost productivity, drive sales & strengthen customer relationships.

VitalPBX is a unified communications PBX system based on Asterisk and Linux (Centos 7). VitalPBX provides a robust and scalable platform, which will allow you to manage your PBX in an easy and intuitive way. VitalPBX is built to be easy to use with tooltips across the system and a coherent user interface throughout. A telephony system should be aware of security hazards and avoid them. VitalPBX offers various levels of protection against cyber threats in the VoIP world.

The “Free” in FreePBX stands for Freedom. That’s because FreePBX, the world’s most popular open source IP PBX, gives users the tools to build a phone system tailored to their needs. Discover the power of open source communications. Take a look at all the built-in features FreePBX has to offer and start building your phone system. Transform your FreePBX install into a robust Unified Communications system with commercial modules from Sangoma and other add-ons for FreePBX. Visit the Knowledge Base on the FreePBX Wiki! Or let FreePBX experts assist with support for installing FreePBX, updating, troubleshooting, and beyond. Completely free to download and use, the power of FreePBX comes from a global community of developers who ensure it remains a high compatibility and customizable platform with all the key features needed to build a scalable business phone system on any budget.

Asterisk is an open source framework for building communications applications. Asterisk turns an ordinary computer into a communications server. Asterisk powers IP PBX systems, VoIP gateways, conference servers and other custom solutions. It is used by small businesses, large businesses, call centers, carriers and government agencies, worldwide. Asterisk is free and open source. Asterisk is sponsored by Sangoma. Today, there are more than one million Asterisk-based communications systems in use, in more than 170 countries. Asterisk is used by almost the entire Fortune 1000 list of customers. Most often deployed by system integrators and developers, Asterisk can become the basis for a complete business phone system, or used to enhance or extend an existing system, or to bridge a gap between systems. Build your own custom system with Asterisk? Buy a powerful, low-cost turnkey system based on Asterisk? Discover which option is right for you.

HoduPBX is the ultimate solution for business communication. Its unprecedented user interface with optimal user experience facilitates businesses to communicate with the end users efficiently. This software offers world-class connectivity within an office and from the remote location leaves communication an uninterrupted process. Get HoduPBX and empower agents’ work management with multitasking abilities and efficiencies. With 1000+ concurrent calls and 10,000 extensions, our IP PBX software is one of the best choices for business communications. Here are HoduPBX’s key features: Multi-Tenant Auto Provisioning Least Call Routing (LCR) Real-time Analytics and Reports Queue Monitoring Multi-level IVR BLF & Presence Prepaid & Postpaid Billing Payment Gateway Integration Audio-conferencing SMS Hosted IP PBX Software

Virtual IP PBX (Private Branch Exchange) or Phone System Appliance – CompletePBX is the heart of Xorcom hardware and software phone systems. Built on top of the solid Linux and Asterisk platforms, it’s built for stability and performance. CompletePBX was designed from the ground up to combine ease of use, advanced features and security at every level. Download free, fully functional CompletePBX 5 VM – evaluation virtual machine now! Check out our FREE online certification training course that will take you step by step through all the basics of CompletePBX system. Save a lot of time and effort by trying to go at it alone. Let us help you get there faster.

MikoPBX is a full-featured phone solution with a top performance level, high stability, and rich functionality. You can install MikoPBX on-premise (hardware installation) or in any type of cloud service (Google cloud, Amazon Cloud Solution, Microsoft Azure, etc.) MikoPBX is a free phone system, all basic business functionality is absolutely free. The system has an easy and convenient web interface, call recording, conference calls, voice mail, call transfers and pickup, call queues, an interactive voice response(IVR) menu, and inbound and outbound call rules. Also, you can find holidays and night-time calls switchers there. MikoPBX has the ability to extend functionality with plugins and modules from developers around the world. MikoPBX is completely free, you can use add-ons that extend the capabilities of your PBX. MikoPBX does not license each user. Expand your company without additional costs for new employees or call recordings.

OpenSIPS is an open-source SIP proxy/server for voice, video, IM, presence, and any other SIP extensions. OpenSIPS is a multi-functional, multi-purpose signaling SIP server used by carriers, telecoms or ITSPs for solutions like Class4/5 residential platforms, trunking/wholesale, enterprise/virtual PBX solutions, session border controllers, application servers, front-end load balancers, IMS platforms, call centers, and many others. OpenSIPS is recommended for any kind of SIP scenario/service by the high throughput, tens of thousands of CPS, millions of ‏simultaneous calls (see official tests), the flexibility of routing and integration, routing scripts for implementing custom routing logics, several interfacing APIs. The effective application building, more than 120 modules to provide features, for SIP handling, backend operations, integration, the routing logic. OpenSIPS 3.2 release was the subject of a security audit, making everything safer and more secure for everybody.

Discover Signalmash, the ultimate Communication Platform as a Service (CPaaS) solution, offering a versatile suite of communication APIs designed to cater to the evolving needs of businesses. Our offerings encompass: SMS API: Unlock enhanced customer interaction capabilities through a range of text-based communication functions, including alerts, authentication, promotional outreach, surveys, appointment scheduling, and nurturing customer connections. Flexible SIP Trunking: Simplify the establishment of communication channels with customizable SIP trunking solutions, empowering businesses with tailored contact center setups, personalized call routing options, and encryption for secure data transmission. VoIP PBX Solutions: Optimize your internal communication infrastructure with Signalmash's VoIP PBX solutions, streamlining the setup and administration of private branch exchange systems, featuring intuitive call routing, voicemail management, conference calling and more.

Vodia is a software PBX available on Windows, Linux, and offering all the advanced features of a full-featured PBX including calling queues, auto attendants, call forwarding, parking, and many others. Vodia is also the pioneer of Multi-Tenancy where you can host Vodia on a dedicated VM ( Virtual Machine) in the cloud or premises and maintaining multiple FQDN for each individual customer. All MSPs and ITSPs can receive a 60 day hosted demo license code at signup

*astTECS provides complete enterprise telecom products based on Open Source Technology- such as IP PBX and Contact Center Solution. *astTECS IP PBX offers all the high-end features at an affordable price enabling your business to be future-ready, flexible, innovative and competitive while reducing your telecom expenses. *astTECS IP PBX can seamlessly integrate with any applications - CRM, ERP, PMS and even with your existing telecom infrastructure. *astTECS call center solution has advanced features with a real-time agent monitoring dashboard helps to improve your agent productivity and customer engagement with advanced analytics. *astTECS also offers a complete call recording solution for your business with real time dashboards.

The Taridium ipbx is a highly capable software VoIP PBX system that replaces a traditional hardware based PBX. ipbx runs on standard server hardware, or in the cloud and does not require any additional software licenses. An End User Panel that let’s you control all aspects of your own extension, such as managing your call via an operator console, listen to voicemail, configure call forwarding, create ad-hoc telephone conferences and much more. Configure your extension to only forward during certain times, using your iCalendar feed or based on a list of favourite caller IDs. A meet-me type conference room for each and every user. No more scheduling conflicts and live conference management, which allows you to mute and kick participants. Invite users with the click of a button and have the system call them instead of having to provide cumbersome login instructions.

Dial-Office is a flexible, feature-rich IP PBX software designed for SMBs and start-ups who want to take full advantage of the business benefits of VoIP technology. Based on the SIP standard platform, this IP PBX software solution comes with a number of advanced phone features that were previously available only with costly circuit-switched enterprise telephone systems. Dial-Office IP PBX software delivers reliable, low-cost unified communications to small- & mid-sized business using an established Local Area Network. This award-winning VoIP solution provides end-users with crystal-clear voice quality and a series of business management features designed to increase productivity, lower communication costs & enhance customer service. Platform features include the following:

Available as hardware and software, Yeastar K2 IP PBX supports 2000 users, 500 concurrent calls and 400 VoIP trunks across locations, providing medium and large enterprises with full flexibility and capability. Engineered for the communications needs of today and tomorrow, Yeastar K2 IP PBX delivers enterprise-level features with exceptional reliability, and enables extraordinary remote working agility with next-level Linkus UC Softphone.

More than just a PBX, Yeastar P-Series PBX System is a converged system purposed-built for SMEs that have higher expectations on business communications. As a “PBX-Plus-More” solution, the P-Series unify voice, video, applications, collaboration, and more together, helping SMEs to enable extra productivity, fill the UC gap, and accomplish more sophisticated tasks.

FusionPBX can be used as a highly available single or domain-based multi-tenant PBX, carrier-grade switch, call center server, fax server, voip server, voicemail server, conference server, voice application server, appliance framework and more. FreeSWITCH™ is a highly scalable, multi-threaded, multi-platform communication platform. It provides unlimited extensions, voicemail-to-email, music on hold, call parking, call center, call queues, phone provisioning and many other features. It provides the functionality your business needs and brings corporate-level phone system features to small, medium and large businesses. A large amount of time has been used to fix bugs and tighten security and improve the code. There is also progress being made on the future release. We have made a good amount of progress and the master branch is far ahead of the current release in many ways. We are working diligently to finish the last changes that we feel are needed for this coming release.

It is an easy-to-install and configure PBX software platform that is fully supported by our telephony hardware. It is suitable for small to medium businesses (SMB) that require a sophisticated phone system usually found in larger enterprises but without the price tag! OpenPBX harnesses the ease of use and efficiency of the desktop PC and combines it with the plain old telephone handset to give you advanced features and the ultimate control and use of your communication system. Whether you need to connect to primary rate (E1/T1) trunk lines or to standard POTS lines, Voicetronix has suitable line interface hardware. Our OpenPBX Do-it-Yourself (DIY) option offers customers the ability to build their own phone system using commodity PC servers running Linux and analog telephone handsets. Configure and maintain your own phone system, eliminating the need for a technician. Use your web browser to navigate, select and listen to voicemail messages.

Everything you need to know to dive into the FreeSWITCH platform. FreeSWITCH is a Software Defined Telecom Stack enabling the digital transformation from proprietary telecom switches to a versatile software implementation that runs on any commodity hardware. From a Raspberry PI to a multi-core server, FreeSWITCH can unlock the telecommunications potential of any device. Combined with our hosted cloud platform, SignalWire, FreeSWITCH can interconnect with the outside world and scale to any size. The Pre-built virtual machine has both a written and video how-to to help you get started. All you need is a copy of virtualbox and you can be off and running in minutes. If you are an experienced developer you can also follow the instructions below to compile the FreeSWITCH platform from source among other manual installation options. If you already have a spare Debian instance handy, you can follow the link below to get detailed instructions on how to get our packages installed.

sipXcom was established in January of 2015 from a fork in the sipXecs project by the development team at eZuce, Inc. sipXecs and SIPfoundry in particular had become a user community and was not seeing any growth in the developer community. Due to a restrictive contributor agreement, sipXcom is without any restrictive contribution rights. sipXcom source code licensing is based on the copyleft-friendly AGLP v3 (Affero General Public License). SWARM is the code name for the next generation of sipXcom/sipXecs projects. Under development with anticipated production readiness scheduled for early 2017, SWARM will be a microservices-based architecture that will improve upon sipX scalability, reliability and configurability. The software is designed to support any compute platform whether you want to use dedicated, virtual, or cloud-based servers. You can also create hybrid implementations with a combination of premise-based and data center or cloud instances.

Kamailio® (successor of former OpenSER and SER) is an open-source SIP server released under GPLv2+, able to handle thousands of call setups per second. Kamailio can be used to build large platforms for VoIP and real-time communications, presence, WebRTC, Instant messaging, and other applications. Moreover, it can be easily used for scaling up SIP-to-PSTN gateways, PBX systems, or media servers like Asterisk™, FreeSWITCH™, or SEMS. Among the powerful features, are asynchronous TCP, UDP and SCTP, secure communication via TLS for VoIP (voice, video, text); WebSocket support for WebRTC; IPv4 and IPv6; simple instant messaging and presence with embedded XCAP server and MSRP relay; asynchronous operations; IMS extensions for VoLTE; ENUM; DID and least cost routing; load balancing; routing fail-over; accounting, authentication and authorization; support for many backend systems such as MySQL, Postgres, Oracle, Radius, LDAP, Redis, Cassandra, MongoDB, Memcached.

We define enterprise interaction as the mashup of collaboration and communication. We enable secure enterprise interaction from your own private or hybrid cloud. Universal Cloud Communications Stack (UCCS) is a revolutionary concept to leverage big data architectures for real-time communications and collaboration, resulting in the elastic scalability of transaction-oriented services on a global scale. Rapid deployment and automated management leveraging standard infrastructure are key immediate benefits. Virtualized, secure, and scalable containers deliver hosted and managed communications and collaboration services with ease, enabling new business models for MSPs, and allowing enterprises to significantly reduce costs. Communications represent the single largest potential for cost savings in all of IT. Combining modern software architecture with data-driven distributed infrastructure creates a potent mix to transform an industry.

Our award-winning, globally available Avaya IP Office delivers a full-featured unified communications platform implemented and managed on your premises. To keep improving both your customer and employee interactions and experiences, explore all your IP Office-ready capabilities. Study your customer calls by setting up call recording, tracking, and reporting. This Avaya Call Reporting capability can be delivered through IP Office to record and document incoming calls. You can also add real-time call center agent seats and dashboards for more efficient call handling. Combine Avaya IP Office with a subscription for Avaya Spaces and you gain a single app for voice, video, messaging, conferencing, and a calendar—all available on any device, in any location.

R.E. Communications offers an Asterisk based IP-PBX system. These systems support all industry standard SIP phones, with enterprise level features that go from 1 phone to 4000 phones. These systems also support SIP trunking which with the combination of SIP trunks and SIP phones you can replicate the services of a Hosted solution but save on the per seat cost. Typically a 10 phone/seat system can be supported with 4 to 6 SIP trunks which is a savings over the 10 seat cost of a Hosted solution.