Best PBX Software in Japan

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!

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.

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.

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.

FreePBX stands as Freedom. FreePBX is the most popular open-source IP PBX in the world. It allows users to create a phone system that suits their needs. Open source communications is a powerful tool. Start building your phone system by taking a look at the many built-in features FreePBX offers. You can transform your FreePBX installation into a robust Unified Communications System with commercial modules from Sangoma or other add-ons. Visit the Knowledge Base at the FreePBX Wiki! FreePBX experts can help you with troubleshooting, installing, updating, and more. FreePBX is completely free to download and to use. Its power comes from a global network of developers who ensure that it remains a high-compatibility and customizable platform with all key features required to build a scalable business telephone system on any budget.

Asterisk is an open-source framework for building communication applications. Asterisk transforms an ordinary computer into a communications host. Asterisk powers IPPBX systems, VoIP gateways and conference servers, as well as other custom solutions. It can be used by small and large businesses, call centres, carriers, government agencies, and other agencies worldwide. Asterisk is open-source and free. Sangoma sponsors Asterisk. Today, more than 1 million Asterisk-based communication systems are in use in more than 170 nations. Nearly all Fortune 1000 customers use Asterisk. Asterisk is most commonly used by system integrators or developers. It can be used to create a complete business telephone system, enhance an existing system, or bridge between systems.

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.

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.

MikoPBX is a fully-featured phone system that offers high performance, stability, and rich functionality. MikoPBX can be installed on-premise (hardware install) or in any cloud service (Google Cloud, Amazon Cloud Solution and Microsoft Azure, for example). MikoPBX is free and provides basic business functionality. The system features a simple web interface that allows for call recording, conference calls and voice mail, voice mail pickup, call transfers and pick up, call queues, interactive voice response menus, and inbound and outside call rules. There are also holiday and night-time call switchers. MikoPBX can be extended with plugins and modules by developers from around the globe. MikoPBX is free and you can add-ons to enhance the capabilities of your PBX. MikoPBX does NOT license each user. You can expand your company without incurring additional costs for call recordings or new employees.

OpenSIPS is an open source SIP proxy/server that supports voice, video, IM and presence. OpenSIPS is a multifunctional, multi-purpose signaling server that can be used by telecoms and ITSPs to provide solutions such as Class4/5 residential platforms. It also supports enterprise/virtual VoIP solutions, session border controllers. Application servers, front-end loadbalers, IMS platforms, call centres, and other solutions. OpenSIPS is recommended for all types of SIP scenarios/services due to its high throughput, tens or thousands of CPS, millions simultaneous calls (see official testing), flexibility of routing, integration, routing scripts to implement custom routing logics, and several interfacing APIs. The application building process is efficient, with more than 120 modules providing features for SIP handling, backend operations and integration. OpenSIPS 3.2 was subject to a security audit. This made everything safer and more secure.

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.

Taridium ipbx, a highly-capable software VoIP PBX system, replaces a traditional hardware-based PBX. ipbx can be run on standard server hardware or in the cloud. It does not require additional software licenses. An End User Panel allows you to manage all aspects of your extension. This includes listening to voicemail, setting up call forwarding, creating ad-hoc phone conferences, and more. You can set up your extension to forward only during certain times using your iCalendar feed, or based on a list favorite caller IDs. A conference room that is meet-me-type for all users. No more scheduling conflicts. Live conference management allows you to mute or kick participants. You can invite users by clicking a button, and the system will call them, instead of asking for cumbersome login instructions.

Dial-Office is an IP PBX software that offers flexible and feature-rich functionality for start-ups and SMBs who want to reap the benefits of VoIP technology. This IP PBX software solution is based on the SIP standard platform and offers advanced phone features that were only available with expensive circuit-switched enterprise telephone systems. Dial-Office IP PBX software provides reliable, low-cost unified communications for small- and medium-sized businesses using an established Local Area Network. This award-winning VoIP solution offers end-users crystal-clear voice quality, a variety of business management features, and the ability to lower communication costs and improve customer service. The following features are available on the platform:

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.

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.

FusionPBX can be used to provide a highly-available single or domain-based multi-tenant VoIP, carrier-grade switch and call center server. It can also be used as a call center server. FreeSWITCH™, a multi-threaded, multiplatform communication platform, is highly scalable. It offers unlimited extensions, voicemail to email, music on hold and call parking. Call queues, phone provisioning, call center, call parking, and call center are all available. It offers the functionality you need and provides corporate-level phone system features for small, medium, and large businesses. It took a lot of time to fix bugs, tighten security, and improve the code. The future release is being worked on. We have made significant progress, and the master branch is now far ahead of the current release in many aspects. We are diligently working to complete the last changes required for this release.

It is easy to install and configure PBX software that is fully supported with our telephony hardware. It is ideal for small to medium-sized businesses (SMBs) who require a more sophisticated phone system than those found in larger companies. However, it does not come with a high price tag. OpenPBX combines the efficiency and ease of use of the desktop computer with the simplicity of a telephone handset to give you advanced features. You can also control your communication system from anywhere. Voicetronix offers line interface hardware that can connect to either standard POTS lines or primary rate (E1/T1) trunklines. OpenPBX's Do-it-Yourself (DIY), option allows customers to create their own phone system with commodity PC servers running Linux or analog telephone handsets. You can set up and manage your own phone system without the assistance of a technician. You can navigate to your web browser and select voicemail messages.

Everything you need to know about the FreeSWITCH platform. FreeSWITCH, a Software Defined Telecom Stack, enables the digital transformation of proprietary telecom switches into a flexible software implementation that runs on all commodity hardware. FreeSWITCH is able to unlock the telecoms potential of any device, from a Raspberry Pi to a multi-core Server. FreeSWITCH, when combined with SignalWire, a hosted cloud platform, can connect with the outside world and scale up to any size. The pre-built virtual machine includes both a written as well as a video how-to. You only need a copy virtualbox to get started. You can also follow these instructions if you are a skilled developer to compile the FreeSWITCH platform directly from source. If you have a spare Debian instance, you can follow this link to find detailed instructions on how to install our packages.

sipXcom was founded in January 2015 by the sipXecs development team at eZuce, Inc. sipXecs, SIPfoundry, and SIPfoundry had become a user community, and were not seeing any growth in their developer community. sipXcom has no restrictive contributor agreement. sipXcom source code licensing follows the copyleft-friendly AGLP version 3 (Affero General Public License). SWARM is the codename for the next generation sipXcom/sipXecs project. SWARM is currently in development and production readiness is expected for early 2017. It will be a microservices-based architecture which will improve sipX's scalability reliability, configurability, and configurationability. The software can be used with any compute platform, whether it is dedicated, virtual, cloud-based, or hybrid. You can also create hybrid instances that combine premise-based, data center, or cloud instances.

Kamailio®, the successor to former OpenSER and SE, is an open-source SIP Server released under GPLv2+. It can handle thousands of call setups per seconds. Kamailio is a platform for VoIP, real-time communications and instant messaging. It can also be used to scale up SIP-to PBX gateways, PBX system, or media servers such as Asterisk™, FREESWITCH™, and SEMS. Among its many powerful features are asynchronous TCP and UDP, secure communication via VoIP (voice, text, and video); WebSocket support WebRTC; IPv4 or IPv6; simple instant messaging with embedded XCAP servers and MSRP relay; asynchronous operation; IMS extensions to VoLTE; ENUM, DID and least-cost routing; load balancing and routing failover; accounting, authentication, and authorization; support for many backend system such as MySQL, Postgres and Oracle, Redis, Cassandra and Memcached, Redis, Redis, Redis, Redis, Redis, Redis, Redis, Redis, Redis, Redis, Redisk, MongoDB and MongoDB, Redis, Redis, Redis, Redis, Redis, Redis, Redis, Redis, Redis, Redist, Redis, Redis, Redis, Casandra, Casandra, Redis, Casandra, MongoDB and Memcached, MongoDB, MongoDB, MongoDB, MongoDB, MongoDB, MongoDB, Casandra, MongoDB and Memcached, Casandra, MongoDB and Memcached

Enterprise interaction is a mashup between collaboration and communication. Secure enterprise interaction is possible from your private or hybrid cloud. Universal Cloud Communications Stack is a revolutionary concept that leverages big data architectures to enable real-time communications and collaboration. It allows for elastic scaling of transaction-oriented services at a global scale. The immediate benefits include rapid deployment and automated management that leverages standard infrastructure. Secure, scalable, and virtualized containers enable hosted and managed communications and collaboration services with ease. This allows MSPs to create new business models and allows enterprises to significantly lower their costs. Communications offer the greatest potential for cost savings across all IT. Combining modern software architecture and data-driven distributed infrastructure can transform an industry.

Avaya IP Office, a globally available and award-winning unified communications platform, is implemented and managed in your premises. Explore all the IP Office capabilities to improve your interactions with customers and employees. Set up call recording, tracking and reporting to study your customer calls. This Avaya call reporting capability can be delivered via IP Office to record and documentation incoming calls. You can also add dashboards and seats for real-time agents in call centers to improve call handling. Avaya IP Office and Avaya Spaces are combined to create a single app that includes voice, video, messaging and conferencing. This app is available on any device and in any location.

With a mobility solution that is feature-rich and has the features your customers need, they can now work from anywhere. With built-in redundancy, we can help you save money by providing instant provisioning for SIP trunks and DIDs. Pulsar360 Corporation Voice over IP (VoIP), solutions deliver high-quality voice communication over affordable broadband services from other providers or Pulsar360 managed carrier services. By combining their phone service with high-speed internet, customers can save money and increase their productivity. We have spent more than 10 years perfecting our VoIP platform and created one of North America's most sophisticated, automated systems. This innovation allows us to offer instant provisioning for SIP trunks and hosted VoIP PBXs. We can also provide local and international DIDs and toll-free number, as well advanced management capabilities.

R.E. Communications offers an Asterisk-based IP-PBX system. These systems can support all industry-standard SIP phones and offer enterprise level features, which allow for up to 4000 phones. These systems also support SIP Trunking, which allows you to replicate the services of a Hosted solution while saving on per-seat costs by using SIP trunks and SIP phone. A typical 10 phone/seat system can support 4 to 6 SIP trunks, which is a significant savings over the 10 seats cost of a Hosted solution.