Cloud Telephony is one of the modern phone systems that easily run through your internet connections, enabling you to move your business phone service to the cloud through effective business communication. It will run on a variety of operating systems or hardware and provides a GUI for QR code softphone provisioning, unlimited extensions, voicemail-to-email, and music on hold, call parking, analog lines, or high-density circuits. The interesting function of this platform is that it offers HD audio or video call at economical rates, multimedia sharing and geolocation, enhanced address book, easily integrated with any third-party system for making off-net calls or SMS, two-factor authentication method, and many more.
Cloud Telephony permits effective centralized management of a distributed system and offers plug & play configuration for all phones and gateways. It automatically captures all the call data, ensuring your customer relationship management is always up to date, and also nullifies the time-consuming manual updates in seconds.
Cloud Telephony Alternatives
Asterisk is one of the classical software that acts as an open-source framework for making seamless communication applications and offers various toolkits like IP PBX systems, VoIP gateways, conference servers, and others. It aids the multinational companies to easily deploy the open-source solutions companies that frequently need training and meet all the necessary needs that newly launched project demands. It offers various products like IP PBX, IP phones, VoIP Gateways, SIP trunking, Telephony cards, IVR voice prompts, and much other attractive software.
Through its IP PBX solution, it lets an engine that efficiently handles all the low-level details, maintaining and manipulating calls between the endpoints. It deals with other characteristics like unified communication capabilities, advanced voicemail messaging, instant messaging, call control, multi-party conferencing and audience auto-attendant IVR and others. With the help of its switchboard, users can quickly perform various functions like click-to-dial, click-to-transfer, move calls to parking lots and monitor the status of co-workers.
Wazo IP Telephony is an open-source platform that is designed to build your own IP telecom platform and provides IP telecopy, mobility. Collaboration and customer experience solutions to SMBs with the help of a vast network of certified partners. It deals with multiple solutions like enhance your ROI, virtualize your workspace, phone provisioning, call recording and call history, audio conferencing, and many others. It facilities you to maximize your ROI by reducing all of your communication expenses and lets you make calls with your international team without any extra deduction. Through its advance analyzing tool, it empowers you to get overall calling stats on a daily, weekly, or monthly basis in the form of attractive graphs or colorful charts.
Wazo IP Telephony has the ability to make smooth workflow and apply different rig strategies between manager and executive assistants. With the help of its audio conferencing, it persuades you to make you dedicated conference rooms, enabling you to simplify the method you engage with your team and precious customers.
FreeSWITCH is one of the powerful, defined telecom stacks that enable the digital transformation from any telecom switches for the implementation of the versatile software that runs on the commodity hardware. It facilities you to unlock the telecommunication potential of any device and offers a cloud-hosted platform, making you connect with the world without any hassle. It deals with multiple attractive products like enterprise support, cloud-hosted APIs, and many more.
FreeSWITCH provides full support for SIP compliance across all transport like UDP, TCP, SCTP, and TLS. It acts as a gateway to SIP applications, legacy systems, and public switched telephone networks. For both mid and small-sized enterprises, it contains other stunning functions like text speech, voice mail, conferencing, and automatic speech recognition. With the help of its integrated video, MCU provides customizable video layouts, video superimposing, screen sharing, and mixed protocol.
FusionPBX is a software that utilizes single or multi-tenant PBX, carrier-grade switch, call center service, fax service, and VoIP server, providing the functionality to your business needs and bring the high phone system features to both small or large-sized organizations. It will run on a variety of operating systems or hardware and provides a GUI for QR code softphone provisioning, unlimited extensions, voicemail-to-email, and music on hold, call parking, analog lines, or high-density circuits.
The main benefits of this platform include, add extra functionality to the Free Switch VoIP platform, entitle to monitor or work directly on Free Switch with a command-line interface. FusionPBX covers high-class security functions, making you protect the data of your users in an appropriate manner. It loads calls into the queues so they can be answered in the specific order as they came into the queue.
Kamailio is an open-source SIP server that has the ability to handle millions of call setups per second and is used to build a large number of platforms for VoIP and effective real-time communication in the presence of WebRTC, instant messaging, and other applications. Through its small footprint or play & plug module interface, it has the ability to add new extensions without changing the core and assuring the powerful stability of every core component. The hot function of this platform includes being fully compatible with transport layer gateways like IPv4, IPv6, UDP, TLS, and others.
Kamailio has a specific function of asynchronous processing that contains TCP handling, SIP message processing, and inter-process message queues communication system, distributed message queue, TLS support for SIP signaling, TLS domain name extension support, internal DNS caching system, and many more. Other exciting functions are stateless and transactional SIP Proxy processing, SCTP multi-homing and multi-streaming, and web socket for WebRTC and others.
RingCentral Engage Digital is a software that provides you an opportunity to improve customer service and maximize customer satisfaction. It covers various attractive services like identify the valuable data that works for you, finds the right agent at the right time, special media support, a Cloud phone system, video conferencing, team messaging, and many more. Through its smart service providing ability, it lets customized and integrated mobile operators, cable/MSOs, and managed service providers, enabling you to accelerate the growth of your business in no time.
RingCentral Engage Digital entitles you to view data from every angle and inform your customer service strategy with the help of powerful dashboards that are based upon artificial intelligence. It allows the agents to ask for approval when responding to inquiries or transfer the conversations to another agent.
miniSIPServer is one of the professional software that easily run on the Windows, Linux, and Raspberry Pi or even other kinds of virtual machine system like Hyper-V, VMWare, KVM, Xen, Virtual box and other, providing an opportunity to the users to manage or deploy their VoIP systems through GUI interface. The program is created with full support for a wide range of popular SIP hardware phones and softphones, making you eliminate the vendor lock-in devices. The service aids you to easily make and receive calls via cellphone network by using the VoIP gateways.
miniSIPServer helps the admins to deploy series of rich services that are necessary for the businesses like voice mail, ring groups, and others. It also enables seamless communication between the employees or with the visitors without making any configuration changes. Moreover, it is fully supported for Call Detail Record in both numeric and graphical representation, black system list, and STUN.
XiVO is an open-source enterprise VoIP communication solution that is designed for IP telephony and unified communication. Through its optimized customer relationship, it fulfills the customer requirement and improves advisor productivity. The main functions of this platform include personalized reporting, listening & recording of the conversations, modification of distribution scripts, monitors the advisor performance, tracking the customer route, personal indicators or data and metrics, and many others.
For customers, it covers various solutions like positive user experience, customer router optimization, dynamic callback option, classification of the request, and many more. Other functions are to visualize the customer route in order to anticipate requests, display detailed statistics in the form of attractive graphs or colorful charts.
Yate – Yet Another Telephony Engine has the ability to instantly connect with the YateBTS over the GTP & SIP, replacing the costly SS7 MAP-A interface, making you get a high return in investment without affecting the reliability of the network of provided service.
VoipSwitch is a rich-featured VoIP solution that is fully compatible with almost all the SIP network elements, billing, APIs, and web protocol. It deals with multiple attractive services like unified communication PBX, rich communication suite, calling cards platform, advance analytical engine, and many others. The main advantage of this platform is that it offers HD audio or video call at economical rates, multimedia sharing and geolocation, enhanced address book, easily integrated with any third-party system for making off-net calls or SMS, two-factor authentication method, and many more.
Through its call-shop feature, it facilitates its customers to make international calls at any landline number at a lost cost that is not present in another traditional platform. Other hot functions are detailed call history records in the form of sections as daily, weekly or monthly, multiple languages supported, rate plans management, visual representation of the call in real-time.
sipXecs is one of the leading open-source enterprise communication systems that lets high available SIP routing core integrated with a growing suite of communication services through unified web-based management applications. It empowers traditional PBX telephony services that are integrated with instant messaging and covers different advanced tools for video calling. The basic characteristics of this platform are that it has a media server for unified messaging and auto attendant IVR services, XMPP instant messaging and Openfire based server, detailed call records in the form of attractive graphs or colorful charts as daily, weekly, or monthly calling, process management server for the centralized cluster management and many more.
sipXecs permits effective centralized management of a distributed system and offers plug & play configuration for all phones and gateways. Other features are multiple calls per line, configurations of individual speed dial soft keys, auto-generation of directory information, and many more.
sipXcom is an open-source unified communication software that is specially designed for the small, mid, or large-sized enterprises for making unified communications and offers various attractive voice solutions like lost cots, voice calls, chats with friends from all over the world, auto-attendant IVR service, and more. The noticeable function of this platform is that it is fully supported with any compute platform whether you want to use dedicated, virtual, or cloud-based servers and aids you to create or apply hybrid implementations with the combination of data centers or cloud instances.
sipXcom has the ability that it is compatible with standard-based SIP desktops phones, soft clients, gateways, SBCs and runs in an IP-based switched network infrastructure. It works in multiple progressive steps like create a DNS record of your domain, launch the sipXcom, configure the server core, telephony, and device services, add users, configure soft or hard phones, add SIP trunking for incoming and outgoing telephone calls.
Thirdlane Multi Tenant PBX is one of the classical software that offers unmatched PBX services and deals with modern communication like VoIP telephony, private & group messaging, video & voice conferencing, salesforce, contact Center, and many others. The interesting function of this platform is that it offers a separate solution for MSPs & service providers, and business customers, making you easily roll out hosted telephony services and decrease the costs by configuring and authorizing self-service features for administrators and users.
Through its efficient customer telephony integration, it facilities you to leverage investments in existing assets and aid you to choose phones or other devices at the best value rate. With the help of its modern multi-tenant PBX or unified communication function, it ensures the service providers stay competitive in the rapidly changing marketplace and entitles hosted PBX or seamless communication to their business customers along with cloud services.
Natterbox is software that covers cloud-based voice CTI solutions fully embedded in salesforce and allows automated data capturing technology, advanced analytical tools, and one-touch communication. It prioritizes the right contact and automatically makes calls in the salesforce, enabling you to accelerate the continuous improvement across your team. Through its advanced analytical tools or in-depth monitoring technology, you can view your customer from different angles and deliver high-class customer service without using any third-party resource.
Natterbox covers one of the efficient dashboards that identify the various complex errors in the salesforce and improve the satisfaction level in no time. It covers different products like voice for salesforce, Naterbox mobile, calls recordings, speech analytics, and many more. It automatically captures all the call data, ensuring your customer relationship management is always up to date, and also nullifies the time-consuming manual updates in seconds.
Ozeki Phone Systems is one of the reliable software that quickly transforms your emulator into a communication server, and provides you an opportunity to make calls all over the world through your softphone. The main feature of this platform is that you can view your registered phone line and active phone calls without any complex configurations or hardware. It facilitates you to select the codec that is used during the call and select the Ozeki VoIP SIP SDK to your respective Ozeki Phone System XE.
Ozeki Phone System will display all the characteristics on the main screen of your computers like total call duration, call rate per minute, total cost, recipient phone number along with its name, remaining balance, and many others. It works in different non-complex steps like register the Ozeki Phone System XE; the call will be forwarded to the selected phone system by using the created extension and notifies you like an alert when the destination phone will be ringing.
Mizutech VoIP Server is an easy-to-use software that is specially designed for the small-sized industry having simultaneous calling ability on a single instance with endless scaling capability. It covers different attractive services like broadband VoIP services, retail business, wholesale termination, SBC, calling card servers like LCR, BRS, Load phone to phone, conferencing, and many more. With the help of its VoIP hosting, it aids you to considerably reduce your hardware and maintenance costs and eliminate all the technical issues, making you focus on your business.
Mizutech VoIP Server handles all the VoIP technical details and provides all the efficient infrastructure for successful VoIP business, such as managed cloud servers, soft switches, and customized softphones. Other functions of this platform include built-in VoIP tunneling and encryption, send your traffic to any carriers, trunks, VoIP call termination providers, unlimited backup space, and preconfigured automatic backup.
Vodia Networks is one of the trusted software that quickly turns your phone, laptops, tablets into a telephone system, allowing your employees to easily use it for seamless communication. It deals with various options for handing calls between multiple parties such as extensions, mailbox, auto-attendant, Groups, call center, conferences, and many more. Through its mailbox, you can receive FAX messages and send them to the email address of users. All the messages can be escalated to managers when the user is unable to retrieve them in time.
With the help of its auto attendant, the caller can use their keyboard to enter extension number or search company directory by the name and enables the different ways to redirect the incoming calls, based on the time of the call. The stunning function of this platform is that when the respective message is delivered to the receipt, it notifies the sender in the form of an alert that is not present in the traditional telephony software
Incredible PBX is a unified communication VoIP platform that comes with advanced support for SIP and IAX. The spectacular unified communication solution is paving the way for the business courtesy of leading standard components, setting out the perfect journey, and starting the preceding of your first call in just a matter of a few minutes. This utility provides you complete cloud support, so you can deploy server machines in order to run SIP VoIP service in an agile way. Incredible PBX’s versions can be deployed in a variety of platforms on-premise hardware or on virtual machine platforms VirtualBox, VMware, and proxmox.
This superior utility features multiple components that include Apache web server, SendMail, MariaDB, Fail2Ban, NeoRouter, and more to add. One important landmark of this VoIP software is that you have the ability to set out an AvantFax email address that will do nicely for your faxes when it comes to their delivery. There is also a possibility to configure to your mail that will allow you to use your Gmail account in order to send an email with the help of your server. Once you are done with the installation process, you have to run the following commands to initialize your server.
Flexisip is one of the leading SIP server suites that come with a scalable and modular way to streamline your business. This program consists of various group chat functions and proxies, and more likely, it offers all the modern solutions that will make your deployment of SIP service more accurate and professional. This thing will be crucial to out-of-the-box mobile and desktop applications. There is also a possibility to integrate SIP infrastructure for various purposes
Noting too important than the cloud support, that will, in turn, be valuable for deploying server on the server machine in order to run SIP VoIP service in an agile way. Flexisip is creating a difference with its modular architecture and its lesser number of tendencies, so this will allow you to run Flexisip on even small hardware efficiently. The key features of this unified communication platform are transport protocols, cluster mode support, external authentication, NAT awareness, push notification system, real-time statistics, TLS client certificate authentication, and more to add.
VitalPBX is an all-in-one functional platform that comes with a unified platform, providing businesses leverage of the unified communications PBX system. This platform is based on Asterisk and Linux and is providing you with complete integration with the Microsoft Teams software. It’s a completely free telephone and communications system for organizations around the globe and has the ability to install on the site or as a hosted application.
This intelligent utility comes with a graphic interface that will be crucial in streamlining your business communication in a modern way. The platform is scalable and extensible for your needs, allowing you to manage PBX in an intuitive way. VitalPBX also takes into account all the security concerns courtesy of the advanced level of protection against Cyber-attacks. There are multiple features on offer that include three-way calling, video calling, and call recording access, audio or video conferencing, routing, advanced dialing, strength indicator, and more to add.
Freepbx is in one web-based IP PBX management tool, providing best-in-class tools in order a build a phone system to streamline their business needs. The software comes with top-notch modules that are valuable for transforming your PBX system into an integrated communication system. The software comes with spectacular built-in features, allowing you to build an extravagant phone system. The software is leveraging business with smart office access, providing complete cloud-based access control. This will be crucial for employees to be on top of their process right from the functional smartphone application.
Freepbx turns into a more solid foundation with its SIP Trunking, providing call centers a seamless approach to connect with SIP station, which in turn paves the way for reliability, quality, and most importantly, auto-provisioning. Now you can add U functionality to FreePBX with things like Phone applications, call center packages, paging, and much more. Certified appliances, video conferencing, VoIP security, hosted FreePBX, complete access control, PRI connectivity, and more are some of its reliable features.
Issabel is an all-in-one functional communication platform that comes with a unified solution to streamline your business activities with the ultimate real-time support. This open-source and centralized platform brings IP communication services to one unified place. There is no headache that will be faced by you because of the stable processing, allowing you to reach your potential goals. This platform provides you certain benefits like no user limitation, ability to share desktop, share desktop, integrated chat, adding documents, and ease of accessing of conference right from the web browser.
It’s a platform where you have the professional management of the business interaction channels via seamless VoIP with CRM, fax, recordings, email, and more to add. Issabel is designed to make your configuration, project deployment, and estimations right regardless of the location that you are in. Get leverage of the sumptuous call center module, providing a most robust and flexible solution, so you are always on top with the efficient management of the contact center.
Acefone is created with full support for a wide range of popular SIP hardware phones and softphones, making you eliminate the vendor lock-in devices. It also enables seamless communication between the employees or with the visitors without making any configuration changes. It allows the agents to ask for approval when responding to inquiries or transfer the conversations to another agent.
The spectacular unified communication solution is based on Asterisk and Linux servers. This utility provides consistent support with a high-performance turnkey PBX that is easy to manage and gets the extravagant functionality of PBX via having an advanced Debian Linux. This centralized communication platform comprises desktop clients that are valid for both windows and smartphones clients, including iOS and Android.
BXinaflash is easy to install and takes not much time to set up, and more likely, there is an automatic configuration system for both trunk and gateways. The most admirable thing about this communication system is its seamless management and auto-upgrade, so no more pain for users. In this system, users will also have the support for video conferring, and besides this, they have the flexibility in mind to run this program in the cloud.