Best PBX Software in the USA

Why work with giants like Twilio where your support requests will take forever to get a response? Signalmash is not just a CPaaS provider, we take pride in providing the best possible CPaaS support. Your developers can have a shared Slack channel with our developers for instant support. You can speak to our CEO anytime. Real-time and top-notch support means shorter development cycles and enables you to provide exceptional support to your own customers. SMS services we offer: SMS API SMS CPaaS SMS UCaaS SMS no-code sending platform 10DLC campaign support Short code SMS Toll-free SMS Voice services we offer: Contact center telecoms Voice termination Voice orgination Local numbers Short code numbers Toll-free numbers SIP Trunking Our No-code telecoms solutions: UCaaS to send SMS CCaaS to manage phone calls AI solutions Signalmash - experience the highest level of support, every step of the way - book a call with us to get expert advice!

Thirdlane is a scalable UCaaS platform that blends multi-tenant PBX, Contact Center, and collaboration apps into one solution. MSPs, Enterprises, and telecom providers adopt Thirdlane when they want freedom of deployment, deep integrations, and reliable control over their communications stack. Whether your team prefers cloud-hosted services or on-premises deployments, Thirdlane adapts to your strategy. IT departments appreciate the API-first design, while business leaders value the ability to cut costs and avoid lock-in. Core benefits: - Enterprise-grade PBX with calling, video meetings, voicemail, and advanced routing. - Unified Thirdlane Connect apps for chat, video meetings, and file sharing. - Built-in Contact Center - Advanced analytics to improve service and sales. - CRM integration for context-rich customer interactions. - White-label branding across all user touchpoints. - Options for geo-redundancy and high availability.

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.

Issabel is an open source unified communications platform that consolidates IP communication services into a single system, surpassing the constraints of a conventional PBX by integrating features such as call center capabilities, email, management reports, fax, video call meetings, audio recording, and CRM and platform integrations. It offers a robust, flexible, and easy-to-use call center module designed to efficiently automate and manage contact centers, handling both incoming and outgoing call campaigns through an intuitive agent console and call management interface. Issabel streamlines PBX solutions for service providers, enabling the management of multiple clients from a single instance, simplifying administration without the need for separate servers for each client. To enhance functionality, Issabel supports add-ons crafted to seamlessly integrate with the platform, adding features that interact with telephony to improve customer service and automate actions for support.

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

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.

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.

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.

Now your customers can truly work from anywhere with a feature-rich mobility solution with the mission-critical features they need. We enable you to save money with built-in redundancy and instant provisioning of SIP trunks, DIDs, and toll-free lines. Pulsar360 Corporation's Voice over IP (VoIP) solutions provide high-quality voice communications, which can be delivered over affordable broadband service from other providers and/or Pulsar360-managed carrier services. Customers save money and increase productivity by streamlining their communications into a single network connection for phone service and high-speed Internet. Having invested more than ten years in perfecting our VoIP platform, we have created one of the most sophisticated, highly-automated systems in North America. With this innovation, we can offer instant provisioning of SIP trunks, hosted and hybrid VoIP PBXs, local and international DIDs, toll-free numbers, as well as advanced management capabilities.