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.
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SessionPBX
SessionPBX is a hosted business phone system for small teams and service providers. Administrators manage users, extensions, and calling through a web portal, while team members use companion softphone apps on Windows, macOS, iOS, and Android. The service supports internal calling, voice and video, messaging, presence, voicemail, call forwarding, and SIP trunk connections to a carrier chosen and paid directly by the customer. Teams Business adds call flows, business-hours routing, IVR menus, ring groups, call queues, call recording, supervision, and reports. QR codes and invitation links help provision team apps. SessionTalk hosts the standard service; approved partners can also deploy multi-tenant SessionPBX on their own servers and manage customer workspaces.
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ICTPBX
ICTPBX is an open source, carrier-grade PBX platform built for ISPs, ITSPs, and service providers. It combines FreeSWITCH 1.10 as the media engine with FusionPBX 6.6 for PBX configuration, unified through the ICTCore REST API and managed from a modern Angular 13 web dashboard.
Unlike hosted PBX solutions, ICTPBX runs entirely on your own infrastructure. You control your data, your tenants, and your billing. Scale from a handful of extensions to thousands of seats without per-user licensing costs.
It’s built for multi-tenant deployment. Each tenant gets their own isolated PBX environment with custom branding, quota limits, and role-based access control.
Core capabilities:
SIP extensions, ring groups, call queues (ACD), and IVR menus
Voicemail, conferences, time conditions, and call flows
SIP trunks (gateways), inbound DID routing, call blocking
Multi-tenant with per-tenant branding and quota management
Voice, fax (T.38/FoIP), SMS (SMPP), and email on one platform
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Kamailio
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.
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