Yeastar P-Series PBX System
Focusing on delivering "Easy-first Unified Communications", Yeastar P-Series Phone System offers companies of all sizes with a complete package for calls, video, messaging, and integrations, out of the box.
With in-built visual call management, integrated video conferencing, advanced contact center features, and ready-made SMS, WhatsApp, Microsoft Teams, CRMs, and more platform integrations, P-Series boosts productivity at all levels and provides everything across desktop, mobile, and browser with simple user apps.
Available in the Appliance, Software, and Cloud Editions, P-Series provides flexible deployment options, allowing you to have it sited on-premises or in the cloud. Balancing costs and future growth, it requires a lower total cost of ownership, less training, and fewer management efforts. The ease of use and future-proof adaptability are paramount.
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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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VaxVoIP
VaxVoIP SIP Server SDK enables developers to build secure, SIP-based VoIP solutions such as IP-PBX, IVR, auto-dialers, smart agents, call centers, and calling card systems. Fully compatible with the SIP protocol, it supports softphones, hardphones, ATA devices, and other SIP hardware. It includes PBX features like call queues, call forwarding, call recording, voicemail, and call routing. With advanced security features, it protects against SIP floods, scans, and brute-force attacks. The SDK also offers real-time access to audio PCM data, enabling integration of AI, machine learning, and smart agents into VoIP systems for intelligent and enhanced communication.
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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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