Best Self Hosted Alternatives to Bicom Systems PBXware

FreeSWITCH is an open-source, software-defined telecom stack used to build and run voice, video, and real-time communications infrastructure on commodity hardware. It is commonly used as a softswitch and media server for SIP-based telephony and can also provide browser-based calling via WebRTC.

Key Features

• SIP signaling and media handling suitable for softswitch and PBX-style deployments
• Native WebRTC capabilities for browser-based real-time communications
• Modular architecture with loadable modules for telephony features and integrations
• Built-in conferencing and real-time media services (including mixing and related functions)
• Common voice features such as call routing, IVR building blocks, call recording, and voicemail support

Use Cases

• Build and operate a SIP softswitch or VoIP application server for carriers and enterprises
• Implement IVR, call routing, and contact-center style call flows
• Provide WebRTC calling to web applications without requiring a separate gateway

Limitations and Considerations

• Configuration and operation can be complex for newcomers due to the breadth of telecom concepts and module options
• Certain advanced capabilities may be available only via commercial modules or enterprise distributions

FreeSWITCH is a mature foundation for telecom and real-time media systems, from embedded devices to large-scale deployments. Its modular design and protocol support make it a flexible core for custom telephony platforms and communication products.

Asterisk is an open-source telephony engine and PBX toolkit implemented primarily in C and developed for GNU/Linux. It exposes traditional PBX features and low-level telephony primitives so developers and operators can build SIP, WebRTC and PSTN-connected communications applications.

Key Features

• Modular, channel-based architecture with pluggable modules for SIP (chan_pjsip / chan_sip), media, codecs and hardware interfaces
• WebRTC support (WSS/DTLS-SRTP), RTP/RTCP handling, and modern codec support including Opus for browser and realtime audio
• ARI (Asterisk REST Interface) exposing REST + WebSocket events for building custom programmable call applications
• AMI and AGI interfaces for management, automation and traditional dialplan scripting; full CLI and menuselect build configuration
• PSTN and telephony hardware integration (traditional telephony cards and drivers) alongside VoIP gateway capability
• Source-driven build system using autoconf/Autotools and GNU Make; extensive documentation, community forum and release advisories

Use Cases

• Deploying an enterprise or branch PBX providing calls, voicemail, conferencing, queues and call routing
• Acting as a VoIP gateway or SBC to bridge SIP/WebRTC clients with PSTN trunks and telephony hardware
• Building programmable communications services (IVR, voicebots, conferencing, call recording) using ARI or AMI

Limitations and Considerations

• Nontrivial operational complexity: requires careful configuration, dependency management and familiarity with telephony concepts
• Requires proactive security and performance tuning (file-descriptor limits, TLS/DTLS configuration); security advisories are periodically published for critical fixes
• Feature surface is large and modularity means some functionality requires enabling/building specific modules or external libraries

Asterisk is a mature, widely adopted telephony engine suited for operators and developers who need deep control over call handling and media. It is maintained by a large community and is intended for production PBX and programmable-telephony deployments.

Kamailio is an open source SIP signaling server used to build and scale VoIP and real-time communication platforms. It focuses on high performance and flexibility, and is commonly deployed as a SIP proxy, registrar, edge proxy, or signaling core in carrier and enterprise environments.

Key Features

• High-throughput SIP routing and proxying suitable for large deployments
• SIP load balancing, failover routing, and least-cost routing capabilities
• Registrar and location service for user registration and contact lookup
• Security and access control features, including authentication/authorization and TLS
• WebSocket support for WebRTC signaling
• Extensible modular architecture with integrations for multiple backend systems
• Support for IPv4 and IPv6 and various SIP transport options (UDP/TCP and more)

Use Cases

• Scaling SIP PBX, SIP-to-PSTN gateways, or media servers with a dedicated signaling layer
• Building carrier-grade SIP routing cores, including IMS/VoLTE-oriented deployments
• Deploying SIP edge proxy and SIP firewall-style protection in front of RTC infrastructure

Limitations and Considerations

• Provides SIP signaling (call control) and typically requires external media servers for RTP/media handling
• Configuration is powerful but can be complex, especially for advanced routing and multi-module setups

Kamailio is a mature SIP server with continuous development since 2001 and a large ecosystem. It is well-suited for operators and organizations that need a robust, customizable signaling component for VoIP, WebRTC, and unified communications.

Routr is a lightweight, cloud-ready SIP server that acts as a proxy, registrar, and location service for modern VoIP networks. It is designed to be programmable and scalable, making it suitable for carriers, communication service providers, and integrators.

Key Features

• Core SIP server functions: proxy, registrar, and location service
• Programmable routing with configurable strategies (intra-domain, ingress/egress, peer egress)
• Load balancing and session affinity for upstream media servers (for example Asterisk or FreeSWITCH)
• Multi-tenant and multi-domain support with domain-level access control lists
• Multiple transports: UDP, TCP, TLS, WS, and WSS
• Pluggable processors and middleware for extending routing and cross-cutting concerns
• Server management via APIs and tooling (CLI and gRPC management API)
• Flexible data sources, including Redis-backed location service and PostgreSQL

Use Cases

• Run a scalable SIP edge/proxy in front of one or more PBXs or media servers
• Build programmable SIP routing for multi-tenant VoIP platforms
• Deploy carrier-grade SIP infrastructure on Docker or Kubernetes

Routr provides a modern approach to SIP infrastructure by combining core SIP capabilities with APIs, extensibility, and cloud-native deployment patterns. It is a strong fit when you need a configurable SIP control plane that can integrate cleanly into automation and platform workflows.

FusionPBX is a full-featured, domain-based multi-tenant PBX and voice switch built around FreeSWITCH. It provides a web-based interface to manage telephony features for businesses, service providers, and multi-customer environments.

Key Features

• Domain-based multi-tenancy for hosting multiple independent PBX tenants
• Extension and user management with typical PBX calling features
• IVR, ring groups, call parking, and call routing via dialplan tools
• Call queues / ACD and call center-oriented applications
• Voicemail features including voicemail-to-email support
• Device provisioning with troubleshooting logs for provisioning requests
• Call detail records (CDR) and call recordings management
• High availability and redundancy options for more resilient deployments

Use Cases

• Hosting a multi-tenant PBX platform for customers or departments
• Building a VoIP business phone system with IVR, voicemail, and queues
• Running a FreeSWITCH-based softswitch with web-managed configuration

Limitations and Considerations

• Requires deploying and operating FreeSWITCH alongside the FusionPBX web application
• Some advanced features (for example, certain reporting modules and the REST API) may depend on membership or add-on applications

FusionPBX is well-suited for organizations that want a FreeSWITCH-based telephony stack with a comprehensive administrative UI and multi-tenant capabilities. It scales from single-instance PBX deployments to carrier-oriented environments when designed with redundancy in mind.

FreePBX is an open-source, web-based graphical user interface designed to configure and manage Asterisk telephony servers. It provides a modular platform of built-in features and extensible modules to build IP PBX, UC, and call-centre systems.

Key Features

• Web-based administrative GUI for configuring Asterisk dialplans, trunks, extensions, IVRs, queues, voicemail and CDR reporting
• Modular architecture with many open-source modules and an add-on marketplace for commercial extensions (provisioning, call center features, CRM integration)
• User Control Panel (UCP) for end-user voicemail, call handling, web softphone and customizable widgets
• Zero-touch phone provisioning and endpoint management for supported IP phones
• SIP trunking and provisioning integrations, session border controller (SBC) support, and tools for analog/PRI gateway integration
• REST/API hooks and token-based access for automation and third-party integration
• Built for common Linux stacks with emphasis on PHP/JavaScript modules and standard LAMP-style components

Use Cases

• Small-to-medium businesses deploying a full-featured IP PBX with IVR, voicemail, and ring groups
• Contact centers and help desks using queueing, CDR reporting, and commercial call-center modules
• Integrators and service providers packaging custom modules, provisioning endpoints, and managing SIP trunking for customers

Limitations and Considerations

• Core FreePBX is open source but many advanced or enterprise modules are commercial and sold through the add-on marketplace
• Relies on Asterisk as the telephony engine; feature set and behavior depend on Asterisk versions and underlying Linux distribution

FreePBX is a mature, widely used platform for building customizable telephony systems. Its modular design and large ecosystem make it suitable for many business telecom deployments, while advanced features may require paid modules or vendor support.

Yeti-Switch is a carrier-grade VoIP softswitch focused on routing and controlling SIP traffic for telecom operators, wholesale voice providers, and enterprises. It combines signaling control, routing policy, and real-time call processing to build scalable voice platforms.

Key Features

• SIP call routing with policy controls (e.g., LCR-style routing, prefixes, routing groups)
• Multi-tenant/accounting-oriented concepts (customers/vendors, trunks, rate plans)
• Real-time call control and session handling suitable for SBC-style deployments
• Number/prefix management and routing rules for large dial-plan environments
• Call detail records (CDRs) and billing-oriented data model for rating workflows
• Operational tooling for managing gateways/trunks, dial-peers, and routing policies
• Designed for scalable deployments (separation of signaling, routing logic, and data)

Use Cases

• Wholesale voice routing platform for multiple vendors/customers
• Enterprise SIP interconnect/SBC layer to control ingress/egress traffic
• Voice termination/origination service with rating/billing data export

Limitations and Considerations

• Best suited for telecom/VoIP operators; requires SIP/telephony expertise to deploy and tune
• Billing/rating typically integrates with external processes/systems depending on operator workflow

Yeti-Switch is a good fit when you need a robust, operator-focused softswitch with strong routing primitives and telecom-centric entities (trunks, dial-peers, rates, CDRs). It is commonly used as a foundation for scalable SIP routing and interconnection networks where policy and cost-based routing are core requirements.