Open Source VoIP and PBX Phone Systems
In This Guide
What Is Open Source VoIP?
Voice over Internet Protocol, commonly known as VoIP, transmits voice calls as digital packets over IP networks instead of traditional copper telephone lines. When the underlying software is released under an open source license, anyone can inspect the source code, modify it, and redistribute it freely. This stands in sharp contrast to proprietary systems from vendors like Cisco, Avaya, or Mitel, where the code is closed and the hardware is locked to specific firmware.
Open source VoIP encompasses a wide range of software. At the lowest level, telephony engines like Asterisk and FreeSWITCH handle the actual protocol negotiation, codec processing, and call routing. On top of these engines, graphical management interfaces like FreePBX and FusionPBX provide web-based dashboards that make it possible for administrators to configure extensions, ring groups, IVR trees, and voicemail without writing configuration files by hand. Beyond PBX systems, the open source telephony ecosystem also includes SIP proxy servers like Kamailio and OpenSIPS, softphone clients like Ooh323c and Ooh323, and specialized applications for call center queuing, conferencing, and fax-over-IP.
The appeal of open source VoIP comes down to three things: cost, control, and flexibility. There are no per-seat licensing fees, no annual support contracts that you are forced to renew, and no vendor lock-in that prevents you from moving your infrastructure to different hardware. If a feature does not exist, you can build it. If a bug appears, you can fix it or report it to a community that responds in days rather than fiscal quarters.
How Open Source PBX Systems Work
A Private Branch Exchange, or PBX, is the system that manages internal telephone calls within an organization and connects them to the outside telephone network. Traditional PBX hardware was expensive, proprietary, and required specialized technicians to install and maintain. An open source PBX replaces all of that with software running on commodity hardware or a virtual machine.
The core of any IP-based PBX is the Session Initiation Protocol, known as SIP. SIP is the signaling standard that establishes, maintains, and terminates voice sessions between endpoints. When a user picks up a SIP phone or opens a softphone application and dials a number, the PBX receives the SIP INVITE message, determines where to route the call based on its dial plan, and bridges the audio streams between the caller and the destination. The actual voice data travels over the Real-time Transport Protocol (RTP), typically using codecs like G.711 for high quality uncompressed audio or Opus and G.729 for bandwidth-efficient compressed audio.
Modern open source PBX systems handle far more than basic call routing. A typical installation includes an interactive voice response system that greets callers with menus, a voicemail server that records messages and optionally transcribes them, call queues that distribute incoming calls across teams of agents, conference bridges that allow multiple participants to join a single call, call recording for compliance or training purposes, and detailed call detail records that log every call for billing and analytics. Many systems also support WebRTC, which enables browser-based calling without any plugins or desktop software.
The dial plan is the central logic engine of a PBX. In Asterisk, this is defined in configuration files using a pattern-matching syntax that routes calls based on the dialed number, the time of day, the caller ID, or any other variable. In FreeSWITCH, the equivalent is the XML dial plan, which uses regular expressions and conditional logic to achieve the same result. Both approaches are powerful, but they require different skills to master. FreePBX and FusionPBX abstract much of this complexity behind graphical interfaces, allowing administrators to build call flows by clicking through web forms rather than editing text files.
The Leading Open Source VoIP Platforms
Asterisk
Asterisk is the project that started the open source telephony revolution. Created by Mark Spencer in 1999 and later maintained by Digium (now part of Sangoma Technologies), Asterisk has been in continuous development for over 25 years. It is written in C for performance and runs on Linux, with community packages available for most major distributions. As of 2026, Asterisk 22 is the current long-term support release, and Asterisk 23 is the latest standard release with the newest features.
Asterisk functions as a telephony toolkit rather than a ready-to-use phone system. It supports SIP, IAX2 (Inter-Asterisk eXchange), H.323, and MGCP protocols, along with dozens of audio codecs including G.711, G.722, G.729, Opus, and Speex. It can interface with traditional PSTN lines through DAHDI channel drivers and analog or digital telephony cards, or it can connect purely through SIP trunks to cloud-based carriers. The system provides built-in applications for voicemail, conferencing (via ConfBridge), call queuing, call parking, music on hold, interactive voice response, and much more.
The trade-off with Asterisk is complexity. Without a GUI, every aspect of the system is configured through text files: sip.conf or pjsip.conf for SIP endpoints, extensions.conf for the dial plan, voicemail.conf for mailbox settings, and many others. This gives experienced administrators extraordinary control, but it also creates a steep learning curve for newcomers. For this reason, most production Asterisk deployments use a management interface like FreePBX on top of the raw engine.
FreePBX
FreePBX is the most widely deployed open source PBX management interface in the world. It wraps Asterisk in a comprehensive web-based GUI that covers virtually every feature the underlying engine supports. Administrators can create extensions, configure ring groups, build IVR menus, set up time-based routing, manage voicemail boxes, and monitor active calls, all through a browser. FreePBX is maintained by Sangoma Technologies, which also distributes FreePBX Distro, a complete Linux distribution with Asterisk and FreePBX pre-installed and ready to go.
The FreePBX module system is one of its greatest strengths. The base installation covers the essentials, and additional functionality can be added through free or commercial modules. Free modules include call recording, follow-me routing, phonebook management, and system administration tools. Commercial modules, available through Sangoma, add features like endpoint management for provisioning SIP phones automatically, high availability for failover configurations, and advanced reporting dashboards. This modular approach lets administrators start simple and expand as their needs grow.
FreePBX is the right choice for small to medium businesses that want a full-featured phone system without deep Linux or Asterisk expertise. The web interface handles the vast majority of daily administration tasks, and the underlying Asterisk configuration is still accessible for advanced customization when needed.
FusionPBX and FreeSWITCH
FusionPBX is to FreeSWITCH what FreePBX is to Asterisk: a web-based management interface built on top of a powerful telephony engine. FreeSWITCH was created by Anthony Minessale, a former Asterisk developer, and it takes a fundamentally different architectural approach. Where Asterisk evolved from a monolithic design, FreeSWITCH was built from the ground up with a modular, threaded architecture that separates the core from its components. This makes FreeSWITCH particularly well-suited for high-volume, carrier-grade deployments where thousands of concurrent calls need to be handled reliably.
The defining feature of FusionPBX is its native multi-tenant architecture. A single FusionPBX server can host dozens or hundreds of separate organizations, each with its own extensions, dial plans, voicemail, and configuration, all completely isolated from one another. This makes it the preferred platform for VoIP service providers, managed service companies, and any organization that needs to serve multiple independent customers from shared infrastructure. FreePBX, by contrast, is designed primarily for single-tenant deployments, where one organization runs one PBX instance.
FreeSWITCH also has superior WebRTC support compared to Asterisk, making it a strong foundation for browser-based calling applications and modern unified communications platforms. Its event socket interface provides a clean API for external applications to control calls programmatically, which is valuable for custom integrations and call center software. FreeSWITCH is licensed under the Mozilla Public License, which offers more flexibility than Asterisk's GPL for developers building proprietary products on top of the engine.
Kamailio and OpenSIPS
Kamailio and OpenSIPS are not PBX systems in the traditional sense. They are SIP proxy servers designed to handle SIP signaling at massive scale. While Asterisk and FreeSWITCH process both the signaling and the media (the actual voice data), Kamailio and OpenSIPS only handle the signaling layer. They route SIP messages, perform load balancing across multiple media servers, handle user registration and authentication, and apply routing logic based on complex rules.
In large-scale VoIP deployments, Kamailio or OpenSIPS typically sits in front of a cluster of Asterisk or FreeSWITCH servers, distributing incoming calls across the cluster and providing a single point of entry for all SIP traffic. This architecture is common among VoIP carriers, wholesale providers, and enterprise contact centers that need to handle tens of thousands of concurrent calls. Kamailio can process over 5,000 SIP transactions per second on modest hardware, making it one of the most efficient SIP processors available.
Kamailio and OpenSIPS share a common ancestor (they forked from the same SER project in 2005 and 2008 respectively) and offer similar capabilities, though their communities and development philosophies have diverged over the years. Both are written in C, both support extensive scripting through their routing languages, and both integrate with databases like MySQL, PostgreSQL, and Redis for subscriber data and call routing tables.
Other Notable Projects
Several other open source VoIP projects deserve mention. Wazo (formerly XiVO) is a full unified communications platform built on Asterisk that includes a REST API, WebRTC support, and a modern web interface. It is designed for companies that want a complete communications stack rather than just a phone system. VitalPBX is another Asterisk-based PBX that offers a polished, user-friendly interface with a free community edition supporting up to a certain number of extensions. Ooh323 and various SIP client libraries provide building blocks for developers creating custom telephony applications.
Asterisk vs FreeSWITCH: Choosing an Engine
The choice between Asterisk and FreeSWITCH is one of the most consequential decisions in any open source VoIP deployment, because the engine determines the capabilities, scalability characteristics, and administrative workflow of everything built on top of it.
Asterisk excels at traditional PBX functionality for small to medium installations. Its feature set is comprehensive out of the box, its community is enormous (with decades of forum posts, tutorials, and documentation available), and its ecosystem of compatible hardware, SIP phones, and management tools is unmatched. If you are building a phone system for a single office with 10 to 500 extensions, Asterisk with FreePBX is the most well-trodden path and the one with the most community support.
FreeSWITCH is the stronger choice when scalability, multi-tenancy, or modern protocols are priorities. Its threaded architecture handles high call volumes more gracefully than Asterisk's single-threaded event loop, and its native support for WebRTC makes it the better foundation for browser-based calling and video conferencing. FreeSWITCH also separates its configuration into XML files organized by context, which some administrators find more logical than Asterisk's collection of .conf files, though others find the XML verbosity harder to work with.
From a development perspective, Asterisk uses AGI (Asterisk Gateway Interface) and AMI (Asterisk Manager Interface) for external integrations, while FreeSWITCH provides the Event Socket Layer (ESL) and a comprehensive set of APIs. Both approaches are powerful, but ESL is generally considered more modern and easier to work with from programming languages like Python, Node.js, and Go. Asterisk's ARI (Asterisk REST Interface), introduced in later versions, has closed some of this gap by providing a RESTful API for call control.
Licensing is another consideration. Asterisk's GPL license requires that derivative works also be released under the GPL, which can be restrictive for companies building commercial products. FreeSWITCH's Mozilla Public License allows more flexibility, permitting proprietary modules to be developed and distributed alongside the open source core without requiring the proprietary code to be open sourced.
Self-Hosting vs Cloud VoIP
One of the first decisions any organization faces is whether to host their VoIP system on their own infrastructure or use a cloud-based service. Both approaches have legitimate advantages, and the right choice depends on the organization's technical capabilities, compliance requirements, and budget structure.
Self-hosted open source VoIP gives you complete ownership of your communications infrastructure. Your call data never leaves your network (unless you choose to route it externally), which is critical for organizations in healthcare, finance, legal, and government sectors where call recordings and metadata may be subject to regulatory requirements like HIPAA, PCI DSS, or GDPR. You also eliminate recurring per-user monthly fees, because the software is free and you only pay for the hardware, internet connectivity, and SIP trunking services that carry your calls to the public telephone network.
The cost structure of self-hosting favors organizations that can amortize the initial setup effort across many users and months of operation. A FreePBX server running on a dedicated machine or a virtual private server can support a 50-person office for the cost of the SIP trunk (typically a few cents per minute for outbound calls and a flat monthly fee per DID number for inbound), compared to cloud PBX services that charge $15 to $35 per user per month. Over a year, the savings for a 50-person office can easily exceed $10,000.
Cloud-hosted VoIP services, on the other hand, eliminate the need for in-house telephony expertise. Providers like RingCentral, 8x8, and Vonage handle all the infrastructure, updates, and maintenance. You get a web portal, preconfigured phones, and a support line to call when something goes wrong. For organizations without dedicated IT staff, this convenience is worth the monthly premium. Some open source projects, including Wazo and FreePBX, also offer commercially hosted versions that combine the open source software with managed infrastructure.
A hybrid approach is increasingly common, where organizations run their PBX on a cloud server (such as an AWS EC2 instance, a DigitalOcean droplet, or a Vultr VPS) rather than on-premises hardware. This provides the control and cost benefits of self-hosting while eliminating the need to maintain physical equipment, deal with power outages, or worry about local network failures affecting phone service.
Open Source VoIP for Business
Open source VoIP is no longer a hobbyist project or a cost-cutting measure for startups. Organizations of all sizes, from five-person offices to enterprises with thousands of employees, deploy Asterisk and FreeSWITCH-based systems in production. The key to a successful business deployment is understanding which features matter for your use case and choosing the right combination of software and SIP provider.
For a typical small business, FreePBX provides everything needed out of the box: auto-attendant menus that greet callers and route them to the right department, ring groups that ring multiple phones simultaneously or sequentially, voicemail with email notification, call recording for quality assurance, and detailed call logs for tracking usage. Setting up a basic system takes a few hours, and the FreePBX community forums provide extensive guidance for common configurations.
Medium businesses and contact centers often need more advanced features like skills-based routing (directing calls to agents based on their expertise), queue callbacks (letting callers request a callback instead of waiting on hold), CRM integration (popping customer records on the agent's screen when a call arrives), and real-time wallboard displays that show queue statistics. These features are available through FreePBX commercial modules, custom Asterisk dialplan development, or dedicated contact center distributions like VICIdial, which is an open source predictive dialer and contact center suite built on Asterisk.
Enterprise deployments typically involve multiple PBX instances connected across different office locations, redundant configurations for high availability, and integration with existing directory services like LDAP or Active Directory for user provisioning. Kamailio or OpenSIPS may be deployed as a SIP proxy to load-balance across multiple Asterisk or FreeSWITCH instances, providing carrier-grade reliability with automatic failover.
SIP Trunking and Connectivity
A SIP trunk is the connection between your PBX and the public telephone network. It replaces the traditional copper lines or ISDN circuits that legacy phone systems used to make and receive external calls. SIP trunking providers deliver voice traffic over the internet, converting between VoIP protocols on your end and the PSTN (Public Switched Telephone Network) on the other end.
Choosing a SIP trunk provider is one of the most important decisions in a VoIP deployment. Key factors include pricing structure (per-minute billing vs. unlimited plans), geographic coverage (local and toll-free number availability), call quality (the provider's network peering and codec support), reliability (uptime guarantees and redundant infrastructure), and technical compatibility (support for your PBX's SIP implementation and codec preferences).
Popular SIP trunk providers that work well with open source PBX systems include Twilio, Telnyx, VoIP.ms, Flowroute (now part of Intrado), and BulkVS. Many of these offer pay-as-you-go pricing starting at less than one cent per minute for outbound calls, with DID numbers (the phone numbers people dial to reach you) available for one to two dollars per month. For businesses that make a high volume of calls, providers like SIPStation (Sangoma's own trunking service) offer unlimited domestic calling plans bundled with FreePBX support.
When configuring SIP trunks, you will need to set up outbound routes (which trunk to use for which dialed numbers), inbound routes (where incoming calls on each DID should be directed), and codec negotiation (ensuring your PBX and the provider agree on audio encoding). Most providers offer configuration guides specific to FreePBX and Asterisk, and the setup typically involves entering the provider's SIP server address, your authentication credentials, and your DID numbers into the PBX interface.
VoIP Security and Encryption
VoIP systems face the same security threats as any networked service, plus some unique challenges related to the real-time nature of voice communications. The most common attacks include brute-force registration attempts (attackers trying to register rogue SIP endpoints using guessed passwords), toll fraud (unauthorized use of your SIP trunks to make expensive international calls), eavesdropping on unencrypted voice streams, and denial-of-service attacks aimed at disrupting phone service.
Securing an open source PBX starts with basic network hygiene. The system should sit behind a properly configured firewall that restricts SIP and RTP traffic to known IP ranges. Fail2ban, the intrusion prevention framework, should be configured to monitor SIP registration logs and automatically block IP addresses that generate repeated failed authentication attempts. Asterisk's built-in security features include ACLs (Access Control Lists) that restrict which IP addresses can register or make calls, and the PJSIP channel driver supports TLS for encrypting SIP signaling.
For encrypting the actual voice data, SRTP (Secure Real-time Transport Protocol) is the standard. Both Asterisk and FreeSWITCH support SRTP, though the configuration details differ between the two. When both TLS (for signaling) and SRTP (for media) are enabled, the entire call is encrypted end-to-end between the PBX and the SIP endpoint. This is essential for organizations handling sensitive information and is required by several regulatory frameworks.
Additional security measures include using strong, unique passwords for every SIP extension, disabling unused protocols and modules, keeping the PBX software up to date with security patches, and monitoring call detail records for unusual patterns that might indicate fraud. Running the PBX on a dedicated VLAN separate from other network traffic adds another layer of isolation, and session border controllers (SBCs) can be deployed at the network edge to normalize SIP traffic and protect the PBX from malformed or malicious packets.
Getting Started with Open Source VoIP
The fastest path to a working open source phone system is to download the FreePBX Distro ISO, install it on a dedicated server or virtual machine, and follow the web-based setup wizard. This gives you a complete, production-ready PBX in under an hour, with Asterisk, FreePBX, and a hardened CentOS-based operating system all preconfigured. From there, you can add extensions, configure a SIP trunk, and start making calls.
For those who prefer FusionPBX and FreeSWITCH, the FusionPBX project provides installation scripts for Debian and Ubuntu that automate the entire setup process. The result is a multi-tenant capable PBX with a web interface, ready for configuration. FusionPBX's documentation covers the initial setup, domain creation, and extension configuration in detail.
If you want to learn the fundamentals before committing to a full deployment, consider setting up a test environment first. A virtual machine with 1 GB of RAM and a single CPU core is sufficient for a development PBX that handles a handful of test calls. Pair it with a free softphone application like Ooh323c, Ooh323, or any standard SIP client on your desktop, and you can explore the system's capabilities without any cost or risk to a production environment.
Regardless of which platform you choose, invest time in understanding the underlying protocols. Knowing how SIP works, what RTP is, how codecs affect call quality and bandwidth, and how NAT traversal challenges are solved will make you a more effective administrator and a better troubleshooter when issues arise. The open source VoIP community is active and welcoming, with forums, IRC channels, and mailing lists where experienced users help newcomers get their systems running.