The Top 10 Best Free Open

Source PBX Software

While adopting an existing Hosted PBX service from one of the top hosted
PBX providers will certainly get the job done for the vast majority of businesses,
from small to enterprise level, the shoe is not necessarily one size fits all. Of
course, providers do their best to tailor services and packages to truly fit the
needs of each business on a case-by-case basis. However, with differences in use
cases, industries, team sizes, required features or security, it can be almost
impossible to dot every “i” and cross every “t.”
For those super specific cases where a cookie-cutter service just won’t fill in the
gaps, businesses can turn to an open source platform, and yes there are open
source PBX software solutions available. With the right team of developers and IT,
any business can develop and tailor their own in house PBX software, or even
Unified Communications suite. Because these platforms are open source, all of
the source coding is available for free online, and can be totally custom tailored
for each specific scenario.

Asterisk
Asterisk is essentially the grand-daddy of all open source VoIP and PBX solutions,
and continues to operate as the gold standard .  As the leading open source
telephony platform, and a massive feature lists that only continues to grow every
year, the Asterisk tool kit is utilized by not only a mass amount of setups around
the world, many of the providers on our list have either started with, or are based
fully off of the Asterisk engine.
Packed with the standard PBX features including an interactive voice response
menu, automatic call distribution, conference calling, and the usual voicemail,
Asterisk makes it possible to turn any computer into a communications server.
Supported by Digium, the software is completely free and open source. To help
get you started, Asterisk supplies live web classes, online training courses, and
even an Asterisk Definitive Guide manual from O’Reilly Press. Asterisk can almost
be considered the go-to platform when it comes to developing your own VoIP,
PBX or UC system.

SIP Foundry
Often considered one of the main competitors to the Asterisk platform,
SIPFoundry was established in 2004 and offers much of the same solutions that
the Asterisk engine can power. SIPFoundry makes it possible to build your own
voice and video communications, as well as conferencing, unified messaging, IM
and chat with presence indications, and even a mobile client. Just like with
Asterisk, the platform includes everything you need to build your own Unified
Communications solution.

However, while Asterisk is 100% free and open source, SIPFoundry has a slightly
different spin, and sells professional support to users based on different rates,
starting at $495 per month for 100 users, up to 20,000 users to be charged on a
case-by-case basis. While an extra cost to consider, adopting a dedicated support
team might be a necessary step for some businesses looking to build their own
system.

Elastix
Originally based off the Asterisk platform, Elastix offers open source unified
communications server software including an IP PBX, email, IM, faxing and even
collaboration functionality. With a strong focus on the entrepreneurial market, the
tool is completely free for commercial or personal use. The project has also
brought in features from other open source projects including FreePBX, HylaFAX,
Openfire and Postfix to round out all of the UC offerings available.
Overall, Elastix aims to bring in the greatest features of Asterisk and other
projects, all under one easy-to-use interface. Elastix boasts support for a wide
range of hardware including Digium, Dinstar, Yeastar, Yealink and Snom. Elastix
was actually one of the first distributions that included a call center module with a
predictive dialer, and continues to offer the robust solution, again free under the
GNU General Public License.
Since writing this post, it has come to our attention that Elastix only seems to
offer up to 8 free SIM calls for about 25 users.

FreeSWITCH
FreeSWITCH was also originally based off the Asterisk platform, and was created
and developed by three of the original developers of the Asterisk platform,
Anthony Minessale II, Brian West and Michael Jerris. With a focus on modulatory,
cross-platform support, scalability and stability, FreeSWITCH offers one of the
most flexible platforms to build your own UC suite around.FreeSWITCH supports
SIP, H.323 and even WebRTC to leverage the latest advancements in the
technology, and easily integrate and interface with other any of the other popular
open source PBX platforms available.
In an effort to reduce the complexity of a system, FreeSWITCH utilizes freely
available software libraries that will preform the necessary functions for your
system to work. FreeSWITCH offers the usual calling features and even adds
some extras like speech recognition and synthesis and even PSTN interfaces for
analogue and digital circuits.

OpenPBX by Voicetronix
Voicetronix is a telephony solutions and equipment provider that offers not only a
range of hardware, but also an OpenPBX open source platform. Voicetronix’s do-
it-yourself OpenPBX is actually a web enabled PBX application that features a web
based user management portal, as well as a management GUI for quick and
simple configuration. The normal features of an auto attendant, automatic call
distribution, least call routing hunt groups and even unlimited voicemail make the
platform are included.

Unique features like unlimited call hunt groups, music on hold and the ability to
display call records make the platform a very strong solution for businesses in
need of a basic call or contact center software. With basic CRM already enabled
and baked into the platform, users may not need to adopt a separate CRM
solution, saving time and money. Voicetronix’s OpenPBX even allows for
voicemail to email, click to dial and call transfers.

PBXInAFlash
PBXInAFlash aims to make it possible to setup your own PBX server, as the  name
puts it, in a flash. The project has collected everything in one place that users will
need to create their own PBX system within under an hour, one that utilizes the
CentOS Linux operating system, including an Apache web server, SendMail
server, and MySQL database server as well as firewalls and necessary protocols.
The system also gives users the choice of Asterisk 1.8, 10, 11 or FreePBX 2.9,
2.10 or 2.11.
Users also have the option to choose from dozens of add-ons to truly custom
tailor the system with helpful features like automatic backups, Caller ID look up
services, SSL keys, Google Voice integration, text-to-speech functionality, and fax
support just to name a few. Everything is designed to be simple, and add-ons can
be installed with one click in under a minute. With the number one goal of no
bloat and no bugs, PBXInAFlash seems to be the quickest and easiest solution to
adopt.

FreePBX
Combining the best of both worlds, and looking to leverage the great work
already done by the Asterisk project,  FreePBX is a web-based, open source
graphical user interface (GUI) to help users better manage and configure their
Asterisk based system. While the project utilizes the Asterisk system, users can
download either just the GUI to add on to their existing system, or the entire
package including a per-configured system OS, Asterisk, and the FreePBX GUI on
top as well as all of the necessary dependencies. So while adopting just Asterisk
may require some more technical knowledge to take full advantage of, or to
create your own GUI, FreePBX brings it all together.
FreePBX also makes it possible to establish your own unlimited SIP Trunks thanks
to SIPSTation integration directly into the platform. FreePBX also includes a long
list of commercial modules and add-ons to enhance your system with even more
features, and a reseller program to ensure proper training, quality and stability to
resellers and end users. Essentially, the program will train and educate your sales
and support team if you are looking to repackage and sell your product.

OpenSIPs
With a stronger focus on open source implementation of a SIP server, OpenSIPs
still makes it possible to establish your own independent, custom Unified
Communications package as well as a PBX. The platform supports voice, video, IM
and presence services with a scalable and modular design, so like any other
platform it should be as customizable as you need it to be. OpenSIPs is labeled as
one of the fastest SIP servers, and offers a robust solution at an enterprise or
carrier-grade class.

OpenSIPs even provides an ongoing list of benchmarks and performance tests to

back up their claim. In a similar fashion to Asterisk, OpenSIPs provides recorded
webinars, and in depth manuals for every version and configuration of the
OpenSIPs platform. A web based configuration portal should make it simple to
gather data and statistics of your service, as well as on the fly configurations.

Kamailio
With 15 years of development under their belt, Kamailio continues to build on and
expand their open source SIP server. With powerful features like asynchronous
TCP, UDP and SCTP, TLS to ensure secure communications for your VoIP data
including voice video and text, and even WebRTC support the hard work shows.
Kamailio also supports instant messaging and presence, along with more behind
the scenes features like least cost routing, load balancing, routing fail-over and
even authentication and authorization for enhanced security.
Kamailio actually offers one of the strongest level of security we’ve seen on this
list, and would be a solid recommendation for any team or business that needs to
keep everything as locked down and secure as possible thanks to the level of
encryption the platform provides. On the flip side, Kamailio might be a bit more
difficult to adopt, requiring more in depth knowledge of the SIP protocol itself to
fully leverage the platform.

3CX
The 3CX Phone System is another software based, open source PBX that is based
on the SIP standard. The solution makes it possible to enable extensions to make
calls on both the PSTN or just standard VoIP services. The platform also offers an
easy to understand web based GUI, and the process to install should be a rather
simple one just by running an executable file on a windows based machine. 3CX
seems to take out the hassle and development needed to establish your own PBX
server, and even offers both iOS and Android mobile clients, as well as Windows
and Mac softphones. Web conferencing is also made possible thanks to WebRTC
adoption.

3CX also makes it possible to drastically improve your customer experience with
Click2Call for your website or apps, and CRM integration so agents are always
prepared. An online training academy is always available for users to learn how to
fully leverage their 3CX platform.

What is a SIP Server?

SIP stands for session initiation protocol. A SIP server is a network protocol that is used for
establishing connections for communication of different subscribers and also deals with call
management. Also, SIP servers are often used to manage call connection in VoIP solutions. A SIP
server can
 Set up a connection between multiple endpoints
 Initialize media parameters for the endpoint, using SDP protocol
 Modify and adjust  the parameters during the session
 Replace one endpoint with an another or new endpoint
 Session termination
Difference between PBX and SIP-enabled PBX
A PBX is a system that connects the individual extensions to the external mobile networks or
telephone lines, whereas a SIP-based PBX connects to the internet and allows to make calls
over the internet using SIP protocol.

Different VoIP SIP servers and their GUI

Here in this article let us understand various VoIP SIP servers and their User Interfaces that are
used to manage each SIP server.

Asterisk
Asterisk is an interactive voice response platform that includes an automatic call distributor
functionality. It is an open-source PBX that allows building own communication applications. It is
a framework that is used for building real-time communication solutions and multi-protocol
solutions. Digium sponsors it.

Features of the asterisk

 Call monitoring
 Call transfer 
 Call waiting
 Append Message
 Blacklists
 Blind transfer

Functions
 Small server footprint for processing and memory capabilities
 It has Longlife and support mechanisms
 It allows access to gateway configurations. Documentation, forums, etc
 For auto-provisioning it has an extensive database of end device templates.

Protocols
 ISDN
 SS7
 MGCP
 H.323
 IAX
 SIP

Supported OS
 Linux
 BSD
 Solaris
 Mac OS X
GUI- The Graphical User Interface that is used to manage the Asterisk is Free PBX

Free switch
It offers the most flexible to build its unified communication suite. It is majorly focused on 
 Cross-platform support
 Stability
 Modulatory
 Scalability
Freeswitch as the essential calling features and also includes the advanced features like PSTN
interfaces for digital and analog circuits, speech recognition and synthesis. 
The free switch makes use of the freely available software libraries that allows the required
functions of the system; this way, it reduces the complexity of the system.

Functions
 It has a Multi-tenant platform, and each tenant is segregated 
 Freeswitch has clustering capabilities 
 Concurrent calls, when given the same underlying hardware, can be maximized with
increased capabilities.

Protocols
 STUN 
 SCCP 
 MRCP 
 SIP 
 IAX 
 Skype
 ISDN 
 H.323 
 JINGLE 
 SIMPLE 
 XMPP 
 RSS 

Supported OS
 Mac OS X
 Linux/BSD
 Windows
 Solaris
GUI- The Graphical User Interface that is used to manage the Freeswitch is
FusionPBX

Yate
The abbreviation of Yate is Yet another telephony Engine. Yate is an extensible GPL licensed
PBX and open-source communication software that supports 
 Video
 Voice
 Instant messaging
It supports scripting languages like PHP, Unix shell, Python, Perl even though C++ is its core
software.

Main components of Yate

 Core
 Message engine 
 Telephony engine
 Yate modules
Protocols
 H.323 
 SIP 
 MGCP 
 SCTP 
 MAP CAMEL
 SS7 over IP
 IAX 
 ISDN 
 JINGLE 
 XMPP 
 Cisco SLT 
 SCCP 
 TCAP 

Supported OS
 Mac OS 
 BSD
 Windows
 Linux
GUI- The Graphical User Interface that is used to manage the Yate is Yateclient

Elastix
Elastix has support for a wide range of hardware like Digium, Snom, Yealink. It has included a
call center module with a predictive dialer. It only offers up to 8 free SIM calls to 25 users.
Elastix offers unified communications server software, which includes the following features:

Features of Elastix
 Email
 Instant Messaging
 Faxing
 Collaboration functionality
 Integrated softphones for Mac and Windows
Elastix also includes the features that are brought from other open-source projects like Postfix,
HylaFax, FreePBX, Openfire

Kamailio/ OpenSER
Kamailio, previously known as OpenSER, is a free and open-source sip sever and offers a high-
security level. Compared to other SIP servers, Kamailio is a bit difficult to adopt as it requires
deep knowledge of the SIP protocol. To provide secure communications, it has powerful features
like

Features of Kamailio
 SCTP
 TLS
 Asynchronous TCP, UDP
 Instant messaging
 Least cost routing
 Load balancing
 Routing fail-over
 Authentication and authorization
Due to its ability to provide high-level encryption, it is one among the top secured servers and is
recommended to businesses that prefer the security and wants everything to be kept inside.
GUI- The Graphical User Interface that is used to manage the Kamailio is Siremis

OpenSIPs
OpenSIPs is one of the fastest SIP servers that offer robust and scalable solutions at an
enterprise level. It is a multi-functionality sip server that majorly targets delivering a high-level
technical solution which can be used in professional SIP server platforms. This technical Solution
providers mainly includes
 Quality
 Performance
 Security

Features of OpenSIPs
 Multi-domain support
 Perl programming Interface
 Least cost routing
 Variables support in the script
 IPv4 and IPv6
 Modular architecture
 Call processing language
GUI- The Graphical User Interface that is used to manage the OpenSIPs is OpenSIPs
CP

Flexisip Server
Flexisip is a scalable and modular SIP server that offers all the required to deploy an own SIP
service for desktop or mobile applications. It is easy to install, and for various purposes, Flexisip
can be integrated into your SIP infrastructure.

Features of Flexisip
 Real-time statistics through a command-line interface
 Push notifications
 Group chats 
 Real-time presence status
 Identifying users of service within the address book
It is essential to understand and do proper research before choosing the SIP server for your VoIP
Solution setup. A single server can not meet all your needs, and every SIP server has its pros
and cons. You wisely need to choose the VoIP SIP server that meets most of your requirements.
Krify is a leading VoIP service providing company with competency in customizing
Linphone softphone by using Fusion PBX. We have customized softphones for various
organizations and have developers who are expertise in Linphone development including the
customization with advanced features. For more information reach us here.