Panoramisk

Test of the 3CX Phone System free edition

A PBX system on a Microsoft Windows operating system? Yes it is possible and 3CX with its Phone System is proposing such a solution. After some tests on the free edition, here is our analysis.

First, we should start this article with the fact this product is really a good one. Most standard features available on a PBX are available and if the target is SOHO and small organisations, you should give it a try. The installation is really simple and no need to have deep knowledge in voice over IP to get its first voice network up and running.

For sure, this free edition is not proposing everything we can think about, but some additional products are available and the cost stays low. Some could prefer to this implementation a more opened one, like an Asterisk based one, but this product is doing the job for most of our needs and is fully integrated, no installation hassle.

If you don’t require an advanced computer integration or API development, it could be a nice way to jump in IP telephony at a reasonable cost. You just need to add some IP phones or softphone and a SIP provider over the Internet or a gateway to get connected to the PSTN, that is all.

You will find in the following table some more information about the product and some of the test we have conducted on it. We got some help on specific topics directly from the forum, but most features are easy to set up. Don’t hesitate anyway if you got question to drop us an email.

Product
iPBX product	3CX Phone System (free edition)
Version	3.1.2434.0
Web site	`http://www.3cx.com/`
OS used for test	Windows XP, SP2
HW used for test	Intel Pentium M, 1.7GHz, 512 MB RAM
Additional tools used for this test	Aastra 53i, Polycom IP310, Softphone X-lite, pjsip suite, Asterisk 1.4	Additional comments
Installation	Really straightforward installation through a single Windows installer program. The parameters asked for are the software directory location, administrator credentials for the web access, as well as the number of digits for the internal dial plan and the SMTP server to be used for voice mail notifications. All components are fully integrated, there is no need for any additional software. We can find installed afterwards an Apache server for web configuration through PHP and a Postgres SQL database. Any Microsoft Windows version is supported, from desktop to servers. This last is probably the good target for production purpose, but a dedicated XP could be stable enough to support a small amount of communication.
Used space on disk	Around 90 MB are used on the disk after installation and a quick configuration of few SIP phones. This amount seems normal and comparable to Asterisk kind of solution. No hassle on this topic since nowadays disks are very large and Windows operating system is already taking a fair large amount of space. We could think about installing this software PBX on a virtual infrastructure¹ if the host server is fast enough and communications between devices are not using the internal media server (RTP flow crossing the PBX during a conversation).
Processes	Processes are identified as Windows services and are started by default at boot time. There are specific processes for the core, the database, the web server, and three specific components which are the digital receptionist, the voice mail manager and the media server.
Memory	After installation and a small configuration with some SIP phones and trunks, the amount of used memory is still good: 35M for Apache, 26M for Postgres, 7M for core system, 14M for the receptionist, 7M for the media server and 6M for the voice mail server. These values could be different on your installation, but this gives a fair amount of the memory that should be affected to the host, especially on a virtual machine.
Management	Management and configuration are performed through the web interface which allows a remote management. There is no support for HTTPS communication, one should be aware of this when dealing with security over the WAN and the LAN. This management interface is simple and clear, main menu on the left side regroups the configuration of extensions (phones), lines (through SIP gateways and providers), routing rules for output dialling. This is enough for a simple system configuration. The next menu options allow advanced configuration with the digital receptionist (IVR), the call groups and queues (not available in the free edition). Last we found the general configuration option and the reporting system. On the web interface, all modules are identified on the top right corner and their running state is represented by a colour, green when present and red otherwise.
Phones
Type	SIP only, analog and digital interface are not supported, nor H.323 devices.
Configuration	Each extension is specified with standard SIP fields like the number, the name, the credential for authentication. From this menu we can also manage the voice mail service and PIN code for access. Each line should match the digit number size specified during the installation of the PBX. The choices are from 3 to 5 but doesn’t seems to be modifiable afterwards. If IP phones doesn’t need to be authenticated, it is possible to clear the credentials fields. This is not recommended but possible, which is a good thing.
Forward	Forwarding of incoming calls can be configured towards voice mail, another extension, a ring group or an outside number. These targets are configured based on the line status: no answer, busy state or for any calls. Another option could be used when phone is not registered from a SIP perspective, this could be useful for mobile users.
Automatic phone configuration	not available
Codecs	not configurable for each extension, the 3CX solution proposes PCMU, PCMA and G729 in the INVITE message.
Trunks	SIP only.
SIP	SIP trunks are used for any connection towards the external world. This includes SIP providers and gateways to PSTN or any other IP voice protocol.
Configuration	Both SIP providers and PSTN gateways trunks are simplified through some templates including telcos (Vonage, Broadvoice, inPhonex, …) and manufacturers (Linksys, Patton, Grandstream, …). It is also possible to create a trunk from a generic template. First we need a provider or gateway configured, afterwards it will be possible to create lines and DiD on top of it. This split allows to separate specific parameters on both sides which seems a practical approach. The VoIP provider settings requires both registrar and proxy servers, and supports also a STUN service, if the connection is towards another PBX on the same domain, the STUN should be suppressed or it will present the external IP address. The registration settings are available and should be setup accordingly to the SIP account at the telco, we can apply authorize check on each calls, both ways. We can also specify the codec list to be used and how to authenticate an incoming call. This part is not easy since a call presented will not be accepted by default, hopefully the log is explicit and provide the administrator with where to look at. Once the gateway or the SIP provider is configured, one or multiple lines could be added on top. A line is composed of the authentication information and the routing part for incoming calls. Each line needs to be linked to a single extension or service on the iPBX. If you need to use direct lines, each one should be created afterwards through the DiD configuration. This part is not the best one of the product, since easy to manage for few internal extensions but really cumbersome for more than 10 of these. Each DiD can direct call towards an extension, a voice mail, and IVR menu and can also change its behavior based on office hours.
Voicemail	Available and per extension. The access is by default associated to a specific extension available on the internal dialplan. There is no direct access to the phone associated voice mail, thus the extension is asked when dialing the voice mail portal. The authentication PIN number is configured at the line level and asked to get access to the box. Vocal messages could be personalized for each line.
email notification	OK
email message forward	OK
navigation	Available through an integrated IVR, voice a bit quick and only available in English. It is possible to change the PIN number from this menu as well as the greeting message. There is only one greeting message, nothing related to busy or not available.
MWI	not available on the free version.	Included in the Small Business Edition and above.
Quotas	not available
Line pickup	not documented, needs modification in the `.ini` file. Information found on the forum, why not in the interface?
Line supervision	not available	could be done with the Windows assistant software, but not the free version.
Parking	not available
Call Transfer	Support for SIP transfer from phone (blind and attended), but no specific extension for phones without this feature.
Intercom	not available
Caller ID presentation
phone configured	yes
3CX PBX configured	yes
Outbound routing	By default, one outbound rule is created for each line (towards gateway or telco), the prefix is suppressed and only the suffix is presented on the SIP call. This can be changed based on the provider or the gateway, but also based on the prefix used and the extension issuing the call. A backup rule is also available, which allows to have a second provider or backup a SIP line to a gateway, but the renumbering rule should be the same which can be tricky in some situation.
Inbound routing	Available only through line and DiD, but no explicit inbound call routing rules are available.
Phone group	Available with all and hunt strategies. Missing a least used strategy.
Queueing	Not available without upgrade.
IVR	Called Digital Receptionist. Each IVR menu is configured through a simple interface stating an action for each numerical key that can be hit during a voice message played. Each page requires a prompt which is played waiting for a DTMF event from the caller. The prompt should be recorded on the management station and pushed to the configuration interface, this is not as efficient as a direct recording extension on the PBX but allows more complex messages. The mp3 format is supported, which is really simple. All actions are presented in a drop down menu, no mistake are thus possible, and any extension or group configured in the PBX are available. This is a simple way of configuring the IVR part but will not be able to handle hundreds of extension easily.
PBX supervision	The line status allows a quick view off the phones, gateways and trunk lines configured and their respective status (registered, ringing, on line). When on call, the extension numbers are displayed. No action is available from this view, but it is clear and usable with a limited amount of lines configured.
Call Assistant	Available on Microsoft Windows only, the free edition is able to display each phone status and the engaged call. For more actions like transfer an upgrade is required.
Fax support	Nothing related to fax support.
CTI	No CTI API available, but Microsoft Exchange connectivity on the Entreprise Edition.