In this article, we are going to talk about the open source telephone system called FreeSwitch. This system can be used as PBX station. It should be mentioned that FreeSwitch works on Windows, Mac OS X, BSD, Solaris and Linux. FreeSwitch developers participate in other open source projects and contribute to Asterisk, SIPX, CallWeaver. FreeSwitch supports various protocols, such as SIP, H.323, IAX2, which allows you to interact with sipX, OpenPBX, Bayonne, Yate, or Asterisk.
We are going to tell you more concerning the configuration, work of FreeSwitch system and features they offer. As this telephone platform has the peculiarities of PBX station, you will understand why do you need to install this for your office telephony or other telephone purposes.
How FreeSwitch phone system functions?
FreeSwitch supports numerous narrow- and wideband codecs, making it an ideal bridge for older devices in the future. Some codecs are supported only in the pass-through mode. This means that compressed data is transmitted right through the subscribers without any processing. Voice channels and conferences can operate at frequencies of 8, 16, 32 and 48 kHz and allow you to combine channels with different frequencies. This also supports advanced SIP technology features. FreeSwitch also can build softphone, interface with other PBX station.
What are the features of FreeSwitch
There are big quantity of various phone services and phone apps that this system consists of. Look at some of them like:
- Call conference;
- IVR voicemenu;
- Speech synthesis;
- Welcome message;
- Call recording.
How to configure FreeSwitch telephone system
We propose you to learn the example of setting and connection the FreeSwitch. For this, you need the username, password, your domain. See the process in details:
- Create xml file and setup it according to SIP data you got from us: /etc/freeswitch/sip_profiles/external/freezvon.xml;
- Create the file with dialplan: /etc/freeswitch/dialplan/freezvon.xml;
- Then create in Freeswitch internal number "SIP number" to what you can connect IP-telephone/softphone for receiving and making calls. Edit the file with needed info /etc/freeswitch/directory/default/sip_number.xml.;
- For making FreeSwitch to read the configuration files you need to do the following command fs_cli -x reloadxml.
You have also a chance to connect and use virtual PBX station that works the same as Freeswitch. Moreover, you can set it and control your call processes via personal account online and do not use heavy technical equipment.
Use virtual phone number for FreeSwitch
You can connect the VoIP phone number to your FreeSwitch system, you should complete some steps in order to purchase virtual numbers for making and receiving calls via our website. Moreover, it is possible to get free SIP-account after buying a virtual number, but you need to submit a request to our technical team via personal account on our site. Then you will be able not only to obtain calls, but make outgoing calls via SIP thanks to free apps as Xlite or Zoiper downloaded to any of your device as (PC, smartphone, tablet, laptop).
If you have several questions, just contact our technical team department here . We are available via online live chat, Skype, email or other phone numbers you may see on this website on a page called “Support”. We are going to make the situation clearer about VoIP telephony services including PBX or FreeSwitch you want to order.