FreePBX v16 Product Overview FreePBX is a web-based, open-source, graphical user interface that manages Asterisk, a voice-over IP (VoIP) telephone system. This image is optimised to run on AWS and has enhancements for operation on more minor instances. It utilises segregated disks so that you […]
FreePBX is a powerful and cost-effective telephony solution that provides businesses with a wide range of features and benefits. Whether you are a small business or a large enterprise, FreePBX can help you to improve your communication processes and provide better customer service. Its flexibility, scalability, and ease of use make it an ideal choice for businesses of all sizes. If you are looking for a reliable and customisable communication platform, FreePBX is definitely worth considering.
The call recordings are kept in the following location by default: /var/spool/asterisk/monitor This location is not backed up by default, so you may want to add this location to the custom backup list
The network files are located at: /etc/sysconfig/network-scripts ifup configfilenameifdown configfilename
In this article, we will explain how to configure a DHCP server in CentOS and Ubuntu Linux distributions to assign IP address automatically to a client machine. Installing DHCP Server in CentOS and Ubuntu The DCHP server package is available in the official repositories of mainstream Linux distributions, installing is quite […]
Install a fresh version of FreePBX on to new hardware Connect to IP via browser and create an admin user Vis SSH connect as root@ipaddress Run the command yum update In Module Admin from the GUI apply several updates at a time applying them to […]
If using sftp-ssh locate and edit /etc/sysconfig/network-scripts/ifcfg-eth0 If using commands nano /etc/sysconfig/network-scripts/ifcfg-eth0 Restart Service service network restart Source: How to set Network Settings from the CLI – PBX Platforms – Documentation
After many hours of testing and searching around in the asterisk config files created by Freepbx, I found out that all calls from the SIP trunks are passed to the context from-pstn, regardless of the trunk name or the context directive in the SIP settings. […]
Audio codecs Most commonly used are: G711 G729 G722 G711 PSTN quality codec which is supported in 99% of all devices/providers (but often disabled to save traffic). It has best quality but highest bandwidth (64kbps + ~20 kbps overhead ~= 84kbps to one direction). It […]
G.729 is a codec that has low bandwidth requirements but provides good audio quality. This is the most commonly used codec in VoIP calling and has a MOS rating of 4.0 G.711 is a codec that was introduced by ITU in 1972 for use in digital telephony. With […]
