The FreeSWITCH telephony engine is a powerful system enabling voice, video, presence, chat, and other media types via a variety of protocols.
The 'release' version AUR. The following instructions assume you are using the freeswitch-git package.AUR and the git version AUR are available in the
Also, you may wish to configure FreeSWITCH build options. Edit the PKGBUILD and change any BUILD CONFIGURATION options to suit your desired usage.
The FreeSWITCH configuration files with the custom modules.conf and modules.conf.xml reside in /etc/freeswitch. For following FreeSWITCH documentation, the base directory is /var/lib/freeswitch (generallly seen as /usr/local/freeswitch in FreeSWITCH documentation).
FreeSWITCH comes out of the box with a default password for registrations to users 1000-1019 as '1234'. You are advised to change this before running it. This variable is set in /etc/freeswitch/vars.xml. The overall default configuration given is a kitchen sink featured PBX, likely many more things than are typically used. Customizing the PBX (or non-PBX) features of FreeSWITCH is beyond the scope of this document; see the FreeSWITCH Wiki for in-depth documentation.
Upstream documentation as well as the original conf/ directory are provided in /usr/share/doc/freeswitch.
Startup options are configured in /etc/conf.d/freeswitch.conf. You may wish to add -nonat if you are not behind nat, see freeswitch --help for a full list of command line options.
FreeSWITCH can be started with
rc.d start freeswitch
To start FreeSWITCH upon each boot, add "freeswitch" to DAEMONS in /etc/rc.conf if using the default archlinux default init system. If using Systemd or Runit, add to your running service via their provided methods. You'll need to use the -nc and -nf options to the freeswitch command line to keep it running in the foreground as supervisors expect.
- Fire up a SIP Client
- Register to your FreeSWITCH box as user 1000, password what you set as default_password in vars.xml
- Dial 9196 (You should connect to an Echo Test)
- To measure call capacity, you can use StarTrinity SIP Tester (see an example performance report for 2250 G.711 channels)
To see interesting things you can do with a dialplan, open up /etc/freeswitch/dialplan/default.xml and scroll through those examples. Dialing the numbers that match the 'expression' of a condition from your SIP client will demonstrate their use.
You can dial into the FreeSWITCH public voice conference, for instance, by dialing 9888 (8k codec), 91616 (16k codec), or 93232 (32k codec)
FreeSWITCH support is available in #freeswitch on Freenode