Getting Started Guide

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The purpose of the following page is to instruct new users on how to configure FreeSWITCH™ in a basic way. It is also intended to provide people with a basic understanding of the configuration files and how they are processed. If FreeSWITCH™ is not already installed on your system you should read the Installation Guide. This guide is not intended to allow for more than a default configuration.

Contents

FreeSWITCH layout

FreeSWITCH™ has a modular design. This design allows it to be extended without breaking functionality or requiring massive recoding efforts. Each module provides specific functionality, often commands that are related are grouped into one module. You will need to enable the modules that you desire, based on their function. By default FreeSWITCH™ comes with a good set of modules loaded, to enable most basic functionality.

Read up on the Default Configuration for more information on the default configuration itself. See also Before You Start for some good information to consider before putting a system into production.

Installation

See the Download & Installation Guide.

Configuration

The following sections will show you each step that you will have to take to get a functional system.

Kernel

If you don't use the new timerfd functionality, the kernel should have

CONFIG_HZ_1000=y
CONFIG_HZ=1000

Firewall

If you are going to route traffic across your network, you need to ensure that you do not have a firewall blocking traffic on the specified network paths. Commonly firewalls are installed at egress points, where your local network connects to the internet. This is not always the case in some corporate environments. If you are not responsible for your network, you may wish to contact the group or individuals who are responsible for it.

Information they may need to enable traffic would include the port numbers. Various protocols use different ports for different things. Some you need bi-directional connectivity, some you only need outbound. Depending on your call scenario you may only need to allow connections that are initiated by your system. If you have any problems with protocols not working, it is recommended that you disable your firewall to the FreeSWITCH™ system, place a test call, and see if the problem persists. A misconfigured firewall is responsible for many common problems.

For more information, including the ports which need to be configured, see the Firewall page.

Configuration Files

FreeSWITCH™ configuration data is stored in XML format. These files are located in the conf directory under the FreeSWITCH™ install directory. By default on a Unix-like system this will be /usr/local/freeswitch/conf. There are preprocessor markup tags that tell the configuration parser to perform tasks, such as including other files, which are carried out before the XML is parsed. It might be worth looking at Default_Configuration_Overview which describes some of the initial default settings. It may be useful to adjust some of them. There are a large number of different configuration files used by FreeSWITCH and fully understanding how they all fit together can take some time.


Folder Description
autoload_configs These configuration files are loaded automatically into FreeSWITCH. They contain configuration information for almost all core modules.
dialplan This is the place where you setup your dialplan. There are some examples of how to configure your dialplan included.
directory The directory contains all users that may register and use freeswitch as their PBX.
jingle_profiles Jingle is the mod that FS uses to handle XMPP. Profiles tell FS how to use the protocol. Each profile has its own ip port.
lang Tells FS how to say currency etc in different languages
mrcp_profiles MRCP is used to allow FreeSWITCH to use speech recognition and TTS
sip_profiles Tells FS how to talk sip. Each profile has it's own port.

Managing your personal configuration files

You should consider using some versioning control system for managing the FreeSWITCH™ configuration files. The obvious advantage of this is being able to easily keep track of changes you made to the configuration and being able to easily switch back to some old configuration. Furthermore it would be possible to merge changes from the official git source tree configuration into your own configuration tree.

One versioning control system to consider is git. It is used to manage the official FreeSWITCH™ source tree, too. Though this would not oblige you to use it for your configuration files, it is a smart idea to do so, because once you are familiar with git, you have the skills even to manage the FreeSWITCH™ source tree with it.

To set up your own personal git repository, use this command sequence:

cd freeswitch/conf
git init
git add .
git commit -m "Initial commit."

To view the git log just use command:

git log

After you have edited your configuration, to see the changes made use

git diff

Unfortunately it is not so easy to have git merge the conf subdirectory in the original source tree of FreeSWITCH™ into your running conf directory. There are some approaches for this where the most promising would be to use the git subtree script with a complicated setup.

A workaround would be to create a complete clone of the FreeSWITCH™ git source tree and use the conf directory of the clone as your personal conf while compiling FreeSWITCH™ in a complete other clone of the git source directory tree. You feel the overhead.

Configuring FreeSWITCH

This guide aims to explain the default configuration layout and how you configure it to your liking. Read up on the default config for more information.

This layout is in no way required, you can minimize the configuration to just one file if needed. The main config file is called freeswitch.xml, you don't need to change it in any way, it just loads all other config files.

vars.xml

vars.xml is used to define a couple of variables that are used system wide. NOTICE you can not comment these X-PRE-PROCESS lines out. They will still be active in an XML comment.

See Config_vars.xml for example.

In the default configuration the vars.xml file is used to define some preprocessor variables. In a first time setup you will most likely want to set:

  <X-PRE-PROCESS cmd="set" data="outbound_caller_name=FreeSWITCH"/>
  <X-PRE-PROCESS cmd="set" data="outbound_caller_id=8777423583"/>
  <X-PRE-PROCESS cmd="set" data="call_debug=false"/>

directives for use in other portions of the configuration.

The default port of FreeSWITCH is also defined in:

<param name="sip_port" value="5060" />

Security advice

Be careful with your passwords. A good idea is to use preprocessor variables for all your passwords and define them all in a user only readable/writable file FS/conf/PASSWORDS.xml (FreeSWITCH config top level directory). As a benefit you will have a nice reference file for all your passwords.

To have these passwords read, you need to put the following line at the bottom of FS/conf/vars.xml:

<X-PRE-PROCESS cmd="include" data="PASSWORDS.xml"/>

Then you define the loads of password like this:

<X-PRE-PROCESS cmd="set" data="default_password=669933"/>

In this example, the default_password is defined in FS/vars.xml, but redefined at the end where we include the PASSWORDS.xml file containing another default_password. The redefinition is done so that the FS/conf/vars.xml can even be shared without the password being compromised.

Don't forget to chmod your FS/conf/PASSWORDS.xml:

chmod 600 PASSWORDS.xml

You should define all your passwords with a fixed scheme like this:

<X-PRE-PROCESS cmd="set" data="PASSWORD_default_vm=448899"/>
<X-PRE-PROCESS cmd="set" data="PASSWORD_sipgate_939393=113388"/>

Notice the default passwords go first, they are the first thing to consider changing / abolishing.

You can use your password variables in the dial plan like this (note the double "$"):

<param name="password" value="$${PASSWORD_default_vm}"/>

So that you really understand how things are to be done, I will now show you my real life PASSWORDS.xml file from my main 24/7 FreeSWITCH installation here:

Just kidding! ;-)

autoload_configs

autoload_configs is a directory where a lot of the freeswitch configuration resides. This directory is located in $PREFIX/conf/autoload_configs. The default freeswitch.xml preprocesses XML files matched by the glob conf/autoload_configs/*.xml

modules.conf.xml

See modules.conf.xml for an example of modules.conf.xml.
modules.conf.xml tells freeswitch which modules to load. There are certain modules required for operation so you should not edit this file unless you know that you wish to add or remove a specific module

sofia.conf.xml

mod_sofia is used to create SIP endpoints in FreeSWITCH. If you are familiar with Asterisk, each SIP Profile in mod_sofia is similar to chan_sip in Asterisk, though much more powerful.

The sofia.conf.xml file contains a "X-PRE-PROCESS" directive that includes other XML files (within, by default, conf/sip_profiles/) that define one or more "SIP Profiles". A SIP Profile is a SIP UA (User Agent or an endpoint), which communicates with other SIP endpoints.

In practical terms, files that talk to your service provider should go under conf/sip_profiles/external/

named perhaps, provider_name.com.xml

A "User Agent" ("UA") is an application used for running a certain network protocol. A Sofia User Agent uses the SIP protocol.

See Sofia_Configuration_Files for an example sofia.conf.xml.

See also SIP Network Elements on wikipedia.

What SIP Profiles Do

The SIP Profiles in FreeSWITCH can often lead to confusion. Unlike other softswitches (like Asterisk), FreeSWITCH allows you to handle media (calls, video, etc.) differently based on where the equipment is attached to your network. This assists with security as well as providing added functionality.

SIP Profiles allow you to define paths to devices or carriers that may live inside or outside your network. These paths can be of many different types, but must consist of a unique combination of port and IP pairs. You could have SIP profiles for your internal network, or multiple profiles for each subnet of your internal network, or even completely different protocols like IPv6 as profile definitions. This helps FreeSWITCH identify how to route different types of calls when necessary, and also gives you the flexibility to tailor your dialplans based on what path a call originates to/from.

In the sample configuration there are three default categories for where a SIP Profile can reside. Each profile is detailed below.

SIP_Profiles

The default profiles are "internal" and "external", each serving a broad, general purpose. External SIP profiles are generally used to communicate with your 'gateway' service provider, such as FlowRoute or similar company providing telephony service via SIP protocol to you. The Internal sip_profile is generally used to communicate with devices on your local network.

Internal

File Location: $PREFIX/conf/sip_profiles/internal.xml

This profile generally refers to devices that reside on your internal network. Those devices will inherit all configuration options in the internal profile you setup. This is typically your internal phones with extensions. By default, these devices must use SIP authentication.

The Internal (formerly called "default") SIP profile is configured to listen on the primary IP address of the machine (unless you set $${domain} to something else in vars.xml) on port 5060 (the default SIP port). The internal SIP profile does authenticate calls and is not suitable for configuring trunks to providers or outside phones, in most cases (i.e., use the External profile for configuring trunks to providers or outside phones).

The Internal profile should be used if you intend on handling registration for sip clients (i.e., a SIP registrar). You may wish to skip to the section Getting_Started_Guide#directory to configure additional sip clients.

Authenticated users of this profile are normally set (via the variable user_context) to use the dialplan defined in the default context.

See Config_default.xml

External

File Location: $PREFIX/conf/sip_profiles/external.xml

This profile generally refers to devices or gateways that reside outside of your network. Generally this is where you define your external trunks/VoIP carriers/etc. Again, those devices will inherit all configuration options in the external profile you setup.

The External (formerly "outbound") profile handles outbound registrations to a SIP provider. The SIP provider sends calls to you, and you send calls to your provider, through the external profile. The external profile allows anonymous calling, which is required as your provider will never authenticate with you to send you a call.

In order to secure your FreeSWITCH it is wise to link your outbound profile to a dialplan context other than 'default', which in the default configuration is the where authenticated users are placed.

NOTE: The default port for external connections is 5080. This also means if you are trying to connect a phone remotely to your FreeSWITCH box, you must set it to connect on port 5080, or change your ports around. This confuses a lot of people. As an example, in X-Lite, if you are connecting to FS from the outside, set your server as "my.server.com:5080".

The dialplan for this profile (per-default) is defined in the public context

See Config_external.xml

NAT

The nat.xml file (and profile) have been deprecated. Use the External Profile and port 5080.

Other Profile Concepts

Ideas for other SIP profiles you might end up building:

  • Routing IPv6
  • Having multiple profiles for different networks or subnets
  • Having multiple profiles for different remote hosts/endpoints (remote offices, etc.)

Dialplan

The FreeSWITCH dialplan is a full-featured, XML-based call-routing mechanism. (There also exists support for Asterisk-like dialplans as well as really fancy real-time and/or back-end database-driven dialplans).

The best place to get started in learning about the FreeSWITCH dialplan is the Dialplan page here on the wiki. It discusses the basics of the FreeSWITCH dialplan. If you need an XML primer, check out the information on the Basic_XML page which will get you pointed in the right direction. There are two other important resources for you to learn more about the dialplan:

It is recommended that you compile FreeSWITCH with the default configuration and make sure it works before you start making customizations. Note that the default config files, including the default.xml dialplan, are often updated.

It makes sense to put your custom dialplan entries into the extensions subdirectory under conf/dialplan. You may create one or more XML files in this subdirectory, all of which will be included in the dialplan thanks to an "include" directive in the default.xml file. By keeping your customized extensions separate from default.xml you will be able to update default.xml without having to re-enter your custom extensions.

Some common extensions for testing

  • 1000, 1001, ..., 1019 - Generic SIP extensions
  • 5000 - demo IVR (requires sounds and music files to be installed)
  • 9195 - five second delay echo test
  • 9196 - standard echo test
  • 9197 - milliwatt extension
  • 9198 - tetris extension for demo TGML generation
  • 9664 - music on hold (requires music files to be installed)

See Also

Directory

See Config_directory.xml for a sample configuration.
The directory holds authentication credentials for SIP endpoints that will register to FreeSWITCH (most commonly users). The directory configuration default is configured to process the glob $PREFIX/conf/directory/default/*.xml by the configuration included in freeswitch.xml $PREFIX/conf/directory/*.xml.
This may seem confusing, but the default will work in most cases. Feel free to look at the examples in: $PREFIX/conf/directory/

User Configuration

See Config_1000.xml for a simple (1000.xml) configuration of a user (endpoint) which will register to FreeSWITCH.

Sample Gateway Configuration

See SIP Provider Examples for sample gateway configurations for different providers.
Outbound registrations to providers can also be made in the outbound profile. See Clarification:gateways this page for a clarification

Dialing out via Gateway

Add the following to default.xml (in prefix/conf/dialplan) for 11 digit dialing (US) via your newly created gateway configuration above. Remember: the default configuration provided by FreeSWITCH does not include a diaplan rule for outbound calls via the gateway you've defined. You will need to add something similar to the following.

   <!-- Dial any 10 digit number (2223334444) or 1+10 number (12223334444) here -->
   <extension name="Long Distance - voicepulse">
    <condition field="destination_number" expression="^(1{0,1}\d{10})$">
      <action application="set" data="effective_caller_id_number=12223334444"/>
      <!-- If your provider does not provide ringback (180 or 183) you may simulate
        ringback by uncommenting the following line. -->
      <!-- action application="ringback" /-->
      <action application="bridge" data="sofia/gateway/voicepulse/$1"/>
     </condition>
   </extension>

Receiving an inbound call from a Gateway

Add the following to public.xml (in prefix/conf/dialplan) to receive inbound calls via your newly created gateway configuration above. Make sure to set destination_number to a DID defined for this gateway. This will route all inbound calls to extension 1001. Remember, you can have multiple endpoints register to one account ( See Multiple Registrations ). You will need to add something similar to the following. Please note, this will only redirect to an extension, you will also need to have the extension bridge with an endpoint/user if that is your desire (See Dialplan_XML#Example_9 ).

   <extension name="Voicepulse">   <!-- your provider or any name you'd like to call it -->
    <condition field="destination_number" expression="15555551212">  <!-- your DID for this gateway-->
     <action application="transfer" data="1001 XML default"/>
    </condition>
   </extension>

If you want users to register with something other than the number that will reach them, then contact mapping is for you.

**Note: Not all providers work this way, if you have difficulty with routing incoming calls from your provider look in the Wiki section for SIP Provider Examples.

Running FreeSWITCH

Starting FreeSWITCH™ is done from the command-line. The location of the executable varies depending on your system.

Starting in Unix-based Systems

Now that you have compiled and configured FreeSWITCH™ its time to place a test call to ensure that everything is working so far. To start FreeSWITCH™ change to the bin subdirectory under the FreeSWITCH™ installation directory, and run FreeSWITCH™. In a Unix-like system that might be:

Type This
cd /usr/local/freeswitch/bin
./freeswitch


REDHAT CENTOS Startup Scripts:

cp build/freeswitch.init.redhat /etc/init.d/freeswitch
chmod 755 /etc/init.d/freeswitch

Starting in Windows-based Systems

The Visual Studio project runs via an executable located under the debug or release folder of your project's directory tree (depending on how your project was built). On a debug build of the Windows system start FreeSWITCH™ as follows:

Type This
cd <PROJECT FOLDER>\debug
FreeSwitch.exe


The FreeSWITCH™ command line will be available after successful loading of the application.

Command Line Parameters

Informational Tip

These are the optional arguments you can pass to freeswitch:

       -waste                 -- allow memory waste
       -no-auto-stack         -- don't adjust thread stack size
       -core                  -- dump cores
       -help                  -- print this message
       -version               -- print the version and exit
       -rp                    -- enable high(realtime) priority settings
       -lp                    -- enable low priority settings
       -np                    -- enable normal priority settings (system default)
       -vg                    -- run under valgrind
       -nosql                 -- disable internal SQL scoreboard
       -heavy-timer           -- Heavy Timer, possibly more accurate but at a cost
       -nonat                 -- disable auto NAT detection
       -nonatmap              -- disable auto NAT port mapping
       -nocal                 -- disable clock calibration
       -nort                  -- disable clock clock_realtime
       -stop                  -- stop freeswitch
       -nc                    -- no console and run in background
       -ncwait                -- no console and run in background, but wait until the system is ready before exiting (implies -nc)
       -c                     -- output to a console and stay in the foreground (default behavior)
       UNIX-like only
       -nf                    -- no forking
       -u [user]              -- specify user to switch to
       -g [group]             -- specify group to switch to
       -ncwait                -- do not output to a console and background but wait until the system is ready before exiting (implies -nc)
       Windows-only
       -service [name]        -- start freeswitch as a service, cannot be used if loaded as a console app
       -install [name]        -- install freeswitch as a service, with optional service name
       -uninstall             -- remove freeswitch as a service
       -monotonic-clock       -- use monotonic clock as timer source
       Options to control locations of files:
       -base [basedir]         -- alternate prefix directory
       -conf [confdir]         -- alternate directory for FreeSWITCH configuration files
       -log [logdir]           -- alternate directory for logfiles
       -run [rundir]           -- alternate directory for runtime files
       -db [dbdir]             -- alternate directory for the internal database
       -mod [moddir]           -- alternate directory for modules
       -htdocs [htdocsdir]     -- alternate directory for htdocs
       -scripts [scriptsdir]   -- alternate directory for scripts
       -temp [directory]       -- alternate directory for temporary files
       -grammar [directory]    -- alternate directory for grammar files
       -recordings [directory] -- alternate directory for recordings
       -storage [directory]    -- alternate directory for voicemail storage
       -sounds [directory]     -- alternate directory for sound files

 


Shutting Down FreeSWITCH™

To exit the FreeSWITCH CLI, simply type 'shutdown' or '...'. Note, this will shutdown FreeSWITCH. If you do not wish to shutdown FreeSWITCH, it is suggested to start the process inside of a "screen" session; In Windows, FreeSWITCH™ may be started as a service. It is not recommended that you start FreeSWITCH™ as a service until you are certain your configuration is valid. Also if started from a remote login session (IE SSH) it is not advised to terminate the connection with FreeSWITCH left in the foreground.

Some stuff to try out!

The optimal way of testing is to get a pair of SIP phones and try out features via the default dialplan. If you're already familiar with Asterisk, you could also start out by Connecting Freeswitch And Asterisk

  • Register two phones to FreeSWITCH with two different extension numbers.
    • Let's assume for the example that you have your SIP phones registered as extensions 1000 and 1001.
    • You can use extension numbers 1000 through 1019 with a password of 1234 (the default voice-mail password is the same number as the extension {Ex: Extension 1000 has its voice-mail password set to 1000}.


Sample softphone configurations can be seen on the Softphones page.

Once you've done that, you can try out a bunch of features already setup in the test dialplan.


Basic Calling

  • Call between 1000 and 1001 by dialing the respective extension.
  • Call 4000 to login to your voicemail box.
  • 870 - Redial last called # from this extension.
  • *69 - Call back the last person to call this extension.
  • 8+[ext] - Place an intercom call to extension [ext]. If no answer, it does not go to voicemail but just keeps ringing. Some phones have additional functionality for intercom calls.


Other Tests

  • 9192 - Info test (prints information to the command line interface).
  • 9193 - Video record test (records video to /tmp/testrecord.fsv)
  • 9194 - Video playback test (plays video from /tmp/testrecord.fsv)
  • 9195 - Echo test (5-second delay before returning echo).
  • 9196 - Echo test (real-time).
  • 9197 - Play milliwatt tone.
  • 9198 - Tetris, via tone streaming.
  • 9664 - Test Music On Hold out (if you installed sound files).


Advanced Calling Features

  • 88+[ext] - Eavesdrop on extension [ext] (ex: 881001). That extension must be on an active call. Once connected, you can listen to the call without being heard, or you can...
    • Press 1 to barge in with the distant party.
    • Press 2 to barge in with the local party.
    • Press 3 to barge in on both parties.
  • 5900 - Park a call in a hold queue (First-in-first-out style).
  • 5901 - Pickup a call parked in queue (that was put there via 5900).


Group Assignments The below functions involve the idea of keeping a database/table of which extensions belong to one of 100 groups (numbered 00-99).

  • 80+[group] - Delete this extension from calling group #[group] (can be two digits 00-99). A beep tone confirms the function worked.
  • 81+[group] - Add this extension to calling group #[group] (can be two digits 00-99). A beep tone confirms the function worked.
  • 82+[group] - Simultaneously ring all extensions in group #[group]. The first person to answer will be connected to you automatically.


Conferencing

  • 30+[conf] - Join conference #[conf] - 8kHz quality.
  • 31+[conf] - Join conference #[conf] - 16kHz quality.
  • 32+[conf] - Join conference #[conf] - 32kHz quality.
  • 9888 - Join the FreeSWITCH conference via a SIP Internet connection (8kHz).
  • 91616 - Join the FreeSWITCH conference via a SIP Internet connection (16kHz).

Note: Conference rooms are unique for the bit-level you choose (i.e. 3000 for conference room 00 will not join with participants in 3100 which is also conference room 00. They are different conferences).


Interactive Voice Response

  • 5000 - IVR demo (if you installed sound files).

Advanced

Preprocessor Commands

The FreeSWITCH™ configuration supports preprocessor commands reminiscent of (but not matching) apache server-side includes:

  • comment - Add a simple comment
<X-PRE-PROCESS cmd="comment" data="This is just a comment"/>
  • exec - Execute a shell command and parse its output [1]
<X-PRE-PROCESS cmd="exec" data="echo XML config loaded at `date` >> /var/log/fs_config.log"/>
  • include - Include another file
<X-PRE-PROCESS cmd="include" data="vars.xml"/>
  • set - Set a global variable
<X-PRE-PROCESS cmd="set" data="my_global_var='some value'"/>

In addition to the pre-processor PIs, which may appear anywhere, the freeswitch.xml file has a root element of "document", and all its children are "section" elements, with "name" attribute values being one of:

Preprocessor commands are included in XML style comment brackets. Even though they appear commented to an XML parser, they are still processed. The comment characters are required or they will not be processed. An example would be:

  <!--#include "switch.conf.xml"-->

freeswitch.xml

This is the main configuration file, and the only one that has a fixed name. When you start FreeSWITCH™ it will locate this file and process it. The default freeswitch.xml file includes other files to allow for easier maintenance of configuration data. The default configuration of most of these files is suitable for first time users, until you become more familiar with FreeSWITCH™ it is advised that you leave them to their default settings.

freeswitch.xml is divided into multiple sections, and each section is used by a different component in FreeSWITCH™. The sections are as follows:

  • The "configuration" section has children for switch.conf and modules.conf which are used by the core, as well as a child for each module in use.
  • The dialplan section defines which action to invoke when a call arrvies to the switch.
  • The directory defines the users that can connect to the switch (user agents).
  • The phrases section defines where freeswitch can find the sound files, and if it can use any TTS engines.

The structure below "document" and "section" varies according to the parent. There is currently no dtd/relaxng/xmlschema available to validate the file, though since it does not use xml mechanisms for inclusion that would be an obstacle to true validation (and the limited xml parser used by FreeSWITCH doesn't implement any of the XML standard mechanisms such as external entities or XInclude). XML namespaces are not currently used. There is no application-level versioning within the file.

How the configuration files are parsed

FreeSWITCH begins with its single, master configuration file (conf/freeswitch.xml). This master config file then loads (via "globs" e.g. "*.xml") additional config files. FreeSWITCH 1.0.4 installs with dozens of config files. You may stray daringly far from the stock install and arrange the files (and the loading thereof) however you wish: it's up to you if you want a single, huge config file or a series of config files. Each config file can include none or many other config files. Whatever method you choose, at runtime FreeSWITCH parses all the (one or many) files into a single, huge, resultant "running config" file (located at log/freeswitch.xml.fsxml) which mustn't be edited while FreeSWITCH is running as portions are memory-mapped into FreeSWITCH.

IMPORTANT When the FreeSWITCH console reports a configuration error, the line number it gives is the line number in the "running config" file, not the line number in any included file.

Example freeswitch.xml

Below is a snippet freeswitch.xml file. Of course, the bulk of the configuration is found in the included files.

Informational Tip

<?xml version="1.0"?>
<document type="freeswitch/xml">
  <X-PRE-PROCESS cmd="include" data="vars.xml"/>

  <section name="configuration" description="Various Configuration">
    <X-PRE-PROCESS cmd="include" data="autoload_configs/*.xml"/>
  </section>

  <section name="dialplan" description="Regex/XML Dialplan">
    <X-PRE-PROCESS cmd="include" data="dialplan/*.xml"/>
  </section>

  <!-- mod_dingaling is reliant on the vcard data in the "directory" section. -->
  <!-- mod_sofia is reliant on the user data for authorization -->
  <section name="directory" description="User Directory">
    <X-PRE-PROCESS cmd="include" data="directory/*.xml"/>
  </section>

  <!-- phrases section (under development still) -->
  <section name="phrases" description="Speech Phrase Management">
    <macros>
      <X-PRE-PROCESS cmd="include" data="lang/*/*.xml"/>
    </macros>
  </section>
</document>

 


Preprocessor Variables

For now, do not worry about all of the configuration properties, this guide will show you the minimum required to get a running FreeSWITCH™ system. After you have verified your installation you may then configure the system in more detail.

The first thing you should do is review conf/vars.xml and conf/freeswitch.xml. For now just make note of the preprocessor variables.

Edit This
  <X-PRE-PROCESS cmd="set" data="bind_server_ip=auto"/>
  <X-PRE-PROCESS cmd="set" data="global_codec_prefs=G722,PCMU@20i,PCMA@20i"/>

 


  • "bind_server_ip" Can be an ip address, a dns name, or "auto".
  • "global_codec_prefs" has a default of G.711 u-law. You can probably leave this alone, unless you need to change it to another codec, but most VoIP equipment will support this codec.

hint: If you need to troubleshoot your configuration, the log/freeswitch.xml.fsxml file contains the entire pre-processed freeswitch.xml file. (For testing and troubleshooting you could replace the freeswitch.xml file with this copy.)

hint2: If you have a [multihomed NIC] (Single Link, Multiple IP address) then remember that you may have to set a pre-process variable in the vars.xml file.

  <X-PRE-PROCESS cmd="set" data="local_ip_v4=<your ip address>"/>


Multi-Tenant Info / Multiple Company HOWTO

The Multi-tenant page gives details regarding running FreeSWITCH to support multiple separate companies. Multiple Companies is a HOWTO that describes setting up two independent companies on FreeSWITCH, where the files go, what they must contain, etc.

The Limitations of FreeSWITCH

If you want to know what FreeSWITCH can't do, refer to this page: The Limitations of FreeSwitch.

Asterisk Rosetta Stone

Rosetta stone is a page of common Asterisk -> FreeSWITCH translations.

See Also