Freeswitch: mod_commands

mod_commands processes the API commands that can be issued to FreeSWITCH via its console, fs_cli, the event socket interface, and scripting interfaces.

To see a list of available API commands simply type help or show api at the CLI.|



See below.

An API command can be called from the dialplan. Example:

 <extension name="Make API call from Dialplan">
   <condition field="destination_number" expression="^(999)$">
     <!-- next line calls hupall, so be careful! -->
     <action application="set" data="api_result=${hupall(normal_clearing)}"/>

Other examples:

 <action application="set" data="api_result=${status()}"/>
 <action application="set" data="api_result=${version()}"/>
 <action application="set" data="api_result=${strftime()}"/>
 <action application="set" data="api_result=${expr(1+1)}"/>

API commands with multiple arguments usually have the arguments separated by a space:

<action application="set" data="api_result=${sched_api(+5 none avmd ${uuid} start)}"/>
If you are calling an API command from the dialplan make absolutely certain that there isn't already a dialplan application that gives you the functionality you are looking for. See mod_dptools for a list of dialplan applications, they are quite extensive.

Extraction Script

Mitch Capper wrote a Perl script to extract commands from mod_commands source code. It's tailored specifically for extracting from mod_commands but should work for most other files.

use strict;
open (fl,"src/mod/applications/mod_commands/mod_commands.c");
my $cont;
    local $/ = undef;
    $cont = <fl>;
close fl;
my $reg_define = qr/[A-Za-z0-9_]+/;
my $reg_function = qr/[A-Za-z0-9_]+/;
my $reg_string_or_define = qr/(?:(?:$reg_define)|(?:"[^"]*"))/;

#load defines
while ($cont =~ /
                    ^\s* \#define \s+ ($reg_define) \s+ \"([^"]*)\"
    warn "$1 is #defined multiple times" if ($DEFINES{$1});
    $DEFINES{$1} = $2;

sub resolve_str_or_define($){
    my ($str) = @_;
    if ($str =~ s/^"// && $str =~ s/"$//){ #if starts and ends with a quote strip them off and return the str
        return $str;
    warn "Unable to resolve define: $str" if (! $DEFINES{$str});
    return $DEFINES{$str};
#parse commands
while ($cont =~ /
                    SWITCH_ADD_API \s* \( ([^,]+) #interface $1
                    ,\s* ($reg_string_or_define) # command $2
                    ,\s* ($reg_string_or_define) # command description $3
                    ,\s* ($reg_function) # function $4
                    ,\s* ($reg_string_or_define) # usage $5
        my ($interface,$command,$descr,$function,$usage) = ($1,$2,$3,$4,$5);
        $command = resolve_str_or_define($command);
        $descr = resolve_str_or_define($descr);
        $usage = resolve_str_or_define($usage);
        warn "Found a not command interface of: $interface for command: $command" if ($interface ne "commands_api_interface");
        print "$command -- $descr -- $usage\n";

Core Commands

Implemented in

Results of some status and listing commands are presented in comma delimited lists by default. Data returned from some modules may also contain commas, making it difficult to automate result processing. They may be able to be retrieved in an XML format by appending the string «as xml» to the end of the command string, or as json using «as json», or change the delimiter from comma to something else using «as delim ».

Compare an ip to an Access Control List

Usage: acl <ip> <list_name>

Alias: a means to save some keystrokes on commonly used commands.

Usage: alias add <alias> <command> | del [<alias>|*]


 freeswitch> alias add reloadall reloadacl reloadxml
 freeswitch> alias add unreg sofia profile internal flush_inbound_reg

You can add aliases that persist across restarts using the stickyadd argument:

 freeswitch> alias stickyadd reloadall reloadacl reloadxml
Only really works from the console, not fs_cli.

Execute an API command in a thread.

Usage: bgapi <command>[ <arg>]


Usage: complete add <word>|del [<word>|*]

Evaluate a conditional expression.

Usage: cond <expr> ? <true val> : <false val>

Operators supported by expr are:

== Equality
< Less than
> Greater than


Return true if first val is greater than the second

cond 5 > 3 ? true : false

Example in dialplan:

<action application="set" data="voicemail_authorized=${cond(${sip_authorized} == true ? true : false)}"/>

Slightly more complex example:

<action application="set" data="voicemail_authorized=${cond(${sip_acl_authed_by} == domains ? false : ${cond(${sip_authorized} == true ? true : false)})}"/>
The whitespace around the questionmark and colon are required since FS-5945. Before that, they were optional. If the spaces are missing, the cond function will return -ERR.

Check if a FreeSWITCH domain exists.

Usage: domain_exists <domain>

Eval (noop). Evaluates a string, expands variables. Those variables that are set only during a call session require the uuid of the desired session or else return «-ERR no reply».

Usage: eval [uuid:<uuid> ]<expression>


eval ${domain}
eval Hello, World!
Hello, World!
eval uuid:e72aff5c-6838-49a8-98fb-84c90ad840d9 ${channel-state}

Execute an API command with variable expansion.

Usage: expand [uuid:<uuid> ]<cmd> <args>


expand originate sofia/internal/1001%${domain} 9999

In this example the value of ${domain} is expanded. If the domain were, for example, «» then this command would be executed:

originate sofia/internal/1001% 9999

Send control messages to FreeSWITCH.

USAGE: fsctl
   api_expansion [on|off] |
   calibrate_clock |
   debug_level [level] |
   debug_sql |
   default_dtmf_duration [n] |
   flush_db_handles |
   hupall |
   last_sps |
   loglevel [level] |
   max_dtmf_duration [n] |
   max_sessions [n] |
   min_dtmf_duration [n] |
   min_idle_cpu [d] |
   pause [inbound|outbound] |
   pause_check [inbound|outbound] |
   ready_check |
   reclaim_mem |
   recover |
   resume [inbound|outbound] |
   save_history |
   send_sighup |
   shutdown [cancel|elegant|asap|now|restart] |
   shutdown_check |
   sps |
   sps_peak_reset |
   sql [start] |
   sync_clock |
   sync_clock_when_idle |
   threaded_system_exec |
   verbose_events [on|off]


Usage: fsctl api_expansion [on|off]

Toggles API expansion. With it off, no api functions can be expanded inside channel variables like ${show channels} This is a specific security mode that is not often used.


Usage: fsctl calibrate_clock

Runs an algorithm to compute how long it actuallymust sleep in order to sleep for a true 1ms.  It's only useful in older kernels that don't have timerfd.  In those older kernels FS auto detects that it needs to do perform that computation. This command just repeats the calibration.


Usage: fsctl debug_level [level]

Set the amount of debug information that will be posted to the log. 1 is less verbose while 9 is more verbose. Additional debug messages will be posted at the ALERT loglevel.

0 - fatal errors, panic

1 - critical errors, minimal progress at subsystem level

2 - non-critical errors

3 - warnings, progress messages

5 - signaling protocol actions (incoming packets, …)

7 - media protocol actions (incoming packets, …)

9 - entering/exiting functions, very verbatim progress


Usage: fsctl debug_sql

Toggle core SQL debugging messages on or off each time this command is invoked. Use with caution on busy systems. In order to see all messages issue the «logelevel debug» command on the fs_cli interface.


Usage: fsctl default_dtmf_duration [int]

int = number of clock ticks


fsctl default_dtmf_duration 2000

This example sets the default_dtmf_duration switch parameter to 250ms. The number is specified in clock ticks (CT) where duration (milliseconds) = CT / 8 or CT = duration * 8

The default_dtmf_duration specifies the DTMF duration to use on originated DTMF events or on events that are received without a duration specified. This value is bounded on the lower end by min_dtmf_duration and on the upper end by max_dtmf_duration. So max_dtmf_duration >= default_dtmf_duration >= min_dtmf_duration . This value can be set persistently in switch.conf.xml

To check the current value:

fsctl default_dtmf_duration 0

FS recognizes a duration of 0 as a status check. Instead of setting the value to 0, it simply returns the current value.


Usage: fsctl flush_db_handles

Flushes cached database handles from the core db handlers. FreeSWITCH reuses db handles whenever possible, but a heavily loaded FS system can accumulate a large number of db handles during peak periods while FS continues to allocate new db handles to service new requests in a FIFO manner. «fsctl flush_db_handles» closes db connections that are no longer needed to avoid exceeding connections to the database server.


Usage: fsctl hupall <clearing_type> dialed_ext <extension>

Disconnect existing calls to a destination and post a clearing cause.

For example, to kill an active call with normal clearing and the destination being extension 1000:

fsctl hupall normal_clearing dialed_ext 1000


Usage: fsctl last_sps

Query the actual sessions-per-second.

fsctl last_sps
+OK last sessions per second: 723987253 

(Your mileage might vary.)


Usage: fsctl loglevel [level]

Filter much detail the log messages will contain when displayed on the fs_cli interface. See mod_console for legal values of «level» and further discussion.



Usage: fsctl max_sessions [int]

Set how many simultaneous call sessions FS will allow. This value can be ascertained by load testing, but is affected by processor speed and quantity, network and disk bandwidth, choice of codecs, and other factors. See switch.conf.xml for the persistent setting max-sessions.


Usage: fsctl max_dtmf_duration [int]

Default = 192000 clock ticks


fsctl max_dtmf_duration 80000

This example sets the max_dtmf_duration switch parameter to 10,000ms (10 seconds). The integer is specified in clock ticks (CT) where CT / 8 = ms. The max_dtmf_duration caps the playout of a DTMF event at the specified duration. Events exceeding this duration will be truncated to this duration. You cannot configure a duration that exceeds this setting. This setting can be lowered, but cannot exceed 192000 (the default). This setting cannot be set lower than min_dtmf_duration. This setting can be set persistently in switch.conf.xml as max-dtmf-duration.

To query the current value:

fsctl max_dtmf_duration 0

FreeSWITCH recognizes a duration of 0 as a status check. Instead of setting the value to 0, it simply returns the current value.


Usage: fsctl min_dtmf_duration [int] 

Default = 400 clock ticks


fsctl min_dtmf_duration 800

This example sets the min_dtmf_duration switch parameter to 100ms. The integer is specified in clock ticks (CT) where CT / 8 = ms. The min_dtmf_duration specifies the minimum DTMF duration to use on outgoing events. Events shorter than this will be increased in duration to match min_dtmf_duration. You cannot configure a DTMF duration on a profile that is less than this setting. You may increase this value, but cannot set it lower than 400 (the default). This value cannot exceed max_dtmf_duration. This setting can be set persistently in switch.conf.xml as min-dtmf-duration.

It is worth noting that many devices squelch in-band DTMF when sending RFC 2833. Devices that squelch in-band DTMF have a certain reaction time and clamping time which can sometimes reach as high as 40ms, though most can do it in less than 20ms. As the shortness of your DTMF event duration approaches this clamping threshold, the risk of your DTMF being ignored as a squelched event increases. If your call is always IP-IP the entire route, this is likely not a concern. However, when your call is sent to the PSTN, the RFC 2833 DTMF events must be encoded in the audio stream. This means that other devices down the line (possibly a PBX or IVR that you are calling) might not hear DTMF tones that are long enough to decode and so will ignore them entirely. For this reason, it is recommended that you do not send DTMF events shorter than 80ms.

Checking the current value:

fsctl min_dtmf_duration 0

FreeSWITCH recognizes a duration of 0 as a status check. Instead of setting the value to 0, it simply returns the current value.


Usage: fsctl min_idle_cpu [int]

Allocates the minimum percentage of CPU idle time available to other processes to prevent FreeSWITCH from consuming all available CPU cycles.


fsctl min_idle_cpu 10

This allocates a minimum of 10% CPU idle time which is not available for processing by FS. Once FS reaches 90% CPU consumption it will respond with cause code 503 to additional SIP requests until its own usage drops below 90%, while reserving that last 10% for other processes on the machine.



Usage: fsctl pause [inbound|outbound] 

Pauses the ability to receive inbound or originate outbound calls, or both directions if the keyword is omitted. Executing fsctl pause inbound will also prevent registration requests from being processed. Executing fsctl pause outbound will result in the Critical log message «The system cannot create any outbound sessions at this time» in the FS log.

Use resume with the corresponding argument to restore normal operation.


Usage: fsctl pause_check [inbound|outbound]

Returns true if the specified mode is active.


fsctl pause_check inbound

indicates that inbound calls and registrations are paused. Use fsctl resume inbound to restore normal operation.

fsctl pause_check

indicates that both inbound and outbound sessions are paused. Use fsctl resume to restore normal operation.


Usage: fsctl ready_check

Returns true if the system is in the ready state, as opposed to awaiting an elegant shutdown or other not-ready state.


Usage: fsctl reclaim_mem


Usage: fsctl recover

Sends an endpoint–specific recover command to each channel detected as recoverable. This replaces “sofia recover” and makes it possible to have multiple endpoints besides SIP implement recovery.


Usage: fsctl resume [inbound|outbound] 

Resumes normal operation after pausing inbound, outbound, or both directions of call processing by FreeSWITCH.


fsctl resume inbound

Resumes processing of inbound calls and registrations. Note that this command always returns +OK, but the same keyword must be used that corresponds to the one used in the pause command in order to take effect.


Usage: fsctl save_history

Write out the command history in anticipation of executing a configuration that might crash FS. This is useful when debugging a new module or script to allow other developers to see what commands were executed before the crash.


Usage: fsctl send_sighup

Does the same thing that killing the FS process with -HUP would do without having to use the UNIX kill command. Useful in environments like Windows where there is no kill command or in cron or other scripts by using fs_cli -x «fsctl send_sighup» where the FS user process might not have privileges to use the UNIX kill command.


Usage: fsctl shutdown [asap|asap restart|cancel|elegant|now|restart|restart asap|restart elegant]

* cancel - discontinue a previous shutdown request. * elegant - wait for all traffic to stop, while allowing new traffic. * asap - wait for all traffic to stop, but deny new traffic. * now - shutdown FreeSWITCH immediately. * restart - restart FreeSWITCH immediately following the shutdown.

When giving «elegant», «asap» or «now» it's also possible to add the restart command:


Usage: fsctl shutdown_check

Returns true if FS is shutting down, or shutting down and restarting.


Usage: fsctl sps [int]

This changes the sessions-per-second limit from the value initially set in switch.conf


Usage: fsctl sync_clock

FreeSWITCH will not trust the system time. It gets one sample of system time when it first starts and uses the monotonic clock after that moment. You can sync it back to the current value of the system's real-time clock with fsctl sync_clock

Note: fsctl sync_clock immediately takes effect, which can affect the times on your CDRs. You can end up underbilling/overbilling, or even calls hungup before they originated. e.g. if FS clock is off by 1 month, then your CDRs will show calls that lasted for 1 month!

See fsctl sync_clock_when_idle which is much safer.


Usage: fsctl sync_clock_when_idle

Synchronize the FreeSWITCH clock to the host machine's real-time clock, but wait until there are 0 channels in use. That way it doesn't affect any CDRs.


Usage: fsctl verbose_events [on|off]

Enables verbose events. Verbose events have every channel variable in every event for a particular channel. Non-verbose events have only the pre-selected channel variables in the event headers.

See switch.conf.xml for the persistent setting of verbose-channel-events.||

Gets the value of a global variable. If the parameter is not provided then it gets all the global variables.

Usage: global_getvar [<varname>]

Sets the value of a global variable.

Usage: global_setvar <varname>=<value>


global_setvar outbound_caller_id=2024561000

Returns the bridge string defined in a call group.

Usage: group_call group@domain[+F|+A|+E]

+F will return the group members in a serial fashion separated by | (the pipe character)

+A (default) will return them in a parallel fashion separated by , (comma)

+E will return them in a enterprise fashion separated by :_: (colon underscore colon).

There is no space between the domain and the optional flag. See Groups in the XML User Directory for more information.

Please note: If you need to have outgoing user variables set in leg B, make sure you don't have dial-string and group-dial-string in your domain or dialed group variables list; instead set dial-string or group-dial-string in the default group of the user. This way group_call will return user/101 and user/ would set all your user variables to the leg B channel.

The B leg receives a new variable, dialed_group, containing the full group name.

Show help for all the API commands.

Usage: help

Performs a DNS lookup on a host name.

Usage: host_lookup <hostname>

Disconnect existing channels.

Usage: hupall <cause> [<variable> <value>]

All channels with <variable> set to <value> will be disconnected with <cause> code.


originate {foo=bar}sofia/internal/,sofia/internal/ &park
hupall normal_clearing foo bar

To hang up all calls on the switch indiscriminately:

hupall system_shutdown

Determine if a user is a member of a group.

Usage: in_group <user>[@<domain>] <group_name>

See if an IP is a LAN address.

Usage: is_lan_addr <ip>


Usage: json {"command" : "...", "data" : "..."}
> json {"command" : "status", "data" : ""}
{"command":"status","data":"","status":"success","response":{"systemStatus":"ready","uptime":{"years":0,"days":20,"hours":20,"minutes":37,"seconds":4,"milliseconds":254,"microseconds":44},"version":"1.6.9 -16-d574870 64bit","sessions":{"count":{"total":132,"active":0,"peak":2,"peak5Min":0,"limit":1000},"rate":{"current":0,"max":30,"peak":2,"peak5Min":0}},"idleCPU":{"used":0,"allowed":99.733333},"stackSizeKB":{"current":240,"max":8192}}}

Load external module

Usage: load <mod_name>


load mod_v8

Return MD5 hash for the given input data

Usage: md5 hash-key


md5 freeswitch-is-awesome

Check if module is loaded.

Usage: module_exists <module> 


module_exists mod_event_socket

Sleep for x number of milliseconds

Usage: msleep <number of milliseconds to sleep>

Manage Network Address Translation mapping.

Usage: nat_map [status|reinit|republish] | [add|del] <port> [tcp|udp] [sticky] | [mapping] <enable|disable>
  • status - Gives the NAT type, the external IP, and the currently mapped ports.
  • reinit - Completely re-initializes the NAT engine. Use this if you have changed routes or have changed your home router from NAT mode to UPnP mode.
  • republish - Causes FreeSWITCH to republish the NAT maps. This should not be necessary in normal operation.
  • mapping - Controls whether port mapping requests will be sent to the NAT (the command line option of -nonatmap can set it to disable on startup). This gives the ability of still using NAT for getting the public IP without opening the ports in the NAT.

Note: sticky makes the mapping stay across FreeSWITCH restarts. It gives you a permanent mapping.

Warning: If you have multiple network interfaces with unique IP addresses defined in sip profiles using the same port, nat_map *will* get confused when it tries to map the same ports for multiple profiles. Set up a static mapping between the public address and port and the private address and port in the sip_profiles to avoid this problem.

Evaluate a regex (regular expression).

Usage: regex <data>|<pattern>[|<subst string>][|(n|b)]
       regex m:/<data>/<pattern>[/<subst string>][/(n|b)]
       regex m:~<data>~<pattern>[~<subst string>][~(n|b)]

This command behaves differently depending upon whether or not a substitution string and optional flag is supplied:

  • If a subst is not supplied, regex returns either «true» if the pattern finds a match or «false» if not.
  • If a subst is supplied, regex returns the subst value on a true condition.
  • If a subst is supplied, on a false (no pattern match) condition regex returns:
    • the source string with no flag;
    • with the n flag regex returns null which forces the response «-ERR no reply» from regex;
    • with the b flag regex returns «false»

The regex delimiter defaults to the | (pipe) character. The delimiter may be changed to ~ (tilde) or / (forward slash) by prefixing the regex with m:


regex test1234|\d                  <== Returns "true"
regex m:/test1234/\d               <== Returns "true"
regex m:~test1234~\d               <== Returns "true"
regex test|\d                      <== Returns "false"
regex test1234|(\d+)|$1            <== Returns "1234"
regex sip:foo@bar.baz|^sip:(.*)|$1 <== Returns "foo@bar.baz"
regex testingonetwo|(\d+)|$1       <== Returns "testingonetwo" (no match)
regex m:~30~/^(10|20|40)$/~$1      <== Returns "30" (no match)
regex m:~30~/^(10|20|40)$/~$1~n    <== Returns "-ERR no reply" (no match)
regex m:~30~/^(10|20|40)$/~$1~b    <== Returns "false" (no match)

Logic in revision 14727 if the source string matches the result then the condition was false however there was a match and it is 1001.

regex 1001|/(^\d{4}$)/|$1

Reload a module.

Usage: reload <mod_name>

Reload Access Control Lists after modifying them in autoload_configs/acl.conf.xml and as defined in extensions in the user directory conf/directory/*.xml

Usage: reloadacl [reloadxml]

Reload conf/freeswitch.xml settings after modifying configuration files.

Usage: reloadxml

Display various reports, VERY useful for troubleshooting and confirming proper configuration of FreeSWITCH. Arguments can not be abbreviated, they must be specified fully.

Usage: show [
   aliases |
   api |
   application |
   bridged_calls |
   calls [count] |
   channels [count|like <match string>] |
   chat |
   codec |
   complete |
   detailed_bridged_calls |
   detailed_calls |
   dialplan |
   endpoint |
   file |
   interface_types |
   interfaces |
   management |
   modules |
   nat_map |
   registrations |
   say |
   tasks |
   timer |
   ] [as xml|as delim <delimiter>]

XML formatted:

show foo as xml

Change delimiter:

show foo as delim |


  • aliases – list defined command aliases
  • api – list api commands exposed by loadable modules
  • application – list applications exposed by loadable modules, notably mod_dptools
  • bridged_calls – deprecated, use «show calls»
  • calls [count] – list details of currently active calls; the keyword «count» eliminates the details and only prints the total count of calls
  • channels [count|like <match string>] – list current channels; see Channels vs Calls
    • count – show only the count of active channels, no details
    • like <match string> – filter results to include only channels that contain <match string> in uuid, channel name, cid_number, cid_name, presence data fields.
  • chat – list chat interfaces
  • codec – list codecs that are currently loaded in FreeSWITCH
  • complete – list command argument completion tables
  • detailed_bridged_calls – same as «show detailed_calls»
  • detailed_calls – like «show calls» but with more fields
  • dialplan – list dialplan interfaces
  • endpoint – list endpoint interfaces currently available to FS
  • file – list supported file format interfaces
  • interface_types – list all interface types with a summary count of each type of interface available
  • interfaces – enumerate all available interfaces by type, showing the module which exposes each interface
  • limits – list database limit interfaces
  • management – list management interfaces
  • module – enumerate modules and the path to each
  • nat_map – list Network Address Translation map
  • registrations – enumerate user extension registrations
  • say – enumerate available TTS (text-to-speech) interface modules with language supported
  • tasks – list FS tasks
  • timer – list timer modules

Tips For Showing Calls and Channels

The best way to get an understanding of all of the show calls/channels is to use them and observe the results. To display more fields:

  • show detailed_calls
  • show bridged_calls
  • show detailed_bridged_calls

These three take the expand on information shown by «show calls». Note that «show detailed_calls» replaces «show distinct_channels». It provides similar, but more detailed, information. Also note that there is no «show detailed_channels» command, however using «show detailed_calls» will yield the same net result: FreeSWITCH lists detailed information about one-legged calls and bridged calls by using «show detailed_calls», which can be quite useful while configuring and troubleshooting FS.

show channels like foo

to list only those channels of interest. The like directive filters on these fields:

* uuid * channel name * caller id name * caller id number * presence_data

NOTE: presence_data must be set during bridge or originate and not after the channel is established.|

Stop the FreeSWITCH program.

Usage: shutdown

This only works from the console. To shutdown FS from an API call or fs_cli, you should use «fsctl shutdown» which offers a number of options.

Shutdown from the console ignores arguments and exits immediately!


Show current FS status. Very helpful information to provide when asking questions on the mailing list or irc channel.

Usage: status
freeswitch@internal> status
UP 17 years, 20 days, 10 hours, 10 minutes, 31 seconds, 571 milliseconds, 721 microseconds
FreeSWITCH (Version 1.5.8b git 87751f9 2013-12-13 18:13:56Z 32bit) is ready  <!-- FS version -->
53987253 session(s) since startup                                            <!-- cumulative total number of channels created since FS started -->
127 session(s) - peak 127, last 5min 253                                     <!-- current number of active channels -->
55 session(s) per Sec out of max 60, peak 55, last 5min 253                  <!-- current channels per second created, max cps set in switch.conf.xml -->
1000 session(s) max                                                          <!-- set in switch.conf.xml -->
min idle cpu 0.00/97.71                                                      <!-- minimum reserved idle CPU time before refusing new calls, set in switch.conf.xml -->

Displays formatted time, converted to a specific timezone. See /usr/share/zoneinfo/ for the standard list of Linux timezones.

Usage: strftime_tz <timezone> [format_string]


strftime_tz US/Eastern %Y-%m-%d %T

Unload external module.

Usage: unload <mod_name>

Show version of the switch

Usage: version [short]


freeswitch@internal> version
FreeSWITCH Version 1.5.8b+git~20131213T181356Z~87751f9eaf~32bit (git 87751f9 2013-12-13 18:13:56Z 32bit)
freeswitch@internal> version short

Write active xml tree or specified branch to stdout.

Usage: xml_locate [root | <section> | <section> <tag> <tag_attr_name> <tag_attr_val>]

xml_locate root will return all XML being used by FreeSWITCH

xml_locate <section>: Will return the XML corresponding to the specified <section>

xml_locate directory
xml_locate configuration
xml_locate dialplan
xml_locate phrases


xml_locate directory domain name
xml_locate configuration configuration name ivr.conf

Wrap another API command in XML.

Usage: xml_wrap <command> <args>

Call Management Commands

Deprecated. See uuid_break.

Creates a new UUID and returns it as a string.

Usage: create_uuid

Originate a new call.

   Usage: originate <call_url> <exten>|&<application_name>(<app_args>) [<dialplan>] [<context>] [<cid_name>] [<cid_num>] [<timeout_sec>]

FreeSWITCH will originate a call to <call_url> as Leg A. If that leg supervises within 60 seconds FS will continue by searching for an extension definition in the specified dialplan for <exten> or else execute the application that follows the & along with its arguments.

* <call_url> URL you are calling. For more info on sofia SIP URL syntax see: FreeSwitch Endpoint Sofia * Destination, one of:

  • <exten> Destination number to search in dialplan; note that registered extensions will fail this way, use &bridge(user/xxxx) instead
  • &<application_name>(<app_args>)
    • «&» indicates what follows is an application name, not an exten
    • (<app_args>) is optional (not all applications require parameters, e.g. park)
    • The most commonly used application names include:

park, bridge, javascript/lua/perl, playback (remove mod_native_file).

  • Note: Use single quotes to pass arguments with spaces, e.g. '&lua(test.lua arg1 arg2)'
  • Note: There is no space between & and the application name

* <dialplan> Defaults to 'XML' if not specified. * <context> Defaults to 'default' if not specified. * <cid_name> CallerID name to send to Leg A. * <cid_num> CallerID number to send to Leg A. * <timeout_sec> Timeout in seconds; default = 60 seconds.


These variables can be prepended to the dial string inside curly braces and separated by commas. Example:

originate {sip_auto_answer=true,return_ring_ready=false}user/1001 9198

Variables within braces must be separated by a comma.

* group_confirm_key * group_confirm_file * forked_dial * fail_on_single_reject * ignore_early_media - must be defined on Leg B in bridge or originate command to stop remote ringback from being heard by Leg A * return_ring_ready * originate_retries * originate_retry_sleep_ms * origination_caller_id_name * origination_caller_id_number * originate_timeout * sip_auto_answer

Description of originate's related variables |

You can call a locally registered sip endpoint 300 and park the call like so Note that the «example» profile used here must be the one to which 300 is registered. Also note the use of % instead of @ to indicate that it is a registered extension.

   originate sofia/example/300%pbx.internal &park()

Or you could instead connect a remote sip endpoint to extension 8600

   originate sofia/example/ 8600

Or you could instead connect a remote SIP endpoint to another remote extension

   originate sofia/example/ &bridge(sofia/example/

Or you could even run a Javascript application test.js

   originate sofia/example/ &javascript(test.js)

To run a javascript with arguments you must surround it in single quotes.

   originate sofia/example/ '&javascript(test.js myArg1 myArg2)'

Setting channel variables to the dial string

   originate {ignore_early_media=true}sofia/ 15555551212

Setting SIP header variables to send to another FS box during originate

   originate {sip_h_X-varA=111,sip_h_X-varB=222}sofia/ 15555551212

Note: you can set any channel variable, even custom ones. Use single quotes to enclose values with spaces, commas, etc.

   originate {my_own_var=my_value}sofia/ 15555551212
   originate {my_own_var='my value'}sofia/ 15555551212

If you need to fake the ringback to the originated endpoint try this:

   originate {ringback=\'%(2000,4000,440.0,480.0)\'}sofia/example/ &bridge(sofia/example/

To specify a parameter to the Leg A call and the Leg B bridge application:

originate {'origination_caller_id_number=2024561000'}sofia/gateway/ &bridge(['effective_caller_id_number=7036971379']sofia/gateway/


If you need to make originate return immediately when the channel is in «Ring-Ready» state try this:

   originate {return_ring_ready=true}sofia/gateway/someprovider/919246461929 &socket(' async full')

More info on return_ring_ready

You can even set music on hold for the ringback if you want:

   originate {ringback=\'/path/to/music.wav\'}sofia/gateway/name/number &bridge(sofia/gateway/siptoshore/12425553741)

You can originate a call in the background (asynchronously) and playback a message with a 60 second timeout.

   bgapi originate {ignore_early_media=true,originate_timeout=60}sofia/gateway/name/number &playback(message)

You can specify the UUID of an originated call by doing the following:

* Use create_uuid to generate a UUID to use. * This will allow you to kill an originated call before it is answered by using uuid_kill. * If you specify origination_uuid it will remain the UUID for the answered call leg for the whole session.

    originate {origination_uuid=...}user/

Here's an example of originating a call to the echo conference (an external sip URL) and bridging it to a local user's phone:

   originate sofia/internal/ &bridge(user/105@default)

Here's an example of originating a call to an extension in a different context than 'default' (required for the FreePBX which uses context_1, context_2, etc.):

   originate sofia/internal/ 3001 xml context_3

You can also originate to multiple extensions as follows:

   originate user/1001,user/1002,user/1003 &park()

To put an outbound call into a conference at early media, either of these will work (they are effectively the same thing)

   originate sofia/example/ &conference(conf_uuid-TEST_CON)
   originate sofia/example/ conference:conf_uuid-TEST_CON inline

See Misc._Dialplan_Tools_InlineDialplan for more detail on 'inline' Dialplans

An example of using loopback and inline on the A-leg can be found in this mailing list post |

Pause <uuid> playback of recorded media that was started with uuid_broadcast.

Usage: pause <uuid> <on|off>

Turning pause «on» activates the pause function, i.e. it pauses the playback of recorded media. Turning pause «off» deactivates the pause function and resumes playback of recorded media at the same point where it was paused.

Note: always returns -ERR no reply when successful; returns -ERR No such channel! when uuid is invalid.

Answer a channel

Usage: uuid_answer <uuid>

Adjust the audio levels on a channel or mute (read/write) via a media bug.

Usage: uuid_audio <uuid> [start [read|write] [[mute|level] <level>]|stop]

<level> is in the range from -4 to 4, 0 being the default value.

Level is required for both mute|level params:

freeswitch@internal> uuid_audio 0d7c3b93-a5ae-4964-9e4d-902bba50bd19 start write mute <level>
freeswitch@internal> uuid_audio 0d7c3b93-a5ae-4964-9e4d-902bba50bd19 start write level <level>

(This command behaves funky. Requires further testing to vet all arguments. - JB)

Seee Also


Break out of media being sent to a channel. For example, if an audio file is being played to a channel, issuing uuid_break will discontinue the media and the call will move on in the dialplan, script, or whatever is controlling the call.

Usage: uuid_break <uuid> [all]

If the all flag is used then all audio files/prompts/etc. that are queued up to be played to the channel will be stopped and removed from the queue, otherwise only the currently playing media will be stopped.

Bridge two call legs together.

Usage: uuid_bridge <uuid> <other_uuid>

uuid_bridge needs at least any one leg to be in the answered state. If, for example, one channel is parked and another channel is actively conversing on a call, executing uuid_bridge on these 2 channels will drop the existing call and bridge together the specified channels.

Execute an arbitrary dialplan application, typically playing a media file, on a specific uuid. If a filename is specified then it is played into the channel(s). To execute an application use «app::args» syntax.

Usage: uuid_broadcast <uuid> <path> [aleg|bleg|both]

Execute an application on a chosen leg(s) with optional hangup afterwards:

Usage: uuid_broadcast <uuid> app[![hangup_cause]]::args [aleg|bleg|both]


 uuid_broadcast 336889f2-1868-11de-81a9-3f4acc8e505e sorry.wav both
 uuid_broadcast 336889f2-1868-11de-81a9-3f4acc8e505e say::en\snumber\spronounced\s12345 aleg
 uuid_broadcast 336889f2-1868-11de-81a9-3f4acc8e505e say!::en\snumber\spronounced\s12345 aleg
 uuid_broadcast 336889f2-1868-11de-81a9-3f4acc8e505e say!user_busy::en\snumber\spronounced\s12345 aleg
 uuid_broadcast 336889f2-1868-11de-81a9-3f4acc8e505e playback!user_busy::sorry.wav aleg

List the media bugs on channel. Output is formatted as XML.

Usage: uuid_buglist <uuid>

Send a chat message.

Usage: <uuid> <text>

If the endpoint associated with the session <uuid> has a receive_event handler, this message gets sent to that session and is interpreted as an instant message.

Debug media, either audio or video.

Usage: <uuid> <read|write|both|vread|vwrite|vboth> <on|off>

Use «read» or «write» for the audio direction to debug, or «both» for both directions. And prefix with v for video media.

uuid_debug_media emits a HUGE amount of data. If you invoke this command from fs_cli, be prepared.
R sofia/internal/1003@ b= 172 pt=0 ts=2981605109 m=0
W sofia/internal/1003@ b= 172 pt=0 ts=12212960 m=0
R sofia/internal/1003@ b= 172 pt=0 ts=2981605269 m=0
W sofia/internal/1003@ b= 172 pt=0 ts=12213120 m=0

Read Format

«R %s b=%4ld %s:%u %s:%u %s:%u pt=%d ts=%u m=%d\n»

where the values are:

  • switch_channel_get_name(switch_core_session_get_channel(session)),
  • (long) bytes,
  • my_host, switch_sockaddr_get_port(rtp_session→local_addr),
  • old_host, rtp_session→remote_port,
  • tx_host, switch_sockaddr_get_port(rtp_session→from_addr),
  • rtp_session→,
  • ntohl(rtp_session→recv_msg.header.ts),
  • rtp_session→recv_msg.header.m

Write Format

«W %s b=%4ld %s:%u %s:%u %s:%u pt=%d ts=%u m=%d\n»

where the values are:

  • switch_channel_get_name(switch_core_session_get_channel(session)),
  • (long) bytes,
  • my_host, switch_sockaddr_get_port(rtp_session→local_addr),
  • old_host, rtp_session→remote_port,
  • tx_host, switch_sockaddr_get_port(rtp_session→from_addr),
  • send_msg→,
  • ntohl(send_msg→header.ts),
  • send_msg→header.m);

Deflect an answered SIP call off of FreeSWITCH by sending the REFER method

Usage: uuid_deflect <uuid> <sip URL>

uuid_deflect waits for the final response from the far end to be reported. It returns the sip fragment from that response as the text in the FreeSWITCH response to uuid_deflect. If the far end reports the REFER was successful, then FreeSWITCH will issue a bye on the channel.


  Content-Type: api/response
  Content-Length: 30
  +OK:SIP/2.0 486 Busy Here


Displace the audio for the target <uuid> with the specified audio <file>.

Usage: uuid_displace <uuid> [start|stop] <file> [<limit>] [mux]


  • uuid = Unique ID of this call (see 'show channels')
  • start|stop = Start or stop this action
  • file = path to an audio source (.wav file, shoutcast stream, etc…)
  • limit = limit number of seconds before terminating the displacement
  • mux = multiplex; mix the original audio together with 'file', i.e. both parties can still converse while the file is playing (if the level is not too loud)

To specify the 5th argument 'mux' you must specify a limit; if no time limit is desired on playback, then specify 0.

cli> uuid_displace 1a152be6-2359-11dc-8f1e-4d36f239dfb5 start /sounds/test.wav 60
cli> uuid_displace 1a152be6-2359-11dc-8f1e-4d36f239dfb5 stop /sounds/test.wav

Updates the display on a phone if the phone supports this. This works on some SIP phones right now including Polycom and Snom.

Usage: <uuid> name|number

Note the pipe character separating the Caller ID name and Caller ID number.

This command makes the phone re-negotiate the codec. The SIPRTP Packet Size should be 0.020 seconds. If it is set to 0.030 on the Cisco SPA series phones it causes a DTMF lag. When DTMF keys are pressed on the phone they are can be seen on the fs_cli 4-6 seconds late.


freeswitch@sidious> uuid_display f4053af7-a3b9-4c78-93e1-74e529658573 Fred Jones|1001
+OK Success


Transfer each leg of a call to different destinations.

Usage: <uuid> <A-dest-exten>[/<A-dialplan>][/<A-context>] <B-dest-exten>[/<B-dialplan>][/<B-context>]

Dumps all variable values for a session.

Usage: uuid_dump <uuid> [format]

Format options: txt (default, may be omitted), XML, JSON, plain

Stops the process of ignoring early media, i.e. if ignore_early_media=true, this stops ignoring early media coming from Leg B and responds normally.

Usage: uuid_early_ok <uuid>

Checks whether a given UUID exists.

Usage: uuid_exists <uuid>

Returns true or false.

Flush queued DTMF digits

Usage: uuid_flush_dtmf <uuid>

Manage the audio being played into a channel from a sound file

Usage: uuid_fileman <uuid> <cmd:val>

Commands are:

  • speed:<+[step]>|←[step]>
  • volume:<+[step]>|←[step]>
  • pause (toggle)
  • stop
  • truncate
  • restart
  • seek:<+[milliseconds]>|←[milliseconds]> (1000ms = 1 second, 10000ms = 10 seconds.)

Example to seek forward 30 seconds:

uuid_fileman 0171ded1-2c31-445a-bb19-c74c659b7d08 seek:+3000

(Or use the current channel via ${uuid}, e.g. in a bind_digit_action)

The 'pause' argument is a toggle: the first time it is invoked it will pause playback, the second time it will resume playback.

Get a variable from a channel.

Usage: uuid_getvar <uuid> <varname>

Place a channel on hold.


uuid_hold <uuid>           place a call on hold
uuid_hold off <uuid>       switch off on hold
uuid_hold toggle <uuid>    toggles call-state based on current call-state

Reset a specific <uuid> channel.

Usage: uuid_kill <uuid> [cause]

If no cause code is specified, NORMAL_CLEARING will be used.

Apply or change limit(s) on a specified uuid.

Usage: uuid_limit <uuid> <backend> <realm> <resource> [<max>[/interval]] [number [dialplan [context]]]

See also mod_dptools: Limit

Reinvite FreeSWITCH out of the media path:

Usage: uuid_media [off] <uuid>

Reinvite FreeSWITCH back in:

Usage: uuid_media <uuid>

API command to tell a channel to send a re-invite with optional list of new codecs to be renegotiated.

Usage: uuid_media_reneg <uuid> <=><codec string>

Example: Adding =PCMU makes the offered codec string absolute.

Park call

Usage: uuid_park <uuid>

The specified channel will be parked and the other leg of the call will be disconnected.

Pre–answer a channel.

Usage: uuid_preanswer <uuid>

See Also: Misc._Dialplan_Tools_pre_answer

Pre-process Channel

Usage: uuid_preprocess <uuid>


Usage: uuid_recv_dtmf <uuid> <dtmf_data>


Send DTMF digits to <uuid>

Usage: uuid_send_dtmf <uuid> <dtmf digits>[@<tone_duration>]

Use the character w for a .5 second delay and the character W for a 1 second delay.

Default tone duration is 2000ms .

Send info to the endpoint

Usage: uuid_send_info <uuid>
Usage: uuid_session_heartbeat <uuid> [sched] [0|<seconds>]

Set a variable on a channel. If value is omitted, the variable is unset.

Usage: uuid_setvar <uuid> <varname> [value]

Set multiple vars on a channel.

Usage: uuid_setvar_multi <uuid> <varname>=<value>[;<varname>=<value>[;...]]

This command directs FreeSWITCH to remove itself from the SIP signaling path if it can safely do so.

Usage: uuid_simplify <uuid>

Execute this API command to instruct macro to inspect the Leg A and Leg B network addresses. If they are both hosted by the same switch as a result of a transfer or forwarding loop across a number of macro systems the one executing this command will remove itself from the SIP and media path and restore the endpoints to their local macro to shorten the network path. This is particularly useful in large distributed macro installations.

For example, suppose a call arrives at a macro box in Los Angeles, is answered, then forwarded to a macro box in London, answered there and then forwarded back to Los Angeles. The London switch could execute uuid_simplify to tell its local switch to examine both legs of the call to determine that they could be hosted by the Los Angeles switch since both legs are local to it. Alternatively, setting sip_auto_simplify to true either globally in vars.xml or as part of a dailplan extension would tell FS to perform this check for each call when both legs supervise. 

Transfers an existing call to a specific extension within a <dialplan> and <context>. Dialplan may be «xml» or «directory».

Usage: uuid_transfer <uuid> [-bleg|-both] <dest-exten> [<dialplan>] [<context>]

The optional first argument will allow you to transfer both parties (-both) or only the party to whom <uuid> is talking.(-bleg). Beware that -bleg actually means «the other leg», so when it is executed on the actual B leg uuid it will transfer the actual A leg that originated the call and disconnect the actual B leg.

NOTE: if the call has been bridged, and you want to transfer either side of the call, then you will need to use <action application=«set» data=«hangup_after_bridge=false»/> (or the API equivalent). If it's not set, transfer doesn't really work as you'd expect, and leaves calls in limbo.

Send hold indication upstream:

uuid_phone_event <uuid> hold|talk

Record/Playback Commands

Record the audio associated with the given UUID into a file. The start command causes FreeSWITCH to start mixing all call legs together and saves the result as a file in the format that the file's extension dictates. (if available) The stop command will stop the recording and close the file. If media setup hasn't yet happened, the file will contain silent audio until media is available. Audio will be recorded for calls that are parked. The recording will continue through the bridged call. If the call is set to return to park after the bridge, the bug will remain on the call, but no audio is recorded until the call is bridged again. (TODO: What if media doesn't flow through FreeSWITCH? Will it re-INVITE first? Or do we just not get the audio in that case?)


uuid_record <uuid> [start|stop|mask|unmask] <path> [<limit>]

Where limit is the max number of seconds to record.

If the path is not specified on start it will default to the channel variable «sound_prefix» or FreeSWITCH base_dir when the «sound_prefix» is empty.

You may also specify «all» for path when stop is used to remove all for this uuid

«stop» command must be followed by <path> option.

«mask» will mask with silence part of the recording beginning when the mask argument is executed by this command. see

«unmask» will stop the masking and continue recording live audio normally.

See record's related variables

you will also want to see mod_dptools: record_session

Limit Commands

More information is available at Limit commands

Reset a limit backend.

Retrieve status from a limit backend.

Retrieve usage for a given resource.

Manually decrease a resource usage by one.

Reset the interval counter to zero prior to the start of the next interval.

Miscellaneous Commands

Execute a system command in the background.

Usage: bg_system <command>

Echo input back to the console

Usage: echo <text to echo>


echo This text will appear
This text will appear

Tests whether filename exists.

file_exists filename


freeswitch> file_exists /tmp/real_file

freeswitch> file_exists /tmp/missing_file

Example dialplan usage:

<extension name="play-news-announcements">
  <condition expression="${file_exists(${sounds_dir}/news.wav)}" expression="true"/>
    <action application="playback" data="${sounds_dir}/news.wav"/>
    <anti-action application="playback" data="${soufnds_dir}/no-news-is-good-news.wav"/>
file_exists tests whether FreeSWITCH can see the file, but the file may still be unreadable because of restrictive permissions.


Checks to see if a user exists. Matches user tags found in the directory, similar to user_exists, but returns an XML representation of the user as defined in the directory (like the one shown in user_exists).

Usage: find_user_xml <key> <user> <domain>

<key> references a key specified in a directory's user tag

<user> represents the value of the key

<domain> is the domain to which the user is assigned.

Lists Users configured in Directory


list_users [group <group>] [domain <domain>] [user <user>] [context <context>]


freeswitch@localhost> list_users group default

2000|default||default|sofia/internal/sip:2000@|techsupport|B#-Test 2000|2000
2001|default||default|sofia/internal/sip:2001@;rinstance=8e2c8b86809acf2a|techsupport|Test 2001|2001
2002|default||default|error/user_not_registered|techsupport|Test 2002|2002
2003|default||default|sofia/internal/sip:2003@|techsupport|Test 2003|2003
2004|default||default|error/user_not_registered|techsupport|Test 2004|2004


Search filters can be combined:

freeswitch@localhost> list_users group default user 2004

2004|default||default|error/user_not_registered|techsupport|Test 2004|2004


Schedule an API call in the future.


sched_api [+@]<time> <group_name> <command_string>[&]

<time> is the UNIX timestamp at which the command should be executed. If it is prefixed by +, <time> specifies the number of seconds to wait before executing the command. If prefixed by @, it will execute the command periodically every <time> seconds; for the first instance it will be executed after <time> seconds.

<group_name> will be the value of «Task-Group» in generated events. «none» is the proper value for no group. If set to UUID of channel (example: ${uuid}), task will automatically be unscheduled when channel hangs up.

<command_string> is the command to execute at the scheduled time.

A scheduled task or group of tasks can be revoked with sched_del or unsched_api.

You could append the «&» symbol to the end of the line to executed this command in its own thread.


sched_api +1800 none originate sofia/internal/1000%${sip_profile} &echo()
sched_api @600 check_sched log Periodic task is running...
sched_api +10 ${uuid} chat verto|||Hello World  

Play a <path> file to a specific <uuid> call in the future.


sched_broadcast [[+]<time>|@time] <uuid> <path> [aleg|bleg|both]

Schedule execution of an application on a chosen leg(s) with optional hangup:

sched_broadcast [+]<time> <uuid> app[![hangup_cause]]::args [aleg|bleg|both]

<time> is the UNIX timestamp at which the command should be executed. If it is prefixed by +, <time> specifies the number of seconds to wait before executing the command. If prefixed by @, it will execute the command periodically every <time> seconds; for the first instance it will be executed after <time> seconds.


sched_broadcast +60 336889f2-1868-11de-81a9-3f4acc8e505e commercial.wav both
sched_broadcast +60 336889f2-1868-11de-81a9-3f4acc8e505e say::en\snumber\spronounced\s12345 aleg

Removes a prior scheduled group or task ID


sched_del <group_name|task_id>

The one argument can either be a group of prior scheduled tasks or the returned task-id from sched_api.

sched_transfer, sched_hangup and sched_broadcast commands add new tasks with group names equal to the channel UUID. Thus, sched_del with the channel UUID as the argument will remove all previously scheduled hangups, transfers and broadcasts for this channel.


sched_del my_group
sched_del 2

Schedule a running call to hangup.


sched_hangup [+]<time> <uuid> [<cause>]
sched_hangup +0 is the same as uuid_kill

Schedule a transfer for a running call.


sched_transfer [+]<time> <uuid> <target extension> [<dialplan>] [<context>]

Executes a STUN lookup.


stun <stunserver>[:port]



Execute a system command.


system <command>

The <command> is passed to the system shell, where it may be expanded or interpreted in ways you don't expect. This can lead to security bugs if you're not careful. For example, the following command is dangerous:

<action application="system" data="log_caller_name ${caller_id_name}" />

If a malicious remote caller somehow sets his caller ID name to «; rm -rf /» you would unintentionally be executing this shell command:

log_caller_name; rm -rf /

This would be a Bad Thing.

Runs a test to see how bad timer jitter is. It runs the test <count> times if specified, otherwise it uses the default count of 10, and tries to sleep for mss microseconds. It returns the actual timer duration along with an average.


time_test <mss> [count]


time_test 100 5

test 1 sleep 100 99
test 2 sleep 100 97
test 3 sleep 100 96
test 4 sleep 100 97
test 5 sleep 100 102
avg 98

Runs a test to see how bad timer jitter is. Unlike time_test, this uses the actual FreeSWITCH timer infrastructure to do the timer test and exercises the timers used for call processing.


timer_test <10|20|40|60|120> [<1..200>] [<timer_name>]

The first argument is the timer interval.

The second is the number of test iterations.

The third is the timer name; «show timers» will give you a list.


timer_test 20 3

Avg: 16.408ms Total Time: 49.269ms

2010-01-29 12:01:15.504280 [CONSOLE] mod_commands.c:310 Timer Test: 1 sleep 20 9254
2010-01-29 12:01:15.524351 [CONSOLE] mod_commands.c:310 Timer Test: 2 sleep 20 20042
2010-01-29 12:01:15.544336 [CONSOLE] mod_commands.c:310 Timer Test: 3 sleep 20 19928

Start Tone Detection on a channel.


tone_detect <uuid> <key> <tone_spec> [<flags> <timeout> <app> <args>] <hits>

<uuid> is required when this is executed as an api call; as a dialplan app the uuid is implicit as part of the channel variables

<key> is an arbitrary name that identifies this tone_detect instance; required

<tone_spec> frequencies to detect; required

<flags> 'r' or 'w' to specify which direction to monitor

<timeout> duration during which to detect tones;

0 = detect forever

+time = number of milliseconds after tone_detect is executed

time = absolute time to stop in seconds since The Epoch (1 January, 1970)

<app> FS application to execute when tone_detect is triggered; if app is omitted, only an event will be returned

<args> arguments to application enclosed in single quotes

<hits> the number of times tone_detect should be triggered before executing the specified app

Once tone_detect returns a result, it will not trigger again until reset. Reset tone_detect by calling tone_detect <key> with no additional arguments to reactivate the previously specified tone_detect declaration.

See also

Unschedule a previously scheduled api command.


unsched_api <task_id>


url_decode <string>

Url encode a string.


url_encode <string>

Retrieves user information (parameters or variables) as defined in the FreeSWITCH user directory.


user_data <user>@<domain> <attr|var|param> <name>

<user> is the user's id

<domain> is the user's domain

<attr|var|param> specifies whether the requested data is contained in the «variables» or «parameters» section of the user's record

<name> is the name (key) of the variable to retrieve


user_data 1000@ param password

will return a result of 1234, and

user_data 1000@ var accountcode

will return a result of 1000 from the example user shown in user_exists, and

user_data 1000@ attr id

will return the user's actual alphanumeric ID (i.e. «john») when number-alias=«1000» was set as an attribute for that user.

Checks to see if a user exists. Matches user tags found in the directory and returns either true/false:


user_exists <key> <user> <domain>

<key> references a key specified in a directory's user tag

<user> represents the value of the key

<domain> is the domain to which the user belongs


user_exists id 1000

will return true where there exists in the directory a user with a key called id whose value equals 1000:

    <user id="1000" randomvar="45">
          <param name="password" value="1234"/>
          <param name="vm-password" value="1000"/>
          <variable name="accountcode" value="1000"/>
          <variable name="user_context" value="default"/>
          <variable name="effective_caller_id_name" value="Extension 1000"/>
          <variable name="effective_caller_id_number" value="1000"/>

In the above example, we also could have tested for randomvar:

user_exists randomvar 45

And we would have received the same true result, but:

user_exists accountcode 1000


user_exists vm-password 1000

Would have returned false.

  • freeswitch/mod/mod_commands.txt
  • Последние изменения: 2018/12/06