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Event Socket Outbound


Discussion and examples for using the Outbound Event Socket.

mod_event_socket's "socket" app is similar to the network based fast-agi (FAGI) of Asterisk. This application makes an outbound TCP connection to the specified ip:port and the other end can control the call in similar ways to an Asterisk AGI only the dialect is much different and it supports async operations e.g. telling the call to play foo.wav and getting an immediate return to listen for DTMF as the file plays rather than block till the file is over.

This document focuses on the outbound operation mode of mod_event_socket, but take note that this module also operates in inbound mode.

When you call outbound socket, FS automatically puts the call in PARK. You can validate that by monitoring the CHANNEL_PARK event.

When you use "sync" mode, many things will not works as you may expect. Particularly, you will not get DTMFs, etc. Please check "async" option.

Click to expand Table of Contents


* | *
* FreeSWITCH™ | mod_event_socket *
* | *
* *
|| ||
\/ \/
****************** ******************
* Server A * * Server B *
* * * *
****************** ******************


The data syntax to invoke the event socket outbound from FreeSWITCH in the dialplan is <ip>:<port> [<keywords>]

<action application="socket" data="address:port [async|full]"/>

Since revision git-8c794ac on 14/03/2012 you can also connect to IPv6 addresses. When using IPv6 addresses the port parameter is required: <action application="socket" data="::1:8021"/> connects to ::1 on port 8021. Since this revision hostnames resolving to IPv6 addresses can be used.



The "async" keyword indicates that all commands will return instantly, allowing voice processing to proceed and making it possible to monitor the socket for events while the stack of commands are executing.

If the async keyword is absent then all calls on this channel will block until the command has finished.


The "full" keyword indicates that the other end will have the full command set for event_socket. This is the same command set on an event socket inbound connection so you can execute API commands, get global events, etc.

If the "full" keyword is absent the command set and events are limited to that particular call.



If this variable is set to true, the dialplan will resume execution with the next action after the call to the socket application. This can be used for example to allow you to do something intelligent in the dialplan if your IVR application dies in an unclean way. If there is a bridge active when the disconnect happens, it is killed. To do this from your application after the socket is already connected, issue the resume command.


Here are examples of how to use it in the dialplan.

<action application="socket" data=""/>
<action application="socket" data=" async"/>
<action application="socket" data=" full"/>
<action application="socket" data=" async full"/>

Event Lock

If you are using async, you need to pay attention to potential race condition. The commands you send may not execute in sequential order. You may force the command to wait by setting event lock until the critical or long-running command finishes:

sendmsg call-command: set execute-app-name: foo=bar\n\n event-lock:true

  You may obtain the same result (setting event-lock to true) in scripting using the setEventLock method of connection object:


Server Examples

There are several example Perl scripts that use the ESL library. Here's good one to review:


To run it you will need to compile ESL and perlmod. More information is available on the ESL page.

Using Netcat

Here is a small tutorial on how to use outbound event socket:

This was used to call this via the dialplan:

<action application="socket" data=" async full"/>

Now lets make netcat listen for connections from our call.

Linux :

nc -v -l -p 8084

On CentOS 5.3 my syntax is this:

nc -v -l 8084

On Windows :

nc -v -L -n -p 8084

Connection from port 8084 [tcp/*] accepted

Once the connection is accepted you type:


Note: The command 'connect' is the first command at your ESL server side to send to FreeSWITCH side.
If without 'connect', FreeSWITCH side would response nothing and waiting.
If first command is not 'connect', FreeSWITCH also output nothing and waiting.

Once you do this you'll be presented the name value pairs of everything related to this call including all variables.


Expand source

Channel-Username: 1001
Channel-Dialplan: XML
Channel-Caller-ID-Name: 1001
Channel-Caller-ID-Number: 1001
Channel-Destination-Number: 886
Channel-Unique-ID: 40117b0a-186e-11dd-bbcd-7b74b6b4d31e
Channel-Source: mod_sofia
Channel-Context: default
Channel-Channel-Name: sofia/default/1001%4010.0.1.100
Channel-Profile-Index: 1
Channel-Channel-Created-Time: 1209749769132614
Channel-Channel-Answered-Time: 0
Channel-Channel-Hangup-Time: 0
Channel-Channel-Transfer-Time: 0
Channel-Screen-Bit: yes
Channel-Privacy-Hide-Name: no
Channel-Privacy-Hide-Number: no
Channel-State: CS_EXECUTE
Channel-State-Number: 4
Channel-Name: sofia/default/1001%4010.0.1.100
Unique-ID: 40117b0a-186e-11dd-bbcd-7b74b6b4d31e
Call-Direction: inbound
Answer-State: early
Channel-Read-Codec-Name: G722
Channel-Read-Codec-Rate: 16000
Channel-Write-Codec-Name: G722
Channel-Write-Codec-Rate: 16000
Caller-Username: 1001
Caller-Dialplan: XML
Caller-Caller-ID-Name: 1001
Caller-Caller-ID-Number: 1001
Caller-Destination-Number: 886
Caller-Unique-ID: 40117b0a-186e-11dd-bbcd-7b74b6b4d31e
Caller-Source: mod_sofia
Caller-Context: default
Caller-Channel-Name: sofia/default/1001%4010.0.1.100
Caller-Profile-Index: 1
Caller-Channel-Created-Time: 1209749769132614
Caller-Channel-Answered-Time: 0
Caller-Channel-Hangup-Time: 0
Caller-Channel-Transfer-Time: 0
Caller-Screen-Bit: yes
Caller-Privacy-Hide-Name: no
Caller-Privacy-Hide-Number: no
variable_sip_authorized: true
variable_sip_mailbox: 1001
variable_sip_auth_username: 1001
variable_mailbox: 1001
variable_user_name: 1001
variable_accountcode: 1001
variable_user_context: default
variable_effective_caller_id_name: Extension%201001
variable_effective_caller_id_number: 1001
variable_sip_from_user: 1001
variable_sip_from_uri: 1001%4010.0.1.100
variable_sip_from_user_stripped: 1001
variable_sip_from_tag: wrgb4s5idf
variable_sofia_profile_name: default
variable_sip_req_params: user%3Dphone
variable_sip_req_user: 886
variable_sip_req_uri: 886%4010.0.1.100
variable_sip_to_params: user%3Dphone
variable_sip_to_user: 886
variable_sip_to_uri: 886%4010.0.1.100
variable_sip_contact_params: line%3Dnc7obl5w
variable_sip_contact_user: 1001
variable_sip_contact_port: 2048
variable_sip_contact_uri: 1001%4010.0.1.241%3A2048
variable_channel_name: sofia/default/1001%4010.0.1.100
variable_sip_call_id: 3c2bb21af10b-ogphkonpwqet
variable_sip_user_agent: snom300/7.1.30
variable_sip_via_port: 2048
variable_sip_via_rport: 2048
variable_max_forwards: 70
variable_presence_id: 1001%4010.0.1.100
variable_sip_h_P-Key-Flags: keys%3D%223%22
variable_switch_r_sdp: v%3D0%0D%0Ao%3Droot%201915884124%201915884124%20IN%20IP4%2010.0.1.241%0D%0As%3Dcall%0D%0Ac%3DIN%20IP4%2010.0.1.241%0D%0At%3D0%200%0D%0Am%3Daudio%2062258%20RTP/AVP%209%202%203%2018%204%20101%0D%0Aa%3Drtpmap%3A9%20g722/8000%0D%0Aa%3Drtpmap%3A2%20g726-32/8000%0D%0Aa%3Drtpmap%3A3%20gsm/8000%0D%0Aa%3Drtpmap%3A18%20g729/8000%0D%0Aa%3Drtpmap%3A4%20g723/8000%0D%0Aa%3Drtpmap%3A101%20telephone-event/8000%0D%0Aa%3Dfmtp%3A101%200-16%0D%0Aa%3Dptime%3A20%0D%0A
variable_remote_media_port: 62258
variable_read_codec: G722
variable_read_rate: 16000
variable_write_codec: G722
variable_write_rate: 16000
variable_open: true
variable_local_media_port: 62258
variable_endpoint_disposition: EARLY%20MEDIA
Content-Type: command/reply
Socket-Mode: async
Control: full

Now lets answer this call: (Note: sendmsg doesn't need uuid arg when in outbound mode)

call-command: execute
execute-app-name: answer\n\n

Which will reply:

Content-Type: command/reply
Reply-Text: +OK

This call is now answered but its sitting there waiting on your every command.

Now lets play this call a sound file:

call-command: execute
execute-app-name: playback
execute-app-arg: /tmp/swimp.raw\n\n

It is also possible to play tones:

call-command: execute
execute-app-name: playback
execute-app-arg: tone_stream://%(2000,4000,440,480)\n\n

The reply should be:

Content-Type: command/reply
Reply-Text: +OK

Now how do you stop the file playing? Here is how you would do that:

call-command: execute
execute-app-name: break\n\n

The reply should be:

Content-Type: command/reply
Reply-Text: +OK

Now lets hang this call up:

call-command: execute
execute-app-name: hangup\n\n

The reply should be:

Content-Type: command/reply
Reply-Text: +OK

Netcat will exit now.

You can also issue "myevents\n\n" or subscribe to other events in the normal way ie "events heartbeat\n\n"

In addition, you can execute API with:

api uuid_bridge <uuid> <uuid>


We have someone working on a C library to use as an event socket library also. but in the meantime there is also a crude C example with no client library but just inline commands in scripts/c/socket2me this small C program demonstrates how you can use the interface to not only control the call but to also request a 2 way media stream and process the data. This application uses spandsp to implement a fax send/recv similar to asterisk's rxfax txfax only over the loopback interface.

Java ESL Client

The Java ESL Client provides an Outbound socket mode, with a simple example of how to implement land run it.

Javascript / Node.js server


By default, the connected socket will not receive any FreeSWITCH events. You can issue the following event commands to let FreeSWITCH send events to the connected socket.

Receive only events from this channel


Receive all events from FreeSWITCH

event text all

Get lingering events on a hung up channel


To make sure you do not miss any events when the channel is hungup, you need to send the "linger command". I found this comment from Anthony Minessale on the FreeSWITCH-users mailing list.

it's a race,

sometimes the socket connection ends before the channel

the linger socket command was added to tell FS to wait for the last channel event before ending the connection

just send the command


Verto Live Array

From a post to the freeswitch-users mailing list dated 2016.03.28 12:02 EDT the following JavaScript code snippet reveals the basics of how to get raw conference events using $.verto.livearray() from the Verto library:


var la = new $.verto.liveArray(verto, data.pvtData.laChannel, data.pvtData.laName,
  {subParams: {callID: dialog ? dialog.callID : null}, "onChange": function (obj, args) {

  la.onChange = function (obj, args) {
  console.error("The change", args);

You can build on this template in your own JS code.


Q: Ordering and async keyword

When using async keyword, is there any guarantee regarding ordering?

Once you send a command you always get the reply before anything else.

In the case of a bgapi command, your reply will simply contain a job id, and when the command has actually finished it will send an event. See Event_Socket

In some situations (for example, when using the bridge app) you want to ensure the application completely executes before the next command is parsed. Use the following to achieve this:

event-lock: true

When you send the command.


SendMsg <uuid>
call-command: execute
execute-app-name: playback
execute-app-arg: /tmp/test.wav
event-lock: true

You may obtain the same result (setting event-lock to true) in scripting using the setEventLock method of connection object:


Q: Should I use sync mode or async mode?

In sync mode many things may not work as you expect, for example no DTMFs are received. When you're playing a file in sync mode it blocks till the file is done playing. In async mode you are returned control instantly. You can stop the playback, transfer the call and various other things in async mode.

Q: Why are API commands returning 'command not found'?

Remember to use "async full" for full API.

Q: Can I bridge a call with an Outbound socket?

Yes you can, simple execute a 'Bridge'

call-command: execute
execute-app-name: bridge
execute-app-arg: {ignore_early_media=true}sofia/gateway/myGW/177808
event-lock: true

The event-lock is key here, if you don't have it set, other events you might have sent can terminate the call, even if you've sent them before the bridge command. Remember that if you want to hangup the call when the bridge completes, either look for the event [CHANNEL_UNBRIDGE], or send a hangup event after the bridge.