__Variables

About

This page is a reference to all FreeSWITCH variables.

This page parallels the __Channel Variables) page. Consult both while the pages are being reconciled.

Variable References

You will see references to variables in the dialplan of the form ${variable} as well as $${variable}.

$${variable} is evaluated once and becomes a static reference to the variable and is therefore suitable for variables that do not change, such as the domain of the FreeSWITCH server. That is why you see $${domain} referenced so frequently in the Vanilla dialplan examples and as pre-processor variables which are evaluated once at startup.

${variable} is evaluated during each pass through the dialplan, so it is used for variables that are expected to change, such as the ${destination_number} or ${sip_to_user} fields.

absolute_codec_string

Sets the absolute codec string to use (nothing will be appended).

Usage:

<action application="set" data="absolute_codec_string=PCMU,GSM"/>
<action application="bridge" data="sofia/gateway/myprovider/5551231234"/>

Per-leg Dialstring:

<action application="bridge" data=[leg_timeout=60,origination_caller_id_number=918039251,
absolute_codec_string=^^:PCMA:PCMU]sofia/gateway/gateway1/${destination_number}|[leg_timeout=60,
absolute_codec_string=PCMA]sofia/gateway/gateway2/${destination_number}"/>

Global Dialstring:

<action application="bridge" data="{absolute_codec_string=PCMA}sofia/gateway/gateway2/${destination_number}"/>

Implemented By:

Module Name	Source File	Last Revised
mod_opal	mod_opal.cpp	10567
mod_sofia	sofia_glue.c	12700
core	switch_ivr_originate.c	12817

accountcode

Account code is mostly an arbitrary value that you can assign on a per leg basis. An important feature of accountcode is that if its value matches one of the CDR CSV templates defined in cdr_csv.conf.xml then that CDR template will be used when generating a CSV CDR.

Usage:

<action application="set" data="accountcode=custom"/>

acl_token

Description needed! Please contribute one.

Usage:

Example needed! Please contribute one.

Implemented By:

Module Name	Source File	Last Revised
mod_sofia	sofia.c	9056

alert_info

Add an Alert-Info Header like Snom and other Phones need for Ring or Ringtone informations.

Usage:

Load an external Wav Rington on all Calls.

<action application="export" data="alert_info=http://192.168.181.5/sounds/ctu.wav"/> 

Use the External Ringer (Change the Ringtone for the alert-external, alert-group and alert-internal in Snom under Setup > Preferences > Alert-Info Ringer)

<action application="export" data="alert_info=http://www.notused.com;info=alert-external;x-line-id=0"/>

(Set under See also: [http://wiki.snom.com/Web%5FInterface/V8/Preferences/Alert-Info%5FRinger](http://wiki.snom.com/Web%5FInterface/V8/Preferences/Alert-Info%5FRinger)

Implemented By:

Module Name	Source File	Last Revised
mod_sofia	sofia.c	6350
mod_sofia	sofia_glue.c	4819

answer_epoch

Description needed! Please contribute one.

Usage:

Example needed! Please contribute one.

Implemented By:

Module Name	Source File	Last Revised
core	switch_channel.c	6917

answer_stamp

Description needed! Please contribute one.

Usage:

Example needed! Please contribute one.

Implemented By:

Module Name	Source File	Last Revised
core	switch_channel.c	7750

answer_uepoch

Description needed! Please contribute one.

Usage:

Example needed! Please contribute one.

Implemented By:

Module Name	Source File	Last Revised
core	switch_channel.c	8686

answermsec

Description needed! Please contribute one.

Usage:

Example needed! Please contribute one.

Implemented By:

Module Name	Source File	Last Revised
core	switch_channel.c	3546

answersec

Description needed! Please contribute one.

Usage:

Example needed! Please contribute one.

Implemented By:

Module Name	Source File	Last Revised
core	switch_channel.c	3546

answerusec

Description needed! Please contribute one.

Usage:

Example needed! Please contribute one.

See also:

Module Name	Source File	Last Revised
core	switch_channel.c	3546

api_after_bridge

Execute an API command after bridge.

Usage:

Paging to PA System via Portaudio (w/ chime before and after announcement)

Implemented By:

Module Name	Source File	Last Revised
core	switch_ivr_bridge.c	7885

api_hangup_hook

Execute an API command on hangup.

Usage:

<action application="set" data="api_hangup_hook=jsrun cleanup.js ${uuid}"/>

See also:

session_in_hangup_hook
api_reporting_hook - like api_hangup_hook but after reporting state (both honor session_in_hangup_hook)

Implemented By:

Module Name	Source File	Last Revised
core	switch_core_state_machine.c	7885

api_on_answer

Execute an api (not an application) when the called party answers. To set an application, use execute_on_answer.

Usage:

<action application="export" data="nolocal:api_on_answer=uuid_broadcast ${uuid} beep.wav both"/>

Or,

<action application="bridge" data="{api_on_answer='uuid_broadcast ${uuid} beep.wav both'}sofia/gateway/provider/5551231234"/> 

See also:

Implemented By:

Module Name	Source File	Last Revised
core	switch_channel.c	15207

api_on_media

Execute a FreeSWITCH API when the far end sends media, i.e. ringing or 183/SDP.

The command is executed only on channels that are not already answered. Just use export or export with nolocal: prefix to make sure it is executed when b-leg answers.

In the second usage example below, we have originated an outbound call to a local extension, where we will wait 30 seconds without ignoring media. In this case we use 'set' and not 'export'.

Usage:

<action application="export" data="nolocal:execute_on_media=lua incrInUse.lua ${uuid}"/> 

Implemented By:

Module Name	Source File	Last Updated
Core	switch_channel.c	850f2e3

api_on_startup

Execute an api (not an application) when the switch starts up.

Usage:

<X-PRE-PROCESS cmd="set" data="api_on_startup=sofia recover"/>

Implemented By:

Module Name	Source File	Last Revised	date
core	switch_channel.c	e164b76caf0a47b6b33891eea80b1928067c9dd1	2011-12-15

asr_intercept_dtmf

Description needed! Please contribute one.

Usage:

Example needed! Please contribute one.

Implemented By:

Module Name	Source File	Last Revised
core	switch_ivr_async.c	8912

auto_answer_destination

Description needed! Please contribute one.

Usage:

Example needed! Please contribute one.

Implemented By:

Module Name	Source File	Last Revised
mod_sofia	sofia.c	10525

auto_hunt

Setting auto_hunt to "true" will alter the normal sequential processing of dialplan extensions. auto_hunt will cause the dialplan to 'jump' to a specific extension name, not processing any other extension. The destination_number and extension name must be the same in order for this to work. The condition must still match, but the extension name is the operative element.

In the example below, there is no way to reach extension 333 without auto_hunt.

Usage: In vars.xml:

<X-PRE-PROCESS cmd="set" data="auto_hunt=true"/>

Example:

<extension name="do_xfer">
  <condition field="destination_number" expression="^.*$">
    <action application="set" data="auto_hunt=true"/>
    <action application="transfer" data="333"/>
  </condition>
</extension>

<extension name="333">
  <condition field="destination_number" expression="^333$">
    <action application="info"/>
  </condition>
</extension>

Implemented By:

Module Name	Source File	Last Revised
mod_dialplan_xml	mod_dialplan_xml.c	12144

billmsec

Description needed! Please contribute one.

Usage:

Example needed! Please contribute one.

Implemented By:

Module Name	Source File	Last Revised
core	switch_channel.c	6917

billsec

Description needed! Please contribute one.

Usage:

Example needed! Please contribute one.

Implemented By:

Module Name	Source File	Last Revised
core	switch_channel.c	6917

billusec

Description needed! Please contribute one.

Usage:bypass_media

Example needed! Please contribute one.

See also:

Implemented By:

Module Name	Source File	Last Revised
core	switch_channel.c	6917

bind_meta_key

Selects the meta key to use with the mod_dptools: bind_meta_app dialplan application.

Usage:

<action application="set" data="bind_meta_key=#"/> 

Implemented By:

mod_dptools

bridge_answer_timeout

Timeout in seconds how long to tolerate a bridge that is in early media without being answered (can be set on either leg). Useful when you want to pass early media from b-leg to a-leg but also use ${call\_timeout}. This will consider the bridge failed if a 200 OK is not received before the bridge_answer_timeout.

Usage:

<action application="set" data="bridge_answer_timeout=20"/>

tags bridge answer timeout ringer ring timer voicemail

bridge_channel

Description needed! Please contribute one.

Usage:

Example needed! Please contribute one.

Implemented By:

Module Name	Source File	Last Revised
core	switch_ivr_bridge.c	6537

bridge_early_media

By default this is false. Set to true, this makes the bridge use the live audio from the b-leg as ringback to the a-leg. Setting bridge_early_media=true means the early media will be buffered.

Consider setting this to true if you are using a loopback channel to execute a bridge to an endpoint which sends back early media and the received early media's audio is degraded. The buffering resulting from setting bridge_early_media=true brings with it a higher resource cost (than bridge_early_media=false), but may improve the sound quality of the early media.

Usage: Set bridge_early_media before the bridge, or in the dial string for the bridge.

bridge_epoch

Description needed! Please contribute one.

Usage:

Example needed! Please contribute one.

Implemented By:

Module Name	Source File	Last Revised
core	switch_channel.c	3492

bridge_filter_dtmf

Setting this variable to true will prevent DTMF digits received on this channel when bridged from being sent to the other channel.

Usage:

You can set this variable in the dialplan before answering or inline as part of a dialstring.

Example dialplan usage:

<action application="set" data="bridge_filter_dtmf=true" />

Example dialstring usage:

{bridge_filter_dtmf=true}sofia/default/blah@baz.com

bridge_generate_comfort_noise

Generates comfort noise when bridged session is receiving discontinuous audio frames (silence suppression).

When greater than 0, this variable is set to the divisor of the silence generating function. 400 or 1400 are common values set, but you may experiment with other choices to pick one that sounds best.

When true, FreeSWITCH will pick a default comfort noise value.

When -1, FreeSWITCH will transmit silence without comfort noise. (As of 2012-10-25)

Usage:

Leg A wants a continuous stream of audio, but leg B is using silence suppression:

<action application="set" data="bridge_generate_comfort_noise=true"/>
<action application="bridge" data="sofia/user/1000"/>

Leg A uses silence suppression, but leg B wants a continuous stream of audio:

<action application="bridge" data="{bridge_generate_comfort_noise=true}sofia/user/1000"/>

Implemented By:

Module Name	Source File	Last Revised
core	switch_ivr_bridge.c	9543

bridge_hangup_cause

This is set to the hangup cause of the last bridged B leg of the call. If you have continue_on_fail=true and hangup_after_bridge=false you can do checks on this to see what "really" happened to the call. You can for instance do execute_extension after bridge, do a condition check on ${bridge_hangup_cause} to see if it contains MEDIA_TIMEOUT and then trigger a redial of the call or transfer to a cell phone. For a list of hangup causes, see Hangup Causes.

Usage:

<action application="log" data="1 B-leg hangup cause: ${bridge_hangup_cause}"/>

bridge_pre_execute_aleg_app

Command or api to be executed on the A leg before bridging the two channels.

Note: this is executed AFTER the call is setup but BEFORE the media (audio) is bridged.

Usage:

Example needed! Please contribute one

Implemented By:

Module Name	Source File	Last Revised
core	switch_ivr_bridge.c	8108

bridge_pre_execute_aleg_data

Arguments to be used with bridge_pre_execute_aleg_app.

Usage:

Example needed! Please contribute one.

Implemented By:

Module Name	Source File	Last Revised
core	switch_ivr_bridge.c	8108

bridge_pre_execute_bleg_app

Command or api to be executed on the B leg before bridging the two channels. Useful when originating a call from the event socket, CLI or XML-RPC.

It could for instance be used to do a HTTP GET with a script or mod_http to the IP address of a Snom phone to increase the ringer volume if you need to do a wakeup call.

Can also be used to bind a dtmf to an app on the b leg of a call so that it can survive a transfer.

Note: this is executed AFTER the call is setup but BEFORE the media (audio) is bridged.

Usage:

<action application="set" data="bridge_pre_execute_bleg_app=bind_meta_app"/>
<action application="set" data="bridge_pre_execute_bleg_data=1 a s att_xfer::sofia/profile/destination"/>

Implemented By:

Module Name	Source File	Last Revised
core	switch_ivr_bridge.c	8108

bridge_pre_execute_bleg_data

Arguments to be used with bridge_pre_execute_bleg_app

Usage:

See also:

Implemented By:

Module Name	Source File	Last Revised
core	switch_ivr_bridge.c	8108

bridge_stamp

Description needed! Please contribute one.

Usage:

Example needed! Please contribute one.

Implemented By:

Module Name	Source File	Last Revised
core	switch_channel.c	3443

bridge_terminate_key

Allows you to bind a key and the bridge will terminate if the dtmf matches

Usage: you can set bridge_terminate_key on either or both legs which will end the bridge, if it hangs up or not is decided by hangup_after_bridge=false or what is next in your dp

Implemented By:

Module Name	Source File	Last Revised
mod_dptools	mod_dptools.c	12479

bridge_to

Description needed! Please contribute one.

Usage:

Example needed! Please contribute one.

Implemented By:

Module Name	Source File	Last Revised
mod_commands	mod_commands.c	7421
core	switch_ivr.c	5738
core	switch_ivr_async.c	4798
core	switch_ivr_bridge.c	12671

bridge_uepoch

Description needed! Please contribute one.

Usage:

Example needed! Please contribute one.

Implemented By:

Module Name	Source File	Last Revised
core	switch_channel.c	3494

bridge_uuid

Description needed! Please contribute one.

Usage:

Example needed! Please contribute one.

Implemented By:

Module Name	Source File	Last Revised
core	switch_ivr_bridge.c	9603

bypass_media

When set, all the media address of the far end of the originating leg will be passed to the far end of the new call leg and vice versa so the signaling goes through FreeSWITCH but the media is point-to-point.

Usage:

 <action application="set" data="bypass_media=true"/>

Implemented By:

Module Name	Source File	Last Revised
mod_dptools	mod_dptools.c	12479
mod_esf	mod_esf.c	5115

bypass_media_after_bridge

This is useful for bypassing media after bridging has happened.

Usage:

 <action application="set" data="bypass_media_after_bridge=true"/>

It could be useful for an already established channel and doing a later bypass.

 uuid_setvar <uuid> bypass_media_after_bridge true
 uuid_broadcast <uuid> bridge::<endpoint>

Implemented By:

Module Name	Source File	Last Revised
mod_dptools	mod_dptools.c	12479
mod_sofia	sofia.c	9591
core	switch_ivr_bridge.c	9591

cache_speech_handles

Description needed! Please contribute one.

Usage:

Example needed! Please contribute one.

Implemented By:

Module Name	Source File	Last Revised
core	switch_ivr_play_say.c	5876

call_clientcode

Description needed! Please contribute one.

Usage:

Example needed! Please contribute one.

Implemented By:

Module Name	Source File	Last Revised
mod_sofia	sofia.c	6987

call_timeout

Controls how long (in seconds) to ring the B leg of a call when using the bridge application. The timeout is set on the A leg, and applies to any bridges that happen in the channel.

If you need to set a timeout on a call that has no A leg, use originate_timeout

If you need to set a timeout with enterprise bridging/originate, use originate_timeout

If you need to set the timeout on a per leg basis (i.e., a different timeout for each destination), use the leg_timeout variable.

Default Value: 60

Usage:

<action application="set" data="call_timeout=20"/>

Notes:

If a call timeout is to be specified against a group_call() list, use the following format:

 <action application="bridge" data="{originate_timeout=24}${group_call(sales@$${domain})}"/>

Beware that if you are not using {ignore_early_media=true} call_timeout is no longer applicable as soon as early media signal is received.

Implemented By:

Module Name	Source File	Last Revised
mod_dialplan_asterisk	mod_dialplan_asterisk.c	6208
mod_dptools	mod_dptools.c	10917
core	switch_ivr_originate.c	9120
core	switch_swig.c	4795

caller_id

Description needed! Please contribute one.

Usage:

Example needed! Please contribute one.

Implemented By:

Module Name	Source File	Last Revised
core	switch_channel.c	6917

caller_id_name

The caller id name set by the inbound call, not a real variable. Practically it is read only.

caller_id_number

The caller id phone number set by the inbound call, not a real variable. Practically it is read only. From sofia.c, the values used (in precedence) are the user parts from: P-Preferred-Identity, P-Asserted-Identity, Remote-Party-ID, and the From header.

campon

Controls whether camping is enabled or not.

Default: false

Usage:

<action application="set" data="campon=true"/>
<action application="bridge" data="sofia/gateway/myprovider/5551231234"/>

campon_announce_sound

File to play back after the first bridge fails (rg to announce what key to press to skip to fallback extension)

Default: none

Usage:

<action application="set" data="campon=true"/>
<action application="set" data="campon_stop_key=1"/>
<action application="set" data="campon_announce_sound=press_one_to_stop.wav"/>
<action application="bridge" data="sofia/gateway/myprovider/5551231234"/>

campon_fallback_context

Controls camping during bridge app (testing needed)

Usage:

<action application="set" data="campon"/>
<action application="bridge" data="sofia/gateway/myprovider/5551231234"/>

campon_fallback_dialplan

Controls camping during bridge app (testing needed)

Usage:

<action application="set" data="campon"/>
<action application="bridge" data="sofia/gateway/myprovider/5551231234"/>

campon_hold_music

If you don't set the hold_music variable, this variable controls hold music while camping.

Usage:

<action application="set" data="campon=true"/>
<action application="set" data="campon_hold_music=/data/campmusic/RelaxingCampSounds.wav"/>
<action application="bridge" data="sofia/gateway/myprovider/5551231234"/>

campon_fallback_exten

Controls camping during bridge app (testing needed)

Usage:

<action application="set" data="campon"/>
<action application="bridge" data="sofia/gateway/myprovider/5551231234"/>

campon_retries

Controls how many times the bridge will be retried while camping.

Default: 100

Usage:

<action application="set" data="campon=true"/>
<action application="set" data="campon_retries=13"/>
<action application="bridge" data="sofia/gateway/myprovider/5551231234"/>

campon_sleep

Controls how long to wait before starting a retry.

Default: 10

Usage:

<action application="set" data="campon=true"/>
<action application="set" data="campon_sleep=30"/>
<action application="bridge" data="sofia/gateway/myprovider/5551231234"/>

Implemented By:

Module Name	Source File	Last Revised
mod_dptools	mod_dptools.c	15201

campon_stop_key

DTMF digit that breaks the campon loop and skips directly to fallback extension

Default: none

Usage:

<action application="set" data="campon=true"/>
<action application="set" data="campon_stop_key=1"/>
<action application="set" data="campon_announce_sound=press_one_to_stop.wav"/>
<action application="set" data="campon_fallback_exten=1000"/>
<action application="bridge" data="sofia/gateway/myprovider/5551231234"/>

campon_timeout

This variable controls how long to attempt each bridge before timing out. It works exactly like call_timeout but only applies to camping.

Default: 10

Usage:

<action application="set" data="campon=true"/>
<action application="set" data="campon_timeout=20"/>
<action application="bridge" data="sofia/gateway/myprovider/5551231234"/>

cdr_csv_base

Description needed! Please contribute one.

Usage:

Example needed! Please contribute one.

See also:

Implemented By:

Module Name	Source File	Last Revised
mod_cdr_csv	mod_cdr_csv.c	6542

CHANNEL

Description needed! Please contribute one.

Usage:

Example needed! Please contribute one.

See also:

Implemented By:

Module Name	Source File	Last Revised
mod_dialplan_asterisk	mod_dialplan_asterisk.c	6205

channel_name

Description needed! Please contribute one.

Usage:

Example needed! Please contribute one.

See also:

Implemented By:

Module Name	Source File	Last Revised
core	switch_channel.c	6542

codec_string

Sets the base codec string to use.

Usage:

<action application="set" data="codec_string=PCMU,GSM"/>

Implemented By:

Module Name	Source File	Last Revised
mod_opal	mod_opal.cpp	10567
mod_sofia	sofia_glue.c	5114

conference_auto_outcall_announce

File name of audio message to play to conference member joining conference via the conference_set_auto_outcall application. Because the conference would be originating an outbound call to a member this typically would be a greeting with an explanation that the recipient will be joining a conference call.

Usage:

<action application="set" data="conference_auto_outcall_announce=sounds/soundfile.wav"/>

Implemented By:

Module Name	Source File	Last Revised
mod_conference	mod_conference.c	11104

conference_auto_outcall_caller_id_name

Caller ID name to use when dialing endpoints to join the conference via the conference_set_auto_outcall application.

Usage:

 <action application="set" data="conference_auto_outcall_caller_id_name=$${effective_caller_id_name}"/>

See also:

Implemented By:

Module Name	Source File	Last Revised
mod_conference	mod_conference.c	11104

conference_auto_outcall_caller_id_number

Caller ID number to use when dialing endpoints to join the conference via the conference_set_auto_outcall application.

Usage:

 <action application="set" data="conference_auto_outcall_caller_id_number=${effective_caller_id_number}"/>

See also:

Implemented By:

Module Name	Source File	Last Revised
mod_conference	mod_conference.c	11104

conference_auto_outcall_flags

Conference flags to set for members joining conference via the conference_set_auto_outcall application

Usage:

 <action application="set" data="conference_auto_outcall_flags=mute"/>

See also:

Implemented By:

Module Name	Source File	Last Revised
mod_conference	mod_conference.c	11104

conference_auto_outcall_maxwait

Maximum time in seconds that the channel that initiated the conference_set_auto_outcall will wait for members to join the conference.

Usage:

 <action application="set" data="conference_auto_outcall_maxwait=10"/>

See also:

Implemented By:

Module Name	Source File	Last Revised
mod_conference	mod_conference.c

conference_auto_outcall_prefix

The value of conference_auto_outcall_prefix is prepended to each of conference_set_auto_outcall values, of which there can be more than one.

Usage:

   <extension name="mad_boss_intercom">
    <condition field="destination_number" expression="^0911$">
      <action application="set" data="conference_auto_outcall_caller_id_name=Mad Boss1"/>
      <action application="set" data="conference_auto_outcall_caller_id_number=0911"/>
      <action application="set" data="conference_auto_outcall_timeout=60"/>
      <action application="set" data="conference_auto_outcall_flags=mute"/>
      <action application="set" data="conference_auto_outcall_prefix={sip_auto_answer=true,execute_on_answer='bind_meta_app 2 a s1 transfer::intercept:${uuid} inline'}"/>
      <action application="set" data="sip_exclude_contact=${network_addr}"/>
      <action application="conference_set_auto_outcall" data="${group_call(sales)}"/>
      <action application="conference" data="madboss_intercom1@default+flags{endconf|deaf}"/>
    </condition>
  </extension>

See also:

See Conferencing and Intercom for an example of using all this to page via multiple extensions.

See conference for details on initiating conferences.

Implemented By:

Module Name	Source File	Last Revised
mod_conference	mod_conference.c	11104

conference_auto_outcall_profile

Conference profile to use for members joining the conference via the conference_set_auto_outcall application.

Usage:

 <action application="set" data="conference_auto_outcall_profile=default"/>

See also:

Implemented By:

Module Name	Source File	Last Revised
mod_conference	mod_conference.c

conference_auto_outcall_timeout

Originate timeout to use when joining a member to a conference via conference_set_auto_outcall.

Usage:

<action application="set" data="conference_auto_outcall_timeout=60"/>

Implemented By:

Module Name	Source File	Last Revised
mod_conference	mod_conference.c	11104

conference_controls

Set this variable to specify which conference control set to use when transferring a caller into a conference. This allows you, for example, to have a control set for the conference moderator and another control set for regular conference members. The control set for the moderator could include the ability to mute or kick people, for example.

NOTE: You must create the desired conference control set. Also, if this is not set then the default conference control set is used for the conference member.

Usage:

<action application="set" data="conference_controls=moderator"/> 

Implemented By:

Module Name	Source File	Last Updated
mod_conference	mod_conference.c	ac19f73c

conference_enforce_security

Allows the conference security to be overridden. This applies in two different scenarios, one for inbound and one for outbound. By default, conference security is always applied to inbound calls and is always skipped for outbound calls. This channel variable allows the behavior to be modified.

Usage:

Inbound

<action application="set" data="conference_enforce_security=false"/>
<action application="conference" data="3000"/>

Outbound

originate {conference_enforce_security=true}sofia/internal/1001 &conference(3000)

Implemented By:

Module Name	Source File	Last Revised
mod_conference	mod_conference.c	12863

conference_enter_sound

When set, this channel variable will override the enter-sound param on conference profile for any conferences into which the call leg is transferred.

Usage:

<action application="set" data="conference_enter_sound=silence_stream://10"/>

Implemented By:

Module Name	Source File	Last Updated
Mod_conference	mod_conference.c

conference_exit_sound

Usage:

<action application="set" data="conference_exit_sound=silence_stream://10"/> 

Implemented By:

Module Name	Source File	Last Updated
Mod_conference	mod_conference.c

conference_last_matching_digits

Contains the last matching digits that the user on this channel sent into the conference.

Usage:

<action application="log" data="INFO Last digits sent by this user: ${conference_last_matching_digits}"/>

See also:

Implemented By:

Module Name	Source File	Last Updated
mod_conference	mod_conference.c	f6bcf830

conference_member_id

Contains the conference_member_id value for any conference to which the channel may be connected.

Implemented By:

mod_conference.c

conference_moderator

Is true if the channel is connected to a conference as a moderator.

Implemented By:

mod_conference.c

conference_name

The name of the last conference joined by this channel.

Usage:

<action application="log" data="INFO Last conference joined by this user: ${conference_name}"/>

Implemented By:

Module Name	Source File	Last Updated
mod_conference	mod_conference.c

conference_recording

Contains the file name of the conference recording for the conference to which the channel is connected.

Implemented By:

mod_conference.c

conference_uuid

Every instance of a conference has its own UUID. This channel variable stores the conference UUID for the most recent conference in which the channel was a member. It is set as soon as the channel enters the conference, and will show up in XML CDRs and uuid_dump calls, as well as any events that show channel variables.

Implemented By:

Module Name	Source File	Last Updated
mod_conference	mod_conference.c	2011-03-18

continue_on_fail

Controls what happens when the called party can not be reached (busy/offline). If "true" the dialplan continues to be processed. If "false" the dialplan will stop processing. Can contain the return messages that will continue on fail also.

Usage:

<action application="set" data="continue_on_fail=true"/>

or,

<action application="set" data="continue_on_fail=NORMAL_TEMPORARY_FAILURE,USER_BUSY,NO_ANSWER,NO_ROUTE_DESTINATION"/>

or Q.850 cause codes,

<action application="set" data="continue_on_fail=3,17,18,27"/>

Implemented By:

Module Name	Source File	Last Revised
mod_dptools	mod_dptools.c	5875
mod_lcr	mod_lcr.c	10510

copy_xml_cdr

Copy's the other leg's XML CDR into this leg's CDR. For example, the A leg's CDR will contain a variable named b_leg_cdr whose contents are the URL-encoded XML CDR data from the B leg. This variable must be set on the B leg, so use {copy_xml_cdr=true} in the dialstring or use exportinstead of set.

Usage:

<action application="bridge" data="{copy_xml_cdr=true} user/${dialed_extension}@${domain_name}"/>

See also:

Mod_xml_cdr

Implemented By:

Module Name	Source File	Last Revised
core	switch_ivr_bridge.c	8955

created_time

Contains the start time (in microseconds) of when the call was created.

Usage:

In the event that a call is transferred, this will contain the time of when the entire call was created, not that specific transfer. If you need the transfer created time, see Variable_profile_created_time and Variable_progress_time

current_application

Description needed! Please contribute one.

Usage:

Example needed! Please contribute one.

Implemented By:

Module Name	Source File	Last Revised
mod_fifo	mod_fifo.c	10277
mod_sofia	sofia.c	11562
core	switch_core_session.c	10653

current_application_data

Description needed! Please contribute one.

Usage:

Example needed! Please contribute one.

See also:

Implemented By:

Module Name	Source File	Last Revised
mod_fifo	mod_fifo.c	10277
mod_sofia	sofia.c	11562

current_application_response

Description needed! Please contribute one.

Usage:

Example needed! Please contribute one.

Implemented By:

Module Name	Source File	Last Revised
mod_dptools	mod_dptools.c	10653
core	switch_core_session.c	10653

default_language

Controls the default language the Say Phrase engine will use when no language is explicitly specified in the API call. This permits you to easily support multiple languages by only changing a single variable at call time.

Usage:

<action application="set" data="default_language=fr"/>

Implemented By:

Module Name	Source File	Last Revised
core	switch_ivr_play_say.c	4796

destination_number

Description needed! Please contribute one.

Usage:

Example needed! Please contribute one.

Implemented By:

Module Name	Source File	Last Revised
mod_sofia	sofia.c	6054
core	switch_ivr_bridge.c	7083

detect_speech_result

The result of play_and_detect_speech.

Usage:

This value is read-only.

Implemented By:

Module Name	Source File	Last Revised
mod_dptools	mod_dptools.c

dialed_domain

Description needed! Please contribute one.

Usage:

Example needed! Please contribute one.

Implemented By:

Module Name	Source File	Last Revised
mod_commands	mod_commands.c	10917
mod_dptools	mod_dptools.c	7225

dialed_group

Description needed! Please contribute one.

Usage:

Example needed! Please contribute one.

Implemented By:

Module Name	Source File	Last Revised
mod_commands	mod_commands.c	10917

dialed_user

Description needed! Please contribute one.

Usage:

Example needed! Please contribute one.

Implemented By:

Module Name	Source File	Last Revised
mod_commands	mod_commands.c	10917
mod_dptools	mod_dptools.c	7225

digits_dialed

Description needed! Please contribute one.

Usage:

Example needed! Please contribute one.

Implemented By:

Module Name	Source File	Last Revised
core	switch_channel.c	12244

direction

It is used to determine if the particular leg of the call is inbound or outbound.

Usage:

<condition field="${direction}" expression="^inbound$">

See also:

__Channel Variables

disable_app_log

Description needed! Please contribute one.

Usage:

Example needed! Please contribute one.

Implemented By:

Module Name	Source File	Last Revised
core	switch_core_session.c	8517

rtp_disable_hold

When set to true the user may not put the call on hold.

Usage:

Implemented By:

mod_sofia

rtp_pass_codecs_on_stream_change

Pass codecs thru from a to b on stream change, example adding or removing video from a bridged call.

Usage

disable_q850_reason

When set to true, this disables sending of the Reason header, which includes the Q.850 reason code, in responses and BYEs. For a list of hangup causes and their Q.850 codes, see Hangup Causes. This is available as of revision 15850 committed 12/8/2009.

Usage:

disable_radius_start

Description needed! Please contribute one.

Usage:

Example needed! Please contribute one.

Implemented By:

Module Name	Source File	Last Revised
mod_radius_cdr	mod_radius_cdr.c	10793

disable_radius_stop

Description needed! Please contribute one.

Usage:

Example needed! Please contribute one.

See also:

Implemented By:

Module Name	Source File	Last Revised
mod_radius_cdr	mod_radius_cdr.c	10793

disable_rtp_auto_adjust

Disable rtp auto adjust if it not behaves as you expected.

It stops the switch from rewriting the RTP destination based on the source

When RTP Auto-Adjust is ON FreeSWITCH will change the destination RTP address (port?) to match the source of the incoming packets, this doesn't work if the other end is really wanting to send and receive on a different addr (port?).

Usage:

Implemented By:

Module Name	Source File	Last Revised
mod_dingaling	mod_dingaling.c	12641
mod_sofia	sofia_glue.c	8908

DISPLACE_HANGUP_ON_ERROR

When set to true this channel variable will cause the call to hangup if there is an error when trying to uuid_displace the call. The default is "false". The default behavior before 2013/03/21 was to hangup.

Usage:

See also:

Channel Variables

Implemented By:

Module Name	Source File	Last Updated
Core	switch_ivr_async.c

dl_cid_msg

Description needed! Please contribute one.

Usage:

Example needed! Please contribute one.

Implemented By:

Module Name	Source File	Last Revised
mod_dingaling	mod_dingaling.c	4270

dl_from_host

Description needed! Please contribute one.

Usage:

Example needed! Please contribute one.

Implemented By:

Module Name	Source File	Last Revised
mod_dingaling	mod_dingaling.c	4943

dl_from_user

Description needed! Please contribute one.

Usage:

Example needed! Please contribute one.

Implemented By:

Module Name	Source File	Last Revised
mod_dingaling	mod_dingaling.c	4943

dl_host

Description needed! Please contribute one.

Usage:

Example needed! Please contribute one.

Implemented By:

Module Name	Source File	Last Revised
mod_dingaling	mod_dingaling.c	4943

dl_user

Description needed! Please contribute one.

Usage:

Example needed! Please contribute one.

Implemented By:

Module Name	Source File	Last Revised
mod_dingaling	mod_dingaling.c	4943

domain_name

Description needed! Please contribute one.

Usage:

Example needed! Please contribute one.

See also:

Implemented By:

Module Name	Source File	Last Revised
core	switch_ivr.c	9056

drop_dtmf

Set this on an inbound channel before answer or on an outbound channel before the bridge/originate in order to prevent DTMF events from being sent to the channel.

Only tested with RFC2833, may also work for INFO / inband. See Jira issue FS-4769. Commit 60f7849cbe72.

Usage:

<action application="set" data="drop_dtmf=true"/>
<action application="answer"/>

or,

<action application="export" data="nolocal:drop_dtmf=true"/>
<action application="bridge" data="sofia/internal/100@1.2.3.4"/>

or,

<action application="bridge" data="{drop_dtmf=true}sofia/internal/100@1.2.3.4"/>

Implemented By:

Module Name	Source File	Last Revised
mod_sofia	sofia_glue.c

dtmf_type

For inband DTMF, Misc. Dialplan Tools start_dtmf must be used in the dialplan.

Usage:

or,

or,

Implemented By:

Module Name	Source File	Last Revised
mod_sofia	sofia_glue.c	6952

dtmf_verbose

Enables verbose logging of Spandsp DTMF detector. Default is false. Set this variable prior to executing spandsp_start_dtmf.

Usage:

See also:

Implemented By:

Module Name	Source File	Last Updated
mod_spandsp	mod_spandsp_dsp.c

duration

Description needed! Please contribute one.

Usage:

Example needed! Please contribute one.

Implemented By:

Module Name	Source File	Last Revised
core	switch_channel.c	6917

easy_acctcode

Description needed! Please contribute one.

Usage:

Example needed! Please contribute one.

See also:

Implemented By:

Module Name	Source File	Last Revised
mod_easyroute	mod_easyroute.c	10929

easy_destnum

Description needed! Please contribute one.

Usage:

Example needed! Please contribute one.

See also:

Implemented By:

Module Name	Source File	Last Revised
mod_easyroute	mod_easyroute.c	10929

easy_dialstring

Description needed! Please contribute one.

Usage:

Example needed! Please contribute one.

Implemented By:

Module Name	Source File	Last Revised
mod_easyroute	mod_easyroute.c	10929

easy_group

Description needed! Please contribute one.

Usage:

Example needed! Please contribute one.

Implemented By:

Module Name	Source File	Last Revised
mod_easyroute	mod_easyroute.c	10929

easy_limit

Description needed! Please contribute one.

Usage:

Example needed! Please contribute one.

Implemented By:

Module Name	Source File	Last Revised
mod_easyroute	mod_easyroute.c	10929

eavesdrop_annnounce_macro

Description needed! Please contribute one.

Usage:

Example needed! Please contribute one.

Implemented By:

Module Name	Source File	Last Revised
core	switch_ivr_async.c	8098

eavesdrop_announce_id

Description needed! Please contribute one.

Usage:

Example needed! Please contribute one.

Implemented By:

Module Name	Source File	Last Revised
core	switch_ivr_async.c	8098

eavesdrop_group

Description needed! Please contribute one.

Usage:

Example needed! Please contribute one.

Implemented By:

Module Name	Source File	Last Revised
core	switch_ivr_async.c	8686

eavesdrop_indicate_failed

Description needed! Please contribute one.

Usage:

Example needed! Please contribute one.

Implemented By:

Module Name	Source File	Last Revised
mod_dptools	mod_dptools.c	8098

eavesdrop_indicate_idle

Description needed! Please contribute one.

Usage:

Example needed! Please contribute one.

Implemented By:

Module Name	Source File	Last Revised
mod_dptools	mod_dptools.c	8091

eavesdrop_indicate_new

Description needed! Please contribute one.

Usage:

Example needed! Please contribute one.

Implemented By:

Module Name	Source File	Last Revised
mod_dptools	mod_dptools.c	8091

eavesdrop_require_group

Description needed! Please contribute one.

Usage:

Example needed! Please contribute one.

See also:

Implemented By:

Module Name	Source File	Last Revised
mod_dptools	mod_dptools.c	8098

effective_caller_id_name

Sets the effective callerid name. This is automatically exported to the B-leg; however, it is not valid in an origination string. In other words, set this before calling bridge, otherwise use origination_caller_id_name

For Snom 370/820 users:

If you want to display LEG A's name (if available) as soon as LEG B (here the local Snom) rings, you have to set origination_caller_id_name or effective_caller_id_name as described. Otherwise, in LEG B's display, you will see LEG A's number during ringing and switching to LEG A's name after picking up the call by LEG B. To remove it set it to "_undef_".

Usage:

<action application="set" data="effective_caller_id_name=Bob Smith"/>

Implemented By:

Module Name	Source File	Last Revised
mod_voicemail	mod_voicemail.c	8854
core	switch_core_session.c	9248

effective_caller_id_number

Sets the effective callerid number. This is automatically exported to the B-leg; however, it is not valid in an origination string. In other words, set this before calling bridge, otherwise use origination_caller_id_number

Usage:

<action application="set" data="effective_caller_id_number=9185551212"/>

Implemented By:

Module Name	Source File	Last Revised
mod_voicemail	mod_voicemail.c	8854
core	switch_core_session.c	9248

enable_file_write_buffering

Enable file buffering when recording a file, defaults to true if not set.

Buffer size defaults to `SWITCH_DEFAULT_FILE_BUFFER_LEN` but can be overridden by putting bytes size instead of true (see below example).

Usage:

<action application="set" data="enable_file_write_buffering=false"/>
<action application="set" data="enable_file_write_buffering=true"/>
<action application="set" data="enable_file_write_buffering=65535"/> 

Implemented By:

Module Name	Source File	Last Revised
core	switch_ivr_play_say.c	11677

enable_heartbeat_events

Description needed! Please contribute one.

Usage:

Example needed! Please contribute one.

Implemented By:

Module Name	Source File	Last Revised
core	switch_channel.c	9882

end_epoch

Description needed! Please contribute one.

Usage:

Example needed! Please contribute one.

See also:

Implemented By:

Module Name	Source File	Last Revised
core	switch_channel.c	6917

end_stamp

Description needed! Please contribute one.

Usage:

Example needed! Please contribute one.

See also:

Implemented By:

Module Name	Source File	Last Revised
core	switch_channel.c	6917

end_uepoch

Description needed! Please contribute one.

Usage:

Example needed! Please contribute one.

See also:

Implemented By:

Module Name	Source File	Last Revised
core	switch_channel.c	6917

ep_codec_prefer_sdp

ep_codec_prefer_sdp contains the "endpoint" codec string on the A leg. The order of preference is chosen by the A leg

Usage:

Need some example.

Implemented By:

Module Name	Source File	Last Updated
Sofia	mod_sofia_glue.c

ep_codec_string

ep_codec_string contains the "endpoint" codec string on the A leg. This codec list includes only the codecs that both the A leg and FreeSWITCH agree upon. This variable is set only when inbound-late-negotiation is enable on the SIP profile. NOTE: When there is more than one codec in the list, the order of preference is chosen by FreeSWITCH, not by the A leg

Usage:

<action application="bridge" data="{absolute_codec_string=${ep_codec_string}}sofia/foo/bar@baz"/>

Implemented By:

Module Name	Source File	Last Updated
Sofia	mod_sofia.c

exec_after_bridge_app

Execute an application command after the bridge has been terminated. To be used with exec_after_bridge_arg. By contrast, to execute when the bridge has been established use execute_on_answer

Usage:

<action application="set" data="exec_after_bridge_app=transfer"/>
<action application="set" data="exec_after_bridge_arg=2102"/> 

See also:

Implemented By:

Module Name	Source File	Last Revised
core	switch_ivr_bridge.c	5546

exec_after_bridge_arg

Argument passed to exec_after_bridge_app.

Usage:

<action application="set" data="exec_after_bridge_app=transfer"/>
<action application="set" data="exec_after_bridge_arg=2102"/> 

Implemented By:

Module Name	Source File	Last Revised
core	switch_ivr_bridge.c	5546

execute_on_answer

Execute an application (not an api) when the called party answers. To set an api, use api_on_answer. execute_on_answer will also allow for more control when dealing with no answer conditions in cases where you cannot ignore early media.

The command is executed only on channels that are not already answered. Just use export or export with nolocal: prefix to make sure it is executed when b-leg answers.

In the second usage example below, we have originated an outbound call to a local extension, where we will wait 30 seconds while manually ignoring media. In this case we use 'set' and not 'export'.

Usage:

<action application="export" data="nolocal:execute_on_answer=lua incrInUse.lua ${uuid}"/> 

or, to wait 30 seconds for an answer while 'manually' ignoring early media

originate {ignore_early_media=true}sofia/gateway/my_gateway/5551212 885551212

<extension name="exe_on_ans">
  <condition field="destination_number" expression="^88(\d+)$">
    <action application="set" data="execute_on_answer=transfer ANSWEREDCALL XML default"/>
    <action application="log" data="INFO Waiting 30 seconds for $1 to answer..."/>
    <action application="sleep" data="30000"/>
    <action application="log" data="INFO Call to $1 was not answered, taking alternative action..."/>
    <action application="transfer" data="UNANSWEREDCALL XML default"/>
  </condition>
</extension>

If you need to set multiple execute_on_answers, see: The execute on family

Implemented By:

Module Name	Source File	Last Revised
core	switch_channel.c	9636

execute_on_fax_detect

This is an example of how to create a channel variable page. This section is the description of the variable. Put the description information here and then the usage example below.

Usage:

<action application="set" data="skeleton=foo"/>

execute_on_fax_failure

This is an example of how to create a channel variable page. This section is the description of the variable. Put the description information here and then the usage example below.

Usage:

<action application="set" data="skeleton=foo"/>

execute_on_fax_result

This is an example of how to create a channel variable page. This section is the description of the variable. Put the description information here and then the usage example below.

Usage:

<action application="set" data="skeleton=foo"/>

execute_on_fax_success

This is an example of how to create a channel variable page. This section is the description of the variable. Put the description information here and then the usage example below.

Usage:

<action application="set" data="skeleton=foo"/>

execute_on_media

Execute an application when the far end sends media, i.e. ringing or 183/SDP.

The command is executed only on channels that are not already answered. Just use export or export with nolocal: prefix to make sure it is executed when b-leg answers.

In the second usage example below, we have originated an outbound call to a local extension, where we will wait 30 seconds without ignoring media. In this case we use 'set' and not 'export'.

Usage:

<action application="export" data="nolocal:execute_on_media=lua incrInUse.lua ${uuid}"/>

execute_on_media_timeout

Execute an application if the far end stops sending media and times out.

Usage:

<action application="export" data="nolocal:execute_on_media_timeout=lua oops_timeout.lua ${uuid}"/>
<action application="set" data="execute_on_media_timeout=transfer HANDLE_MEDIA_TIMEOUT XML default"/>

See also:

mod_dptools

Implemented By:

Module Name	Source File	Last Revised
Sofia	mod_sofia.c	7eafa85d

execute_on_originate

Executes code on successful origination. Use the '<app> <arg>' format to execute in the origination thread or use '<app>::<arg>' to execute asynchronously.

Successful origination means the remote server responds, NOT when the call is actually answered.

Usage:

originate {ignore_early_media=true,execute_on_originate='cng_plc'}sofia/gateway/foo/123456789 9664

originate {ignore_early_media=true,execute_on_originate='my_app::my_arg'}sofia/gateway/foo/123456789 9664

execute_on_pre_answer

Execute an application (not an api) when the called party "preanswers" - that is, when some form of early media is coming or the far end sends a 180 Ringing.

The command is executed only on channels that are not already answered. Just use export or export with nolocal: prefix to make sure it is executed when b-leg answers.

In the second usage example below, we have originated an outbound call to a local extension, where we will wait 30 seconds without ignoring media. In this case we use 'set' and not 'export'.

Usage:

<action application="export" data="nolocal:execute_on_pre_answer=lua incrInUse.lua ${uuid}"/>

execute_on_ring

Execute a command when the called party rings.

Usage:

<action application="set" data="nolocal:execute_on_ring=lua markring ${uuid}"/>

execute_on_sip_extra_headers

Usage:

Avi Marcus - could this be used to send an incoming LRN header (e.g. alcazar) to a non-blocking curl or lua script (for caching)? Or would api_on_sip_extra_headers be good for that?

execute_on_sip_reinvite

Execute a command when SIP Reinvite.

Usage:

Add description here.

export_vars

export_vars lists variables to be exported to the other leg upon bridge. Unlike export, it only lists the variables to export without actually setting them. Note: This is useful to transfer information from a-leg (INVITE) to the future b-leg of REFER.

See also:

Misc._Dialplan_Tools_export

Implemented By:

Module Name	Source File	Last Revised
mod_dptools	mod_dptools.c	5118
core	switch_core_session.c	4796

EXTEN

Description needed! Please contribute one.

Usage:

Example needed! Please contribute one.

See also:

Implemented By:

Module Name	Source File	Last Revised
mod_dialplan_asterisk	mod_dialplan_asterisk.c	6205

fail_on_single_reject

This is useful when using the "," AND operator in the DATA field of a bridge. The AND operator notifies a list of destinations, bridging to the first destination that accepts the call. Typically if a destination in the list rejects the call, the bridge will continue to be attempted until either another destination accepts the call, or a timeout occurs.

This variable allows one to terminate the bridging attempt on a single rejection of the call. This means the bridge attempt would fail, and if continue_on_fail has not been set, the call is terminated. This variable would be set within a condition before a bridge application. When used in conjunction with the continue_on_fail variable, one can perform operations such as rolling over a rejected caller to an answering machine application.

The default setting is FALSE, meaning a single rejection will not terminate the bridging attempt.

It can also be set to a list of failure causes to stop on, and can be negated to a list of causes not to stop on (i.e. stop on all other failure causes).

Usage:

or,

or,

or to use a list,

or for negated list,

failed_xml_cdr_prefix

If you set that on the A leg and any and all failed B originates generate a full XML CDR report and set it as a variable, this includes during a forked dial.

So say you try to call sofia/profile/a@xxxxxxx,sofia/profile/b@xxxxxxx

And it fails completely, before you make the call you set failed_xml_cdr_prefix to "bad_call"

Then you end up with ${bad_call_1} and ${bad_call_2} which are each a full XML report including all the vars etc.

Usage:

Implemented By:

Module Name	Source File	Last Revised
core	switch_ivr_originate.c	8730

failure_causes

Controls which failure causes will be considered as a failure to the bridge(s). This will change the values for which continue_on_fail will fail by default unless continue_on_fail is set to true.

Usage:

or Q.850 cause codes,

Implemented By:

Module Name	Source File	Last Revised
mod_dptools	mod_dptools.c	12058

fax_bad_rows

Description needed! Please contribute one.

Usage:

Example needed! Please contribute one.

See also:

Implemented By:

Module Name	Source File	Last Revised
mod_spandsp	mod_spandsp_fax.c	3026e639fe0b8cfec9a37f2ce99aee7779b9e736

fax_disable_v17

fax_disable_v17 prevents the use of the V.17 (9,600bps to 14,400bps) FAX modem. This means FAXes will be limited to the use of V.29 (9,600bps and 7,200bps) and V.27ter (4,800bps). Some VoIP systems handle V.17 so poorly there are often good reasons to want to disable the use of this modem.

Usage:

Example needed! Please contribute one.

See also:

Implemented By:

Module Name	Source File	Last Revised
mod_spandsp	mod_spandsp_fax.c	3026e639fe0b8cfec9a37f2ce99aee7779b9e736

fax_document_total_pages

Description needed! Please contribute one.

Usage:

Example needed! Please contribute one.

See also:

Implemented By:

Module Name	Source File	Last Revised
mod_spandsp	mod_spandsp_fax.c	3026e639fe0b8cfec9a37f2ce99aee7779b9e736

fax_ecm_requested

Description needed! Please contribute one.

Usage:

Example needed! Please contribute one.

See also:

Implemented By:

Module Name	Source File	Last Revised
mod_spandsp	mod_spandsp_fax.c	3026e639fe0b8cfec9a37f2ce99aee7779b9e736

fax_ecm_used

Description needed! Please contribute one.

Usage:

Example needed! Please contribute one.

See also:

Implemented By:

Module Name	Source File	Last Revised
mod_spandsp	mod_spandsp_fax.c	3026e639fe0b8cfec9a37f2ce99aee7779b9e736

fax_end_page

Description needed! Please contribute one.

Usage:

Example needed! Please contribute one.

See also:

Implemented By:

Module Name	Source File	Last Revised
mod_spandsp	mod_spandsp_fax.c	3026e639fe0b8cfec9a37f2ce99aee7779b9e736

fax_filename

Description needed! Please contribute one.

Usage:

Example needed! Please contribute one.

See also:

Implemented By:

Module Name	Source File	Last Revised
mod_spandsp	mod_spandsp_fax.c	3026e639fe0b8cfec9a37f2ce99aee7779b9e736

fax_force_caller

Description needed! Please contribute one.

Usage:

Example needed! Please contribute one.

See also:

Implemented By:

Module Name	Source File	Last Revised
mod_spandsp	mod_spandsp_fax.c	3026e639fe0b8cfec9a37f2ce99aee7779b9e736

fax_ident

Description needed! Please contribute one.

Usage:

Example needed! Please contribute one.

See also:

Implemented By:

Module Name	Source File	Last Revised
mod_spandsp	mod_spandsp_fax.c	3026e639fe0b8cfec9a37f2ce99aee7779b9e736

fax_image_resolution

Description needed! Please contribute one.

Usage:

Example needed! Please contribute one.

See also:

Implemented By:

Module Name	Source File	Last Revised
mod_spandsp	mod_spandsp_fax.c	3026e639fe0b8cfec9a37f2ce99aee7779b9e736

fax_image_size

Description needed! Please contribute one.

Usage:

Example needed! Please contribute one.

See also:

Implemented By:

Module Name	Source File	Last Revised
mod_spandsp	mod_spandsp_fax.c	3026e639fe0b8cfec9a37f2ce99aee7779b9e736

fax_local_station_id

Description needed! Please contribute one.

Usage:

Example needed! Please contribute one.

See also:

Implemented By:

Module Name	Source File	Last Revised
mod_spandsp	mod_spandsp_fax.c	3026e639fe0b8cfec9a37f2ce99aee7779b9e736

fax_prefix

Description needed! Please contribute one.

Usage:

Example needed! Please contribute one.

See also:

Implemented By:

Module Name	Source File	Last Revised
mod_spandsp	mod_spandsp_fax.c	3026e639fe0b8cfec9a37f2ce99aee7779b9e736

fax_remote_station_id

Description needed! Please contribute one.

Usage:

Example needed! Please contribute one.

See also:

Implemented By:

Module Name	Source File	Last Revised
mod_spandsp	mod_spandsp_fax.c	3026e639fe0b8cfec9a37f2ce99aee7779b9e736

fax_result_code

Result Code	Message
0	OK
2	Timed out waiting for initial communication
3	Timed out waiting for the first message
5	The HDLC carrier did not stop in a timely manner
6	Failed to train with any of the compatible modems
13	Unexpected message received
14	Received bad response to DCS or training
15	Received a DCN from remote after sending a page
17	Received a DCN while waiting for a DIS
20	Received no response to DCS or TCF
23	Invalid ECM response received from transmitter
31	Timer T2 expired while waiting for fax page
32	Timer T2 expired while waiting for next fax page
48	Disconnected after permitted retries
49	The call dropped prematurely

Usage:

Example needed! Please contribute one.

See also:

Implemented By:

Module Name	Source File	Last Revised
mod_spandsp	mod_spandsp_fax.c	3026e639fe0b8cfec9a37f2ce99aee7779b9e736

fax_result_text

Description needed! Please contribute one.

Usage:

Example needed! Please contribute one.

See also:

Implemented By:

Module Name	Source File	Last Revised
mod_spandsp	mod_spandsp_fax.c	3026e639fe0b8cfec9a37f2ce99aee7779b9e736

fax_success

Description needed! Please contribute one.

Usage:

Example needed! Please contribute one.

See also:

Implemented By:

Module Name	Source File	Last Revised
mod_spandsp	mod_spandsp_fax.c	3026e639fe0b8cfec9a37f2ce99aee7779b9e736

fax_transfer_rate

Description needed! Please contribute one.

Usage:

Example needed! Please contribute one.

See also:

Implemented By:

Module Name	Source File	Last Revised
mod_spandsp	mod_spandsp_fax.c	3026e639fe0b8cfec9a37f2ce99aee7779b9e736

fax_use_ecm

Description needed! Please contribute one.

Usage:

Example needed! Please contribute one.

See also:

Implemented By:

Module Name	Source File	Last Revised
mod_fax	mod_fax.c	9468

fax_v17_disabled

Description needed! Please contribute one.

Usage:

Example needed! Please contribute one.

See also:

Implemented By:

Module Name	Source File	Last Revised
mod_spandsp	mod_spandsp_fax.c	3026e639fe0b8cfec9a37f2ce99aee7779b9e736

fax_verbose

Description needed! Please contribute one.

Usage:

Example needed! Please contribute one.

See also:

Implemented By:

Module Name	Source File	Last Revised
mod_spandsp	mod_spandsp_fax.c	3026e639fe0b8cfec9a37f2ce99aee7779b9e736

fifo_announce

Description needed! Please contribute one.

Usage:

Example needed! Please contribute one.

See also:

Implemented By:

Module Name	Source File	Last Revised
mod_fifo	mod_fifo.c	8689

fifo_bridged

Description goes here

Usage:

Example needed

fifo_caller_consumer_import

Import list of variables from the caller to the consumer.

Usage:

See also:

Implemented By:

Module Name	Source File	Last Revised
mod_fifo	mod_fifo.c	12722

fifo_caller_exit_key

Description needed! Please contribute one.

Usage:

Example needed! Please contribute one.

See also:

Implemented By:

Module Name	Source File	Last Revised
mod_fifo	mod_fifo.c	8026

fifo_chime_freq

Description needed! Please contribute one.

Usage:

Example needed! Please contribute one.

See also:

Implemented By:

Module Name	Source File	Last Revised
mod_fifo	mod_fifo.c	8026

fifo_chime_list

Description needed! Please contribute one.

Usage:

Example needed! Please contribute one.

See also:

Implemented By:

Module Name	Source File	Last Revised
mod_fifo	mod_fifo.c	8026

fifo_consumer_caller_import

Import list of variables from the consumer to the caller

Usage:

Implemented By:

Module Name	Source File	Last Revised
mod_fifo	mod_fifo.c	12722

fifo_consumer_exit_key

Description needed! Please contribute one.

Usage:

Example needed! Please contribute one.

See also:

Implemented By:

Module Name	Source File	Last Revised
mod_fifo	mod_fifo.c	9043

fifo_consumer_id

Description needed! Please contribute one.

Usage:

Example needed! Please contribute one.

See also:

Implemented By:

Module Name	Source File	Last Revised
mod_fifo	mod_fifo.c	8026

fifo_consumer_wrapup_key

Description needed! Please contribute one.

Usage:

Example needed! Please contribute one.

See also:

Implemented By:

Module Name	Source File	Last Revised
mod_fifo	mod_fifo.c	7961

fifo_consumer_wrapup_sound

Description needed! Please contribute one.

Usage:

Example needed! Please contribute one.

See also:

Implemented By:

Module Name	Source File	Last Revised
mod_fifo	mod_fifo.c	7961

fifo_consumer_wrapup_time

Description needed! Please contribute one.

Usage:

Example needed! Please contribute one.

See also:

Implemented By:

Module Name	Source File	Last Revised
mod_fifo	mod_fifo.c	9860

fifo_hold_music

Description needed! Please contribute one.

Usage:

Example needed! Please contribute one.

See also:

Implemented By:

Module Name	Source File	Last Revised
mod_fifo	mod_fifo.c	8065

fifo_manual_bridged

Description goes here

fifo_member_wait

If it's set to 'wait', then the consumer's leg of the call will not hangup when the caller hangs up [default]. If it's set to 'nowait' then the consumer's leg of the call will hangup when the caller hangs up.

Usage:

See also:

Implemented By:

Module Name	Source File	Last Revised
mod_fifo	mod_fifo.c	12782

fifo_orbit_announce

Description needed! Please contribute one.

Usage:

Example needed! Please contribute one.

See also:

Implemented By:

Module Name	Source File	Last Revised
mod_fifo	mod_fifo.c	8026

fifo_orbit_context

Sets the context for the fifo_orbit_exten when the que times out. This must be set before you place the caller in the que.

Usage:

<action application="set" data="fifo_orbit_context=MyContext"/>"

See also:

Implemented By:

Module Name	Source File	Last Revised
mod_fifo	mod_fifo.c	12782

fifo_orbit_exten

Set a destination extension and timeout, and it will cause FreeSWITCH to run the caller through the FIFO until the specified timeout, then the caller will get transferred to the destination extension. Can be set before launching the caller into the FIFO.

Usage: Use before a FIFO statement:

This will cause 60 seconds of time to pass with the caller in the FIFO before sending the call to the Voicemail extension specified.

See also:

Implemented By:

Module Name	Source File	Last Revised
mod_fifo	mod_fifo.c	8026

fifo_outbound_uuid

Description needed! Please contribute one.

Usage:

Example needed! Please contribute one.

See also:

Implemented By:

Module Name	Source File	Last Revised
mod_fifo	mod_fifo.c	10277

fifo_override_announce

Description needed! Please contribute one.

Usage:

Example needed! Please contribute one.

See also:

Implemented By:

Module Name	Source File	Last Revised
mod_fifo	mod_fifo.c	8014

fifo_pop_order

Description needed! Please contribute one.

Usage:

Example needed! Please contribute one.

See also:

Implemented By:

Module Name	Source File	Last Revised
mod_fifo	mod_fifo.c	8689

fifo_position

Description goes here

Usage:

Example needed

See also:

Implemented By:

mod_fifo.c

fifo_record_template

Description needed! Please contribute one.

Usage:

Example needed! Please contribute one.

See also:

Implemented By:

Module Name	Source File	Last Revised
mod_fifo	mod_fifo.c	8039

fifo_role

For reporting purposes, i.e. in the CDRs, the variable will contain "consumer" or "caller" depending upon the call leg.

Usage:

None

See also: Mod_fifo

fifo_timestamp

Contains the timestamp of when the caller was bridged to the consumer.

Usage:

None

See also:

Implemented By:

Module Name	Source File	Last Revised
mod_fifo	mod_fifo.c	8689

force_transfer_context

When handling transfer/REFER FreeSWITCH normally inherits the context from the original channel. This variable forces the context in which to handle the transfer/REFER

Usage:

See also:

Implemented By:

Module Name	Source File	Last Revised
core	switch_ivr.c	10466

force_transfer_dialplan

When handling transfer/REFER FreeSWITCH normally inherits the diaplan from the original channel. This variable forces the dialplan in which to handle the transfer/REFER

Usage:

Example needed! Please contribute one.

See also:

Implemented By:

Module Name	Source File	Last Revised
core	switch_ivr.c	10466

group_confirm_cancel_timeout

If set, cancels a leg timeout after the call is answered.

When using group confirm, a call passes through three phases:

Call is ringing.
Call is answered, waiting to be confirmed.
Call is confirmed and bridged.

Normally, a timeout on the leg will apply to phases 1 and 2, but the example below would apply the leg timeout only during phase 1

Usage:

group_confirm_file

This variable is used together with group_confirm_key. In group_confirm_file, you specify the wav file you want to play when the called party picks up the call. In the group_confirm_key variable, you define the DTMF that the called party should send to FS to bridge the call. If a wrong DTMF or no DTMF is sent, the called won't be bridged and the wav file will be repeated. Although this is the standard usage of these variables (group_confirm_key and group_confirm_file), they can be used in a more flexible manner. Please see Freeswitch_IVR_Originate#Answer_confirmation.

Usage:

Implemented By:

Module Name	Source File	Last Revised
core	switch_ivr_originate.c	4796

group_confirm_key

When using group_confirm_file to play a file asking the user if they want to actually accept the call, use the group_confirm_key for which key to bind to to actually connect the call.

Usage:

Implemented By:

Module Name	Source File	Last Revised
core	switch_ivr_originate.c	4796

hangup_after_bridge

Controls what happens to a calling (A) party when in a bridge state and the called (B) party hangs up. If "true" the dialplan will stop processing and the A leg will be terminated when the B leg terminates. If "false" (default) the dialplan continues to be processed after the B leg terminates. This is checked after park_after_bridge and transfer_after_bridge.

Usage:

See also:

Implemented By:

Module Name	Source File	Last Revised
mod_dptools	mod_dptools.c	8412
mod_enum	mod_enum.c	3495
mod_lcr	mod_lcr.c	10510
core	switch_ivr.c	5738
core	switch_ivr_bridge.c	10504

hangup_after_conference

Controls what happens to a calling (A) party when in a conference and the conference ends (e.g. endconf flag set and moderator leaves). If "true" (default) the dialplan will stop processing and the A leg will be terminated. If "false" the dialplan continues to be processed after the end of conference.

Usage:

See also:

Implemented By:

Module Name	Source File	Last Revised
mod_conference	mod_conference.c	7a7f386886a069

hangup_cause

This is set to the hangup cause of the A leg of the call (note that as such it doesn't make much sense before the end of the call). Often this will take the hangup cause from the B leg of the call, if there is one. For a list of hangup causes, see Hangup Causes.

Usage:

Implemented By:

Module Name	Source File	Last Revised
core	switch_core_state_machine.c	12784

hangup_cause_q850

This is set to the Q850 numeric code of the hangup cause of the A leg of the call (note that as such it doesn't make much sense before the end of the call). Often this will take the hangup cause from the B leg of the call, if there is one. For a list of hangup causes, see Hangup Causes.

Usage:

hold_events

It's a variable that display start and stop times for each hold.

Example:

This CDR shows that the phone was put on hold twice with hold start and stop time.

variable_hold_events: [{{1347487292379229,1347487293856872},{1347487288539686,1347487290757780}}]

hold_hangup_xfer_exten

Controls what happens to a calling (A) party when in a bridge state and the bridge ends while the called (B) party is on hold. If not set on leg B (ie. the default), then A leg is hung up. If it is set on leg B, then leg A is transferred to the given extension, as per transfer_after_bridge.

Usage:

Implemented By:

Module Name	Source File	Last Revised
core	switch_ivr_bridge.c	10 Dec 09

hold_music

Per-channel hold music. Supports all audio formats and audio streams. The hold_music variable can also be set globally at vars.xml.

Usage:

You can also set your hold_music to the special value "indicate_hold" instead of a music source and it will pass the hold req through but not the SDP.

or,

For multi-tenant environment, if you want to have a separate MOH for the phone with hold button (like Polycom) that utilizes RE-INVITE with no media ip addr (0.0.0.0) for hold, you can override the hold-music values in the sip profile parameter similar to the following example:

See also:

Implemented By:

Module Name	Source File	Last Revised
mod_fifo	mod_fifo.c	8065
mod_sofia	sofia_glue.c	5449
core	switch_ivr.c	8232
core	switch_ivr_play_say.c	8065

ignore_display_updates

Tells freeswitch not to send display UPDATEs to the leg of the call. (update_display)

Usage: From dialplan/default.xml:

To set on A-Leg

To set on B-Leg

See also:

Implemented By:

Module Name	Source File	Last Revised
mod_sofia	mod_sofia.c	??
mod_sofia	sofia.c	??

ignore_early_media

Default value is false. Controls if the call returns on early media or not. You may specify a value for ignore_early_media in the argument to the bridge application, using the { } syntax. (ignore_early_media may not be specified on a per-leg basis, using the [ ] syntax, as it specifically is a global variable to the originate session.)

Usage:

or,

Implemented By:

Module Name	Source File	Last Revised
core	switch_ivr_originate.c	4796

import

The import variable, when used before a bridge, imports the variables of the other channel on the actual channel.

Usage:

<action application="set" data="import=this_is_a_variable_name"/>

Implemented By:

Module Name	Source File	Last Revised
mod_lcr	mod_lcr.c	11976

inherit_codec

If late negotiation is on, and you set inherit_codec=true on the A leg, the negotiated codec of the B leg will be forced onto the A leg.

Usage:

See also:

Implemented By:

Module Name	Source File	Last Revised
core	switch_ivr_originate.c	11256

intercept_unanswered_only

If set to true, the leg will only be intercepted if the channel is not answered.

Default: false

Usage:

Implemented By:

Module Name	Source File	Last Revised
core	switch_ivr_bridge.c	git-36ba0f24

intercept_unbridged_only

If set to true, the leg will only be intercepted if the channel is not bridged to anyone.

Default: false

Usage:

Implemented By:

Module Name	Source File	Last Revised
core	switch_ivr_bridge.c	git-58fe45a

ivr_menu_status

ivr_menu_status, is a channel variable with options of success, failure or timeout

variable_ivr_menu_status: [success] variable_ivr_menu_status: [failure] variable_ivr_menu_status: [timeout]

ivr_menu_terminator

You can set to none or the dtmf chars you want to terminate input.

Usage:

jitterbuffer_msec

Activates the jitter buffer. The jitter buffer has three params: length, max length, and max drift.

Usage:

Or to set it on the subsequent outbound call: export sets a variable on both the current channel and on any channels it creates, the 'nolocal:' disables setting it on the current channel and only sets it on the subsequent outbound channels.

You can also activate the Jitter Buffer in the bridge as follows:

This will add a jitter buffer to packets flowing from a remote gateway towards a local freeswitch user. The network would look like this:

(local sip user) -----> FreeSWITCH -----> (remote gateway)

Where the link between the freeswitch and the remote gateway has a bad, jitter causing connection, and say the local sip user has no jitter buffering on their IP-phone. This will help the voice quality for the incoming audio.

Implemented By:

Module Name	Source File	Last Revised
mod_fax	mod_fax.c	9468
mod_sofia	sofia_glue.c	5114

last_bridge_hangup_cause

This is set to the hangup cause of the last bridged B leg of the call. For a list of hangup causes, see Hangup Causes.

Usage:

last_bridge_proto_specific_hangup_cause

This shows the last bride hangup cause by SIP response code, e.g. "sip:404"

last_matching_digits

Contains the last set of digits that the user dialed when using the "dmachine" digit-handling. This is most commonly used with the bind_digit_action dialplan application.

Usage:

See also:

Implemented By:

Module Name	Source File	Last Updated
mod_dptools	mod_dptools.c

last_transferred_conference

Contains the name of the last conference that this channel was connected to.

Usage:

See also:

Implemented By:

Module Name	Source File	Last Updated
mod_conference	mod_conference.c

leg_delay_start

You can specify a wait time for before each leg is called in a forked dial scenario. Useful for follow me dialplans. Leg delay start will ring an extension early if all other less delayed (or no delayed) legs have declined, failed, or are not available (as of FS 1.0.5). Leg_delay_start is best thought of as the minimum time to ring the other extensions prior to this one if they are able to ring.

Note that this doesn't work with Enterprise Originate. For Enterprise originate see variable originate_delay_start

Usage:

A more complex example with breakdown and timeline (seconds in brackets):

Assuming all users just let it ring:

[00] - user 1000 rings [08] - user 2302 rings [20] - user 1231231234 rings

Assuming user 1000 decline after 2 seconds, other users ring:

[00] - user 1000 rings [02] - user 2302 rings [14] - user 1231231234 rings

See also:

Implemented By:

Module Name	Source File	Last Revised
core	switch_ivr_originate.c	12403

leg_progress_timeout

Description needed! Please contribute one.

Usage:

Make all bridged calls fail over to the next in 6 seconds. <action application="export" data="leg_progress_timeout=6"/>

See also:

Implemented By:

Module Name	Source File	Last Revised
core	switch_ivr_originate.c	11286

leg_timeout

Timeout for each leg in an originate dialstring. Can be used in per-leg [], but not in global {} for the dialstring. For global, use originate_timeout.

You can also use leg_progress_timeout to specify the maximum time we will wait before we get media (whether its early media, ringing or answer), allowing you to avoid going to voicemail for a particular line.

If you are using group confirm then you can cancel the timeout by using the group_confirm_cancel_timeout channel variable. If leg_delay_start is also used, leg_timeout will not start the timeout counter until after the extension starts to be bridged to.

Usage:

See also:

Implemented By:

Module Name	Source File	Last Revised
core	switch_ivr_originate.c	11286

loopback_bowout_on_execute

Set to true to have one-legged loopback channels "bow out" of the call.

Usage:

See also: Freeswitch mailing list

bowout_on_execute is only useful to 1 leg calls you never have to set it. loopback_bowout is true by default it tries to cut loopback out by doing uuid_bridge

Comment: When setting "loopback_bowout=false", a bridged loopback call result in 4 legs (a-leg, loopback-a, loopback-b, b-leg).

When setting "loopback_bowout=true", a bridged loopback call results in 2 legs (a-leg, b-leg).

During call setup with "loopback_bowout=true", there will always be 4 legs however at the beginning. loopback-a and loopback-b will be destroyed when a-leg and b-leg are successfully bridged

Implemented By:

Module Name	Source File	Last Revised
mod_loopback	mod_loopback.c

loopback_export

A list of channel variables to pass from loopback-a to loopback-b.

Usage:

See also:

Implemented By:

Module Name	Source File	Last Revised
mod_loopback	mod_loopback.c

media_bug_answer_req

Start recording only when the channel has been answered.

NOTE: RECORD_ANSWER_REQ should be used on releases prior to 1.0.5, or builds prior to the revision 15235

Usage:

Implemented By:

Module Name	Source File	Last Revised
core	switch_ivr_async.c	15235
core	switch_core_media_bug.c	15235

min_dup_digit_spacing_ms

Duplicate inband DTMF that starts sooner than this time will be ignored. That is, this is the minimum gap from the end of the first digit and the start of the repeated digit required for two digits to be detected. This value is 0 by default. Set this variable prior to executing spandsp_start_dtmf.

Usage:

See also:

Channel Variables
Mod_spandsp

Implemented By:

Module Name	Source File	Last Updated
mod_spandsp	mod_spandsp_dsp.c

monitor_early_media_fail

Monitors early media for failure conditions, such as a busy signal. This allows faster processing of failed calls when ignoring early media.

The syntax is a series of ! delimited early media conditions in the following format:

condition_name:number_of_hits:tone_detect_frequencies

condition_name	user defined name for the error condition
number_of_hits	the number of times the tone must be heard before considering it a fail
tone_detect_frequencies	the frequencies to listen for (delimited by + instead of ,). See tone_detect

NOTE: this variable only works when ignore_early_media is set to true.

Usage:

Implemented By:

Module Name	Source File	Last Revised
core	switch_ivr_originate.c	11295

monitor_early_media_ring

Monitors early media for a user-specific ring tone. Each time the tone is heard, the switch will increment an internal counter for that leg. Once the counter reaches monitor_early_media_ring_total (or this variable has not been set) then the early media will be sent.

The syntax is a series of ! delimited early media conditions in the following format:

condition_name:number_of_hits:tone_detect_frequencies

condition_name	Optional? user-defined name for the error condition
number_of_hits	the number frequencies for the tone detector to find before considering it a hit. 1:400.0+480.0 means the ring count is incremented if 400hz OR 480hz is detected. 2:400.0+480.0 means the ring count is incremented if 400 hz AND 480 hz are detected.
tone_detect_frequencies	the frequencies to listen for (delimited by + instead of ,). Examples are 400.0+480.0 [for a US Ring] See tone_detect

NOTE: this variable only works when ignore_early_media is not present.

Usage:

This will bridge to a special conference that rings (US_RING) 10 times, and then plays the Star Wars Imperial March. You may optionally call 1239a@conference.freeswitch.org to test a UK_RING. A successful log looks like:

2010-10-22 10:26:53.975865 [DEBUG] switch_ivr_originate.c:614 sofia/internal/1239@conference.freeswitch.org setting ring total to 3 2010-10-22 10:26:55.007970 [DEBUG] switch_rtp.c:2544 Correct ip/port confirmed. 2010-10-22 10:26:56.427869 [DEBUG] switch_ivr_async.c:2424 TONE monitor_early_media_ring_2 HIT 1/2 2010-10-22 10:27:01.707869 [DEBUG] switch_ivr_async.c:2424 TONE monitor_early_media_ring_2 HIT 1/2 2010-10-22 10:27:02.667868 [DEBUG] switch_ivr_async.c:2424 TONE monitor_early_media_ring_2 HIT 2/2 2010-10-22 10:27:02.667868 [DEBUG] switch_ivr_async.c:2430 TONE monitor_early_media_ring_2 DETECTED 2010-10-22 10:27:02.667868 [DEBUG] switch_ivr_originate.c:352 Ring 1/3 2010-10-22 10:27:02.667868 [DEBUG] switch_ivr_async.c:2436 Re-enabling monitor_early_media_ring_2 2010-10-22 10:27:07.767869 [DEBUG] switch_ivr_async.c:2424 TONE monitor_early_media_ring_2 HIT 1/2 2010-10-22 10:27:13.487868 [DEBUG] switch_ivr_async.c:2424 TONE monitor_early_media_ring_2 HIT 1/2 2010-10-22 10:27:14.027868 [DEBUG] switch_ivr_async.c:2424 TONE monitor_early_media_ring_2 HIT 2/2 2010-10-22 10:27:14.027868 [DEBUG] switch_ivr_async.c:2430 TONE monitor_early_media_ring_2 DETECTED 2010-10-22 10:27:14.027868 [DEBUG] switch_ivr_originate.c:352 Ring 2/3 2010-10-22 10:27:14.027868 [DEBUG] switch_ivr_async.c:2436 Re-enabling monitor_early_media_ring_2 2010-10-22 10:27:20.147869 [DEBUG] switch_ivr_async.c:2424 TONE monitor_early_media_ring_2 HIT 1/2 2010-10-22 10:27:20.907868 [DEBUG] switch_ivr_async.c:2424 TONE monitor_early_media_ring_2 HIT 2/2 2010-10-22 10:27:20.907868 [DEBUG] switch_ivr_async.c:2430 TONE monitor_early_media_ring_2 DETECTED 2010-10-22 10:27:20.907868 [DEBUG] switch_core_media_bug.c:441 Removing BUG from sofia/internal/1239@conference.freeswitch.org 2010-10-22 10:27:20.911864 [INFO] switch_ivr_originate.c:3290 Sending early media

The last line is most important. The call continues, but early media is delayed from sending until monitor_early_media_ring_total is matched. If monitor_early_media_ring_total is not met, then the call will timeout.

anthm had the following to say in IRC

<anthm> The point of the feature is because of problems with providers who send a 183 then play a busy tone and do not indicate fail over sip <anthm> so it looks like a good call. <anthm> so the monitor is to confirm that it is a ring tone as expected.

<anthm> monitor_early_media_ring stops the default behavior to have originate return when it hears a ring <anthm> well when it gets 183 <anthm> cos you don't know if its a ring <anthm> so it listens for one <anthm> when it really hears one, it lets the channel accept the 183

Implemented By:

Module Name	Source File	Last Revised
core	switch_ivr_originate.c	11295

monitor_early_media_ring_total

Specifies the number of user defined ring tones that can be heard before failing. To be used with monitor_early_media_ring

Usage:

See also:

monitor_early_media_ring

Implemented By:

Module Name	Source File	Last Revised
core	switch_ivr_originate.c	11738

monitor_fail_dispo

This variable can be used to provide a custom originate_disposition as the result of an early media failure using monitor_early_media_fail or monitor_early_media_ring. If this variable is not set, a default value of 'monitor_early_media_fail' or 'monitor_early_media_ring' will be placed inoriginate_disposition

Usage:

See also:

Early Media
monitor_early_media_fail
monitor_early_media_ring

Implemented By:

Module Name	Source File	Last Revised
core	switch_ivr_originate.c	11437

no_throttle_limits

You set this variable to true on your outbound calls to have them not count in sps

Usage:

{no_throttle_limits=true}

originate_delay_start

You can specify a wait time in milliseconds before origination. This variable can be used in Enterprise originate where the variable leg_delay_start cannot be used.

Usage:

A more complex example with breakdown and timeline (seconds in brackets):

Assuming all users just let it ring:

[00] - user 1000 rings [10] - user 2302 rings [15] - user 1231231234 rings

Assuming user 1000 decline after 2 seconds, other users ring:

[00] - user 1000 rings [08] - user 2302 rings [14] - user 1231231234 rings

See also:

Implemented By:

Module Name	Source File	Last Revised
core	switch_ivr_originate.c	032761c

originate_disposition

Read Only. This is the originate disposition aka hangup cause returned. (LEG B)

It is also redefined after every bridge attempts if the bridge is not successful.

See also:

Implemented By:

Module Name	Source File	Last Revised
core	switch_ivr_originate.c	11437

originate_timeout

Determines how long a bridge or originate action action will stay in the "originate" state. In effect, it is a way to control the timeout for a bridge/originate consisting of multiple endpoints. Default value is 60.

Usage:

WARNING: Beware that if you are not using {ignore_early_media=true} call_timeout is no longer applicable as soon as early media signal is received.

See also:

Implemented By:

Module Name	Source File	Last Revised
core	switch_ivr_originate.c

originating_leg_uuid

Shows the uuid of the originating leg on an outbound channel

Usage:

In A-leg CDR:

In B-leg CDR:

Note that the leg uuid's are different. The call_uuid matches the two legs together, but the originating_leg_uuid can do so as well.

See also:

Originate

Implemented By:

Module Name	Source File	Last Updated
Core	switch_ivr_originate.c

origination_callee_id_name

Set on the inbound leg to control what caller ID name appears in the caller phone's display. Also see ignore_display_updates which affect the processing of these variables.

Usage:

If you find using set doesn't work, try using export instead.

origination_callee_id_number

Set on the inbound leg to control what caller ID number appears in the caller phone's display. Also see ignore_display_updates which affect the processing of these variables.

Usage:

origination_caller_id_name

Sets the origination callerid name (LEG A).

If you want to set the Caller ID on an origination call you should add this inside the {} brackets before the dialstring. Set it to "_undef_" if you want to remove it. Setting it to a null string \"\" may break your system.

For Snom 370/820 users:

If you want to display LEG A's name (if available) as soon as LEG B (here the local Snom) rings, you have to set origination_caller_id_name or effective_caller_id_name as described. Otherwise, in LEG B's display, you gonna see LEG A's number during ringing and switching to LEG A's name after picking up the call by LEG B.

Usage:

originate {origination_caller_id_name='Caller Name',origination_caller_id_number=5551231234}sofia/gateway/test/1231231234 &park()

See also:

effective_caller_id_name

Implemented By:

Module Name	Source File	Last Revised
mod_dptools	mod_dptools.c	10917
core	switch_ivr_originate.c	12604

origination_caller_id_number

Sets the origination callerid number. (LEG A)

If you want to set the Caller ID on an origination call you should add this inside the {} brackets before the dialstring.

Usage:

originate {origination_caller_id_name='Caller Name',origination_caller_id_number=5551231234}sofia/gateway/test/1231231234 &park()

See also:

Implemented By:

Module Name	Source File	Last Revised
mod_dptools	mod_dptools.c	10917
core	switch_ivr_originate.c	12604

origination_channel_name

Set this in the {} when doing an originate to create a custom channel name

Usage:

originate {origination_channel_name='this_is_my_channel_name'}loopback/9664 9195

See also:

Mod_commands#originate

Implemented By:

Module	Source File	Last Updated
core	switch_ivr_originate.c

origination_privacy

Sets privacy profile for caller. Options are any combination of "screen", "hide_name", "hide_number". Screen is on by default.

Note: screen is the keyword that makes the caller ID as P-Asserted-Identity vs P-Preferred-Identity.

Usage:

To do 3 at once:

Usage:

Note: There is no real separator; therefore, you can use separator to make it readable or nothing.

See also: Channel_Variables#sip_cid_type

Implemented By:

Module Name	Source File	Last Revised
core	switch_ivr_originate.c	12604

originator_codec

Sets the codec for calls originated from LEG A (setting the codec for LEG B). This will automatically be appended to the codec_string unless an absolute_codec_string has been set.

Usage:

See also:

Implemented By:

Module Name	Source File	Last Revised
mod_opal	mod_opal.cpp	10567
mod_sofia	sofia_glue.c	5114
core	switch_core_session.c	4796

outbound_redirect_fatal

When doing a simultaneous call to multiple endpoints, a 302 redirect can cause all the endpoints to stop ringing and the call will follow the redirect. When this channel variable is set it causes an endpoint that sends back a 302 redirect to be removed from the call list and the other endpoints continue to ring.

Usage:

See also:

Implemented By:

Module Name	Source File	Last Revised
Sofia	sofia.c

park_after_bridge

If set to true, it will park the call after mod_dptools: bridge returns. This is checked before transfer_after_bridge and hangup_after_bridge.

Default: false

Usage:

Implemented By:

Module Name	Source File	Last Revised
core	switch_ivr_bridge.c	12691

park_timeout

When set, a parked call will disconnect after the timeout has occurred. Timeout is specified in seconds. If no park_timeout value is set then the parked call will be held indefinitely or until it is removed with a uuid_transfer.

Usage:

You can also specify which hangup_cause you need when the channel is disconnected by park_timeout.

Usage: <action application="set" data="park_timeout=30:MEDIA_TIMEOUT"/>

See also:

park
uuid_transfer
Hangup_causes

Implemented By:

Module Name	Source File	Last Revised
mod_sofia	sofia.c	10391
core	switch_ivr.c	10391

pass_rfc2833

If set, it passes RFC 2833 DTMF's from one side of a bridge to the other, untouched. If unset, it decodes and re-encodes them before passing them on.

Note: this has no effect when bypass_media or proxy_media is set.

Usage:

See also:

Implemented By:

Module Name	Source File	Last Revised
mod_sofia	sofia_glue.c	5114

passthru_ptime_mismatch

If ptime from leg A and leg B don't match and if mod_com_g729 is used, the call would normally use the codec to re-packetize the RTP stream.

With this parameter, mod_com_g729 will re-packetize without decoding/encoding, as mod_g729 would do.

Usage:

This has to be set in {} before bridging. That will probably not work if set using export before bridging.

Note: It may also be set globally in vars.xml.

See also:

Implemented By:

Module Name	Source File	Last Revised
core	switch_channel.c	e7cafae29d0720f06280255a55cc94548e7e12a4
core	switch_core_io.c	e7cafae29d0720f06280255a55cc94548e7e12a4

process_cdr

Indicates how to process CDR records.

Can be undefined or set to "false", "true", "a_only", "b_only"

false - indicates to not process the record.
true - or undefined indicates the default behavior which is to process all CDR records.
a_only - indicates to only process CDR records on the inbound leg of a call.
b_only - indicates to only process CDR records on the outbound leg of a call.

This variable is unconditionally exported

Usage:

See also:

Implemented By:

Module Name	Source File	Last Revised
core	switch_core_session.c	7214
core	switch_core_state_machine.c	7212

profile_created_time

Contains the start time (in microseconds) of when the call profile was created.

Usage:

In the event that a call is transferred, this is the effective 'created_time' for that transfer.

For example, if you did the following call flow:

2000 > 2001 (created_time=1357138714 / profile_created_time=1357138714) transfer to 2002 (created_time=1357138714 / profile_created_time=1357138752) transfer to 2003 (created_time=1357138714 / profile_created_time=1357138766) transfer to 2004 (created_time=1357138714 / profile_created_time=1357138784)

You would still use 'progress_time' to retrieve the progress start time, no matter if it's a transfer leg or not.

See also:

created_time
progress_time

Implemented By:

Logic taken from here:

src/switch_channel.c:switch_channel_set_caller_profile() caller_profile->times->profile_created = switch_micro_time_now();

progress_time

TODO

Usage:

TODO

See also:

TODO

Implemented By:

TODO

proto_specific_hangup_cause

This variable will cause FreeSWITCH to force the SIP response code to a specific setting when hanging up a call. The example below is one where all possible extensions have been tested and failed and you want FreeSWITCH to generate and respond with a specific code. (This is not a passthrough example).

By the way, you'll be unable to rewrite the hangup cause for a bridge that gets a 180 or 183 packet from the gateway before getting a 4xx, 5xx or 6xx packet (because those bridges don't 'fail'). This happens with SIP providers that give a 183 Session Progress before a 404 Not Found if the PSTN number dialled doesn't exist.

Usage:

Example: SIP Response Map

See also:

Implemented By:

Module Name	Source File	Last Revised
mod_sofia	mod_sofia.c	10090
mod_sofia	sofia.c	12676

read_codec

Read only. The negotiated codec of the inbound call leg.

See also:

Implemented By:

Module Name	Source File	Last Revised
core	switch_core_codec.c	10045

RECORD_APPEND

Recording is appended to file. set RECORD_APPEND=true on the channel and all recordings will behave this way to formats which support it (curently mod_sndfile for WAV, etc.)

Usage:

RECORD_ARTIST

Set prior to performing a record to store in the file header meta data (provided the file format supports meta headers).

Usage:

See also:

Implemented By:

Module Name	Source File	Last Revised
mod_voicemail	mod_voicemail.c	6984
core	switch_ivr_async.c	4796
core	switch_ivr_play_say.c	4796

RECORD_BRIDGE_REQ

Record session only when the channel is bridged.

Usage:

RECORD_COMMENT

Set prior to performing a record to store in the file header meta data (provided the file format supports meta headers).

Usage:

See also:

Implemented By:

Module Name	Source File	Last Revised
mod_voicemail	mod_voicemail.c	6984
core	switch_ivr_async.c	4796
core	switch_ivr_play_say.c	4796

RECORD_COPYRIGHT

Set prior to performing a record to store in the file header meta data (provided the file format supports meta headers).

Usage:

See also:

Implemented By:

Module Name	Source File	Last Revised
mod_voicemail	mod_voicemail.c	6984
core	switch_ivr_async.c	4796
core	switch_ivr_play_say.c	4796

RECORD_DATE

Set prior to performing a record to store in the file header meta data (provided the file format supports meta headers).

Usage:

See also:

Implemented By:

Module Name	Source File	Last Revised
mod_voicemail	mod_voicemail.c	6984
core	switch_ivr_async.c	4796
core	switch_ivr_play_say.c	4796

RECORD_DISCARDED

If a recording gets dropped or discarded then this channel variable is set to true. Useful for diagnostics.

Usage:

N/A

See also:

Implemented By:

Module Name	Source File	Last Updated
core	switch_ivr_async.c	c465c435

record_fill_cng

Description need please.

Usage:

See also:

Variable record_waste_resources

Implemented By:

Module Name	Source File	Last Updated
Core	switch_ivr_play_say.c

RECORD_HANGUP_ON_ERROR

When set to true this channel variable will cause the call to hangup if there is an error when trying to record the call. This is not a common feature, however in cases where a call MUST be recorded it makes it impossible to have calls that are not recorded. (Useful in some business scenarios.)

Usage:

Implemented By:

Module Name	Source File	Last Updated
Core	switch_ivr_async.c	a3e6bead

RECORD_MIN_SEC

Sets the minimum recording length. Normally a recording must be at least 3 seconds long. If a recording does not meet the minimum length, it is deleted after being recorded.

Usage:

See also:

record_session

Implemented By:

Module Name	Source File	Last Revised
core	switch_ivr_async.c	15272

record_ms

Read Only. Contains the length in milliseconds of the most recent recording.

See also:

record

Implemented By:

Module Name	Source File	Last Revised
core	switch_ivr_play_say.c	15155

record_sample_rate

Set the sample rate of the recording.

Usage:

See also:

Implemented By:

Module Name	Source File	Last Revised
mod_dptools	mod_dptools.c	6706

RECORD_READ_ONLY

Record read stream only.

Usage:

See also:

RECORD_STEREO
RECORD_WRITE_ONLY

Implemented By:

Module Name	Source File	Last Revised
core	switch_ivr_async.c	15257

record_restart_limit_on_dtmf

When set to true it allows the person recording to press a DTMF key and extend the amount of time they have before the recording times out.

Usage:

See also:

Misc._Dialplan_Tools_record

Implemented By:

Module Name	Source File	Last Updated
Core	switch_ivr_play_say.c

record_sample_rate

Specify the sampling rate of the recorded file.

Usage:

See also:

Misc._Dialplan_Tools_record

Implemented By:

Module Name	Source File	Last Updated
Core	switch_ivr_play_say.c

record_samples

Read Only. Contains the number of audio samples in the most recent recording.

RECORD_SOFTWARE

Set prior to performing a record to store in the file header meta data (provided the file format supports meta headers).

Usage:

See also:

Implemented By:

Module Name	Source File	Last Revised
mod_voicemail	mod_voicemail.c	6984
core	switch_ivr_async.c	4796
core	switch_ivr_play_say.c	4796

RECORD_STEREO

Record leg A and leg B streams (i.e. the caller is recorded to the left channel and the reciever is recorded on right channel) into different channel in a stereo file.

Usage:

See also:

RECORD_WRITE_ONLY
RECORD_READ_ONLY
RECORD_STEREO_SWAP

Implemented By:

Module Name	Source File	Last Revised
core	switch_ivr_async.c	5772

RECORD_STEREO_SWAP

Allow to invert the recording channel when RECORD_STEREO variable is set to true. So the caller is recorded to the right channel and the receiver is recorded on left channel.

Usage:

See also:

RECORD_STEREO

Implemented By:

Module Name	Source File	Last Revised
core	src/switch_core_media_bug.c	2011-12-14

RECORD_TITLE

Set prior to performing a record to store in the file header meta data (provided the file format supports meta headers).

Usage:

See also:

Implemented By:

Module Name	Source File	Last Revised
mod_voicemail	mod_voicemail.c	6984
core	switch_ivr_async.c	4796
core	switch_ivr_play_say.c	4796

record_waste_resources

By default record doesn't send RTP packets. This is generally acceptable, but for longer recordings or depending on the RTP timer of your gateway, your channel may hangup with cause MEDIA_TIMEOUT. Setting this variable will 'waste' bandwidth by sending RTP even during recording. The value can be true/false/<desired silence factor>. By default the silence factor is 1400 if you set record_waste_resources=true.

Usage:

See also:

RECORD_WRITE_ONLY
RECORD_READ_ONLY

Implemented By:

Module Name	Source File	Last Revised
core	switch_ivr_play_say.c	13152

RECORD_WRITE_ONLY

Record write stream only.

Usage:

See also:

RECORD_STEREO
RECORD_READ_ONLY

Implemented By:

Module Name	Source File	Last Revised
core	switch_ivr_async.c	15257

recording_follow_transfer

Set to true if you want recording to continue after a transfer.

Example

rtcp_octet_count

Contains number of RTCP octets collected during the call.

Usage:

N/A

See also:

Variable_rtcp_packet_count
CDR

Implemented By:

Module Name	Source File	Last Updated
Mod_sofia	sofia_glue.c	d5ff3e04

rtcp_packet_count

Contains number of RTCP packets collected during the call.

Usage:

N/A

See also:

Variable_rtcp_octet_count
CDR

Implemented By:

Module Name	Source File	Last Updated
Mod_sofia	sofia_glue.c	d5ff3e04

rtp_disable_vad_in

Description needed! Please contribute one.

Usage:

Example needed! Please contribute one.

See also:

Implemented By:

Module Name	Source File	Last Revised
mod_sofia	sofia_glue.c	5364

rtp_disable_vad_out

Description needed! Please contribute one.

Usage:

Example needed! Please contribute one.

See also:

Implemented By:

Module Name	Source File	Last Revised
mod_sofia	sofia_glue.c	5364

rtp_enable_vad_in

Description needed! Please contribute one.

Usage:

Example needed! Please contribute one.

See also:

Implemented By:

Module Name	Source File	Last Revised
mod_sofia	sofia_glue.c	5364

rtp_enable_vad_out

Description needed! Please contribute one.

Usage:

Example needed! Please contribute one.

See also:

Implemented By:

Module Name	Source File	Last Revised
mod_sofia	sofia_glue.c	5364

sdp_m_per_ptime

Adds a new m= line for each distinct ptime in codec list.

When this variable is not set:

When mixing codecs with various ptime in a codec list, they will now be allowed to co-exist in the sdp but it will send no ptime attr. This means the ptime preferences on the offer will be ignored when mixing codecs with various ptimes. When receiving a codec list with no ptime attr, the ptime will be chosen from local preference instead of assuming 20ms. This means if offer contains PCMU with no ptime and FS has PCMU@40i
Dynamic payloads will now start at 98 and increment per additional dynamic codec per call. So now you can add CELT@32000h,CELT@48000h and each one will be auto-assigned a dynamic payload type.

Is now implied to be true, if you don't like this set it to false but its going to be undefined behaviour. This basically means if you call in with ptime 30 then you have a bunch of ptime 20 codecs in your outbound list that there will be one m= line with 30 and the original inbound codec and more m= lines for each discinct ptime in your list. This is, of course, will depend on disable_trancoding or absolute_codec_string as well

Usage:

See also:

Codecs

Implemented By:

Module Name	Source File	Last Updated
mod_sofia	sofia_glue.c	dfa78985

sdp_secure_savp_only

When sip_secure_media=true FreeSWITCH will offer both AVP and SAVP in the SDP. Setting sdp_secure_savp_only=true (in addition to sip_secure_media=true) will cause FreeSWITCH to offer only SAVP in the SDP.

Usage:

See also:

Implemented By:

Module Name	Source File	Last Updated
mod_sofia	sofia_glue.c	097d9a1

send_silence_when_idle

When greater than 0, this variable tells FreeSWITCH to transmit comfort noise when idle. The value of this variable is set to the divisor of the silence generating function. 400 or 1400 are common values set, but you may experiment with other choices to pick one that sounds best.

When true, FreeSWITCH will pick a default comfort noise value.

When -1, FreeSWITCH will transmit silence without comfort noise.

Usage:

See also:

bridge_generate_comfort_noise

Implemented By:

Module Name	Source File	Last Revised
core	switch_ivr.c	11274
core	switch_ivr_originate.c	10664

session_in_hangup_hook

Allows channel variables to be accessible in the api_hangup_hook that gets executed for the channel.

Usage:

See Lua#Special_Case:_env_object for an example of how to use this.

See also:

Implemented By:

Module Name	Source File	Last Revised
core	switch_core_state_machine.c	10468

signal_bond

UUID of the channel this channel is bridged/bonded to. Not present on a one-legged call.

Usage:

Example needed! Please contribute one.

See also:

Implemented By:

Module Name	Source File	Last Revised
mod_dptools	mod_dptools.c	7885
mod_loopback	mod_loopback.c	9646
mod_sofia	mod_sofia.c	7103
mod_sofia	sofia.c	11562
mod_sofia	sofia_glue.c	8232
core	switch_channel.c	7546
core	switch_core_session.c	4944
core	switch_ivr.c	7870
core	switch_ivr_async.c	7075
core	switch_ivr_bridge.c	9603
core	switch_ivr_play_say.c	8065

sip_acl_authed_by

Contains the name of the ACL node that authorized this call.

Usage:

None.

See also:

Implemented By:

Mod_sofia

sip_acl_token

Contains the ACL auth token for the current call.

Usage:

N/A

See also:

[Mod_sofia]

Implemented By:

[Mod_sofia]

sip_auth_password

For mod_sofia use with sip_auth_username to answer auth challenges without defining a full gateway.

Usage:

originate {sip_auth_username=<your_user_name>,sip_auth_password=<your_password>}sofia/external/1xxxxxxx@12.34.56.78 &echo

See also:

sip_auth_username

Implemented By:

Module Name	Source File	Last Revised
mod_sofia	sofia_reg.c	12819

sip_auth_username

For mod_sofia answer auth challenges without defining a full gateway. Used in tandem with sip_auth_password. Also indicates the SIP username a device successfully registered to FreeSWITCH with.

Usage:

originate {sip_auth_username=<your_user_name>,sip_auth_password=<your_password>}sofia/external/1xxxxxxx@12.34.56.78 &echo

Notes:

This should contain the username of the authenticated user that has triggered this request, if applicable.

Depending on how your FreeSWITCH instance is configured, you may experience problems with this variable being incorrect or blank after calling the 'bridge' application. If this happens, you may want to force the channel variable, for example;

See also:

sip_auth_password
sip_authorized

Implemented By:

Module Name	Source File	Last Revised
mod_sofia	sofia_reg.c	12819

sip_authorized

sip_authorized indicates whether the SIP device accessing the dialplan is authorized to FreeSWITCH or not. The SIP device can be authorized either via an ACL or via digest authentication.

Usage:

Example needed! Please contribute one.

See also:

sip_auth_username
sip_acl_authed_by
sip_acl_token_vars

Implemented By:

Module Name	Source File	Last Revised
mod_sofia	sofia.c	5299

sip_auto_simplify

When set, this directs FreeSWITCH to remove itself from the SIP signaling path if it can safely do so.

Usage:

See also:

uuid_simplify

sip_callee_id_name

DEPRECATED. Use variable origination_callee_id_name instead

sip_callee_id_number

DEPRECATED. Use variable effective_callee_id_number instead

<anthm>: sip_callee_id_X was deprecated in favor of effective_callee_id_X <anthm>: effective and sip are the same it was just a mistake to call it sip in case we use it for other protocols so both work for now

sip_callee_id_number

Set on the inbound leg to control what caller ID number appears in the caller phone's display.

Usage:

If you find using set doesn't work, try using export instead.

See also:

sip_callee_id_name

Implemented By:

Module Name	Source File	Last Revised
mod_sofia	sofia.c	15110
mod_sofia	mod_sofia.c	15211

sip_cid_in_1xx

Prevents FreeSWITCH when it receives 183 from leg-B to automatically insert RPID before sending 183 to leg-A.

Usage:

See also: This can be defined profile-wide with: pass-callee-id=false

sip_cid_type

Modify caller ID in SDP header. Can be set to none, default, pid, or rpid (default).

rpid=Remote-Party-ID header pid=P-Asserted-Identity header none=Caller ID will be in the SIP "From" Note: for gateways, this will not work. You will require caller-id-in-from=true in the gateway settings

Usage:

{sip_cid_type=none}sofia/default/user@example.com

Or,

{sip_cid_type=rpid}sofia/default/user@example.com i.e. Remote-Party-ID: "9094445555" <sip:9094445555@1.2.3.4>;party=calling;screen=yes;privacy=off

Or,

{sip_cid_type=pid}sofia/default/user@example.com i.e. P-Asserted-Identity: "9094445555" <sip:9094445555@1.2.3.4>

Or,

{sip_cid_type=rpid,origination_caller_id_name=test,origination_caller_id_number=1234}sofia/default/user@example.com

See also:

Implemented By:

Module Name	Source File	Last Revised
mod_sofia	sofia.c	12427
mod_sofia	sofia_glue.c	12427

sip_copy_multipart

FreeSWITCH supports only INVITEs with multipart bodies. FreeSWITCH does not support response messages such as 183 Session Progress or 200 Ok with multipart bodies. Typically SIP bodies only have one MIME part with an SDP using MIME type application/sdp. The SIP spec allows for multiple bodies defined with MIME type multipart/mixed. In this case FreeSWITCH will do it's best to find the MIME part with the SDP and parse that as it normally does. However, you can change FreeSWITCH behavior with multipart bodies and bridge using this variable.

Usage:

To have FreeSWITCH keep the multiple MIME parts intact when using bridge (default):

NOTE: FreeSWITCH will "do the right thing" and attach an application/sdp type generated by FreeSWITCH (per your settings) for the B leg as it normally would. The other non-SDP MIME parts just pass through.

To have FreeSWITCH strip the multiple MIME parts when using bridge:

sip_enable_soa

For per call basis which can be set to false to disable SIP SOA from sofia and most likely result in untouched exchange of SDP.

Usage:

sip_execute_on_image

Execute an application as soon as you get a T.38 invite. This variable is similar to execute_on_answer. You can run t38_gateway or rxfax etc when you get a T.38 re-invite but no CNG tone, or you want to ignore the tone and only react when getting a T.38 re-invite.

Usage: Need an example.

See also:

execute_on_answer
mod_dptools

Implemented By:

Module Name	Source File	Last Revised
mod_sofia	sofia_glue.c

sip_force_audio_fmtp

Set the audio fmtp.

Usage:

Please add example if you have one.

See also:

Implemented By:

Module Name	Source File	Last Updated
mod_sofia	mod_sofia.c	6360264f

sip_from_display

Description needed! Please contribute one.

Usage:

Example needed! Please contribute one.

See also:

Implemented By:

Module Name	Source File	Last Revised
mod_sofia	sofia_glue.c	12566

sip_hangup_disposition

This variable contains the value of who sent the SIP BYE message. Some examples from XML CDRs:

<sip_hangup_disposition>send_bye</sip_hangup_disposition> <sip_hangup_disposition>recv_bye</sip_hangup_disposition> <sip_hangup_disposition>send_refuse</sip_hangup_disposition> <sip_hangup_disposition>send_cancel</sip_hangup_disposition>

Interpretation of these values differs on incoming and outgoing calls since FreeSWITCH is at different ends of the session.

Value	Incoming	Outgoing
send_bye	FS sent BYE to the caller (we hung up)	FS sent BYE to the endpoint (we hung up)
recv_bye	FS received BYE from the caller (they hung up)	FS received BYE from the endpoint (they hung up)
send_refuse	FS rejected the call (e.g. 4xx or 5xx)	Endpoint rejected the call (e.g. 4xx or 5xx)
send_cancel	n/a	FS aborted the call (we sent CANCEL)

Usage:

Look in CDR for the value; only valid after call has ended.

See also:

Implemented By:

Module Name	Source File	Last Revised
mod_sofia	mod_sofia.c
mod_sofia	sofia.c

sip_ignore_183nosdp

Ignoring 183 w/o SDP. This option is not for normal basic call flow.

Usage:

See also:

Channel_Variables#sip_ignore_183nosdp

sip_ignore_reinvites

Tells FreeSWITCH to accept/ignore re-INVITEs from remote end.

Usage:

Don't allow any re-INVITEs once bridged.

See also:

Implemented By:

Module Name	Source File	Last Revised
mod_sofia	sofia.c	9634

sip_invite_contact_params

Description needed! Please contribute one.

Usage:

Example needed! Please contribute one.

See also:

Implemented By:

Module Name	Source File	Last Revised
mod_sofia	sofia_glue.c	12290

sip_invite_domain

Set the from domain in leg (B).

Usage:

See also:

Implemented By:

Module Name	Source File	Last Revised
mod_sofia	sofia_glue.c	9127

sip_invite_from_params

Sets the from parameters on the B-leg of the call. The from parameters come after user@host:port and before '>'. The initial semi-colon is added after the port.

Usage:

{sip_invite_from_params=otg=mytrunk}sofia/gateway/sonus/$1

Result:

From: <sip:5552345678@sonus:5060;otg=mytrunk>;tag=blah

See also:

Implemented By:

Module Name	Source File	Last Revised
mod_sofia	sofia_glue.c	12287

ip_invite_params

Description needed! Please contribute one.

Usage:

Example needed! Please contribute one.

See also:

Implemented By:

Module Name	Source File	Last Revised
mod_sofia	mod_sofia.c	11069
mod_sofia	sofia_glue.c	7861

sip_invite_req_uri

Sets the uri in the header Request-Line INVITE when calling bridge or originate.

NOTE: RFC 3261 specifies that compliant endpoints SHOULD route based on the Request URI, not the URI in To:

Usage:

Result:

INVITE sip:11112222@test1.com SIP/2.0 From: "" <sip:0000000000@192.168.2.7>;tag=N6K579y4g6j0D To: <sip:33334444@192.168.4.6>

See also:

Implemented By:

Module Name	Source File	Last Revised
mod_sofia

sip_invite_route_uri

Sets the uri in the header Route when calling bridge or originate.

Usage:

originate {sip_invite_route_uri=<sip:+48399999999@10.0.0.51:5080;lr;orig>,origination_caller_id_number=399999000}sofia/internal/+48399999999@domain.com &echo

Result:

INVITE sip:+48399999999@domain.com SIP/2.0 Via: SIP/2.0/UDP 10.0.0.51;rport;branch=z9hG4bKpmFv4aXv4tKcK Route: <sip:+48399999999@10.0.0.51:5080;lr;orig>

Implemented By:

Module Name	Source File	Last Revised
mod_sofia	sofia_glue.c

sip_invite_tel_params

Tel URI parameters (npdi, rn) appearing in SIP URI on outbound calls.

Usage:

Result:

INVITE sip:555000002;npdi=yes;rn=5555550001@1.2.3.4;user=phone SIP/2.0

Implemented By:

Module Name	Source File	Last Revised
mod_sofia	sofia_glue.c	a3786d5

sip_invite_to_params

Description needed! Please contribute one.

Usage:

Example needed! Please contribute one.

See also:

Implemented By:

Module Name	Source File	Last Revised
mod_sofia	sofia_glue.c	12287

sip_invite_to_uri

Sets the uri in the header To when calling bridge or originate.

Usage:

originate {sip_invite_to_uri=<sip:11112222@test1.com>}sofia/internal/33334444@192.168.4.6 &park

Result:

INVITE sip:33334444@192.168.4.6 SIP/2.0 From: "" <sip:0000000000@192.168.2.7>;tag=N6K579y4g6j0D To: <sip:11112222@test1.com>

Alternatively, if you need to set just the username in the header To, you can pass it at the end of the dial string:

originate sofia/internal/33334444@192.168.4.6^11112222 &park

Result:

INVITE sip:33334444@192.168.4.6 SIP/2.0 From: "" <sip:0000000000@192.168.2.7>;tag=Qr6pB00BBrZ5m To: <sip:11112222@@192.168.4.6>

See also:

Implemented By:

Module Name	Source File	Last Revised
mod_sofia

rtp_jitter_buffer_during_bridge

Enables/disables the jitter buffer during bridge.

Usage:

or,

See also:

jitterbuffer

Implemented By:

Module Name	Source File	Last Revised
mod_sofia	mod_sofia.c

sip_jitter_buffer_plc

Enables/disables packet loss concealment (PLC) when using the jitter buffer. PLC is enabled by default when the jitter buffer is enabled. Set this variable before enabling the jitter buffer for it to have an effect.

Usage:

or,

See also:

jitterbuffer

sip_local_sdp_str

Description goes here

Usage:

Example needed.

See also:

Implemented By:

Module Name	Source File	Last Revised
mod_sofia	sofia_glue.c	?
mod_sofia	sofia_glue.c	?

sip_mirror_remote_audio_codec_payload

To tell sip to break the rfc and expect the codec payload the other side replies with rather than the one it offered which is the correct behavior.

Usage:

Examples needed!!!

Note: This variable can be set globally or per channel.

See also:

sip_network_destination

It is intended for use with devices registering behind a NAT where the Request-URI should contain the contact that was bound to the AOR during the registration request while the request itself should be sent to the public IP and port number of the NAT "pinhole". It does not add a Route header field to the request like the ;fs_path= or the sip_route_uri options do.

Usage:

See also:

Implemented By:

Module Name	Source File	Last Revised
mod_sofia	sofia_glue.c	12409

sip_recovery_break_rfc

To NOT reverse the from and to on UAS Re-INVITEs. This breaks RFC.

Usage:

Examples needed!!!

Note: This variable can be set globally or per channel.

sip_renegotiate_codec_on_reinvite

Allow SDP codec change with re-INVITE

Usage:

Note: It may also be set globally in vars.xml or set it in sip profiles (<param name="renegotiate-codec-on-reinvite" value="true"/>)

sip_wait_for_aleg_ack

When you set the variable sip_wait_for_aleg_ack on the b leg in the {} for the outbound call, this should make the B leg delay sending the ACK until it sees that the A leg has recv'd an ack.

Usage:

skeleton

This is an example of how to create a channel variable page. This section is the description of the variable. Put the description information here and then the usage example below.

Usage:

See also:

__Channel Variables
Add_Your_See_Other_Items_Here

Implemented By:

Module Name	Source File	Last Updated
Leave blank if you don't know itLeave blank if you don't know it		Leave blank if you don't know it

skip_cdr_causes

This is a list of call hangup causes that should not trigger cdr processing.

Implemented By:

Module Name	Source File	Last Revised
mod_sofia	sofia.c	2012-09-19, git commit 3cf238fc9a1370a45d362c2193b8d3634ccd1d11

spandsp_dtmf_rx_filter_dialtone

Sets the filter dialtone parameter in the spandsp DTMF detector. Dialtone filtering is disabled by default. Set this variable prior to executing spandsp_start_dtmf.

Usage:

See also:

Implemented By:

Module Name	Source File	Last Revised
mod_spandsp	mod_spandsp_dsp.c

spandsp_dtmf_rx_reverse_twist

Sets the reverse twist setting in the spandsp DTMF detector. Reverse twist is set to 4 dB by default. This value can be safely increased up to 6 or 7 without a significant increase in talk-off to allow DTMFs that exceed this threshold to be detected. Set this variable prior to executing spandsp_start_dtmf.

Usage:

See also:

Implemented By:

Module Name	Source File	Last Revised
mod_spandsp	mod_spandsp_dsp.c

spandsp_dtmf_rx_threshold

Sets the threshold parameter in the spandsp DTMF detector. Threshold is set to -42 dBm0 by default. Set this variable prior to executing spandsp_start_dtmf.

Usage:

See also:

Implemented By:

Module Name	Source File	Last Revised
mod_spandsp	mod_spandsp_dsp.c

spandsp_dtmf_rx_twist

Sets the twist parameter in the spandsp DTMF detector. Twist is set to 8 dB by default. Set this variable prior to executing spandsp_start_dtmf.

Usage:

See also:

Implemented By:

Module Name	Source File	Last Revised
mod_spandsp	mod_spandsp_dsp.c

suppress_cng

Enable/Disable comfort noise. (RTP/AVP 13)

You may wish to suppress comfort noise (which actually supresses VAD) if the call audio is being compromised by aggressive or poorly implemented VAD+silence suppression (eg. short words are missed in conversation)

Usage:

See also:

suppress-cng profile param

Implemented By:

Module Name	Source File	Last Revised
mod_spandsp	mod_spandsp_dsp.c

switch_m_sdp

The B-leg remote SDP.

Usage:

Used to store the remote SDP used by the other leg/channel of a call. (In the A-leg that will be the remote SDP of the B-leg.)
This variable is set, but never used, by FreeSWITCH. ("read-only")
Known as SWITCH_B_SDP_VARIABLE in mod_sofia and the core switch code

See also:

Implemented By:

Module Name	Source File	Last Revised
mod_sofia	mod_sofia.c	8372
mod_sofia	sofia.c	10463
mod_sofia	sofia_glue.c	5114
core	switch_core_session.c	4796

switch_r_sdp

Read only. This variable holds the remote SDP for the current leg/channel.

Usage:

Rewriting SDP:

note: Don't add a carriage return after "set"> or you'll end up writing a variable with a different name, leaving switch_r_sdp with the same value.

Known as SWITCH_R_SDP_VARIABLE in source code

See also:

Implemented By:

Module Name	Source File	Last Revised
mod_sofia	mod_sofia.c	8542
mod_sofia	sofia.c	9591
core	switch_core_session.c	4796

temp_hold_music

This variable specifies a hold music value that gets played to a caller only until they get transferred. After the transfer, the hold_music variable will apply.

Usage:

See also:

Implemented By:

Module Name	Source File	Last Revised
Core	switch_channel.c

timer_name

If set will make playback and speak use a timer to clock the audio instead of the read.

Usage:

See also:

Implemented By:

Module Name	Source File	Last Revised
Core	switch_ivr_play_say.c	4796

timezone

Sets the timezone for this particular call. Can be used, e.g., to set the timezone for a caller who is checking his/her voicemail. The value is expressed in Linux timezone format (ex. America/New_York -- see /usr/share/zoneinfo/zone.tab for the standard list of Linux timezones).

Note that this channel variable is only respected by the phrase layer -- ie, if you're using the 'say' application to announce times, this will work fine.

You can set the time zone Globally for Freeswitch in /conf/vars.xml by adding this line: <X-PRE-PROCESS cmd="set" data="timezone=America/Toronto"/> (of course replace the 'America/Toronto' with your own time zone.

If you would like to specify the time zone in the dialplan add <action application="set" data="timezone=America/Toronto"/> to your dialplan. (again replace with the proper timezone)

Finally you can specify a time zone for a particular extension in /conf/directory/default/'extension'.xml

Usage:

or,

Directory Usage:

See also:

Time_of_Day_Routing

Implemented By:

Module Name	Source File	Last Revised
mod_say_en	mod_say_en.c	10214

tod_tz_offset

Sets the GMT offset to be used on this call for tod based conditions.

NOTE: the variable must actually be set before the comparison, so either set it inline, transfer, or set it in the user directory.

You can set the offset Globally for Freeswitch in /conf/vars.xml by adding this line: <X-PRE-PROCESS cmd="set" data="tod_tz_offset=5"/> (of course replace the '5' with your GMT offset.

If you would like to specify the time offset in the dialplan add <action application="set" data="tod_tz_offset=5"/> to your dialplan. (again replace with the proper offset)

Finally you can set the variable for a particular extension in /conf/directory/default/'extension'.xml

Usage:

Directory Usage:

See also:

Time_of_Day_Routing

transfer_after_bridge

This variable can control what happens when a call is hang up. This can be used in conjunction with mod_fifo to control the "agent", possibly sending them back to an agent queue. This is checked after park_after_bridge and before hangup_after_bridge.

Usage:

or,

(Note the : separator)

See also:

Implemented By:

Module Name	Source File	Last Revised
core	switch_ivr_bridge.c	12691

transfer_on_fail

Allows you to tranfer call flow when a called party can not be reached for specific reasons ( unallocated_number, etc )

Usage:

In this example, if you were to attempt a bridge that resulted in "UNALLOCATED_NUMBER" , the call flow would be "transferred" to the "UNALLOCATED_NUMBER" destination in your current dialplan ( probably default xml dialplan )

or,

or,

<hangupcause> can be the string "auto_cause" which will be replaced by the hangup cause returned by the bridge. this is the default action if no destination, dialplan or context are specified.

<action application="set" data="hangup_on_bridge=true"/>
<action application="set" data="continue_on_fail=true"/>
<action application="set" data="transfer_on_fail=UNALLOCATED_NUMBER auto_cause xml error"/>
<action application="bridge" data="A..."/>
<action application="bridge" data="B..."/>

then have the an "Error" context in your dialplan.

<context name="error">
  <extension name="UNALLOCATED_NUMBER" continue="true">
    <condition field="${originate_disposition}" expression="UNALLOCATED_NUMBER" continue="false" break="on-true">
      <action application="playback" data="/usr/local/freeswitch/sounds/NotInService.wav"/>
      <action application="hangup" data="NORMAL_CLEARING"/>
    </condition>
  </extension>
</context>

See also:

transfer_to

Description needed

Example: Example needed

See also:

mod_cdr_csv

uuid_bridge_continue_on_cancel

When set to true causes the system to move on in the dialplan if it hits a bad b-leg. Default is false because this behavior is probably not recommended.

You may find this variable useful when implementing Dialplan_FollowMe

Usage:

See also:

Implemented By:

Module Name	Source File	Last Revised
core	switch_ivr_bridge.c	17044

verbose_sdp

Enable this variable to fix a bug in certain VoIP phones.

The a=rtpmap lines are optional (RFC 3264) for static payload types according to the SDP standard, but not all phones implement this correctly and fail if these lines are missing.

By default FreeSWITCH will omit these lines so that the SDP is smaller (which lowers bandwidth use and avoids packet fragmentation). If this variable is set to true, FS will instead send these a=rtpmap for all codecs.

For example, Polycom phone requires you to list all codecs in the RTP map even though the SDP is valid, but they seem not to look at any codecs that are not in the map.

Usage: in vars.xml

verbose_sdp=true

See also:

Implemented By:

Module Name	Source File	Last Revised
mod_sofia	sofia_glue.c	n/a

write_codec

Read only. The negotiated codec of the outbound call leg.

See also:

Implemented By:

Module Name	Source File	Last Revised
core	switch_core_codec.c	10045

data

Description needed! Please contribute one.

Usage:

Example needed! Please contribute one.

See also:

Implemented By:

Module Name	Source File	Last Revised
mod_dptools	mod_dptools.c	7402

endpoint_disposition

Description needed! Please contribute one.

Usage:

Example needed! Please contribute one.

See also:

Implemented By:

Module Name	Source File	Last Revised
mod_sofia	mod_sofia.c	7779
mod_sofia	sofia.c	12652
mod_sofia	sofia_glue.c	5114
core	switch_channel.c	10098

fax_document_transferred_pages

Description needed! Please contribute one.

Usage:

Example needed! Please contribute one.

See also:

Implemented By:

Module Name	Source File	Last Revised
mod_fax	mod_fax.c	10506

fax_header

Description needed! Please contribute one.

Usage:

Example needed! Please contribute one.

See also:

Implemented By:

Module Name	Source File	Last Revised
mod_fax	mod_fax.c	9468

fax_start_page

Description needed! Please contribute one.

Usage:

Example needed! Please contribute one.

See also:

Implemented By:

Module Name	Source File	Last Revised
mod_fax	mod_fax.c	9468

fifo_music

Description needed! Please contribute one.

Usage:

Example needed! Please contribute one.

See also:

Implemented By:

Module Name	Source File	Last Revised
mod_fifo	mod_fifo.c	8065

fifo_priority

Description needed! Please contribute one.

Usage:

Example needed! Please contribute one.

See also:

Implemented By:

Module Name	Source File	Last Revised
mod_fifo	mod_fifo.c	8689

fifo_serviced_by

Description needed! Please contribute one.

Usage:

Example needed! Please contribute one.

See also:

Implemented By:

Module Name	Source File	Last Revised
mod_fifo	mod_fifo.c	8012

fifo_serviced_uuid

Description needed! Please contribute one.

Usage:

Example needed! Please contribute one.

See also:

Implemented By:

Module Name	Source File	Last Revised
mod_fifo	mod_fifo.c	11356

fifo_status

The status of the consumer or caller. Usually "WAITING" or "TALKING".

Usage:

None

See also:

Implemented By:

Module Name	Source File	Last Revised
mod_fifo	mod_fifo.c	9860

fifo_strategy

Description needed! Please contribute one.

Usage:

Example needed! Please contribute one.

See also:

Implemented By:

Module Name	Source File	Last Revised
mod_fifo	mod_fifo.c	8028

fifo_target

Description needed! Please contribute one.

Usage:

Example needed! Please contribute one.

See also:

Implemented By:

Module Name	Source File	Last Revised
mod_fifo	mod_fifo.c	8689

fire_asr_events

If set, fire an event when speech is detected.

Usage:

Example needed! Please contribute one.

See also:

Implemented By:

Module Name	Source File	Last Revised
core	switch_ivr_async.c	7402

flow_billmsec

Description needed! Please contribute one.

Usage:

Example needed! Please contribute one.

See also:

Implemented By:

Module Name	Source File	Last Revised
core	switch_channel.c	6917

flow_billsec

Description needed! Please contribute one.

Usage:

Example needed! Please contribute one.

See also:

Implemented By:

Module Name	Source File	Last Revised
core	switch_channel.c	6917

flow_billusec

Description needed! Please contribute one.

Usage:

Example needed! Please contribute one.

See also:

Implemented By:

Module Name	Source File	Last Revised
core	switch_channel.c	6917

funny_stun

Description needed! Please contribute one.

Usage:

Example needed! Please contribute one.

See also:

Implemented By:

Module Name	Source File	Last Revised
mod_sofia	sofia_glue.c	9088

group_context

Description needed! Please contribute one.

Usage:

Example needed! Please contribute one.

See also:

Implemented By:

Module Name	Source File	Last Revised
mod_dptools	mod_dptools.c	10917

has_t38

Description needed! Please contribute one.

Usage:

Example needed! Please contribute one.

See also:

Implemented By:

Module Name	Source File	Last Revised
mod_sofia	sofia_glue.c	11006

holding_uuid

Description needed! Please contribute one.

Usage:

Example needed! Please contribute one.

See also:

Implemented By:

Module Name	Source File	Last Revised
mod_dptools	mod_dptools.c	7885
core	switch_ivr_originate.c	8475

id

Description needed! Please contribute one.

Usage:

Example needed! Please contribute one.

See also:

Implemented By:

Module Name	Source File	Last Revised
mod_voicemail	mod_voicemail.c	12887

inbound_dialplan

Description needed! Please contribute one.

Usage:

Example needed! Please contribute one.

See also:

Implemented By:

Module Name	Source File	Last Revised
mod_sofia	sofia.c	6113

instant_ringback

When set, ringback will not wait for indication before sending ringback tone to calling party.

Usage:

<action application="set" data="instant_ringback=true" />

See also:

Implemented By:

Module Name	Source File	Last Revised
core	switch_ivr_originate.c	11740

is_outbound

Description needed! Please contribute one.

Usage:

Example needed! Please contribute one.

See also:

Implemented By:

Module Name	Source File	Last Revised
core	switch_channel.c	10115

language

Description needed! Please contribute one.

Usage:

Example needed! Please contribute one.

See also:

Implemented By:

Module Name	Source File	Last Revised
mod_dptools	mod_dptools.c	10516
core	switch_ivr_play_say.c	8054

last_app

Description needed! Please contribute one.

Usage:

Example needed! Please contribute one.

See also:

Implemented By:

Module Name	Source File	Last Revised
core	switch_channel.c	6917

last_arg

Description needed! Please contribute one.

Usage:

Example needed! Please contribute one.

See also:

Implemented By:

Module Name	Source File	Last Revised
core	switch_channel.c	6917

last_dtmf_duration

Description needed! Please contribute one.

Usage:

Example needed! Please contribute one.

See also:

Implemented By:

Module Name	Source File	Last Revised
core	switch_core_io.c	9526

last_file_position

Description needed! Please contribute one.

Usage:

Example needed! Please contribute one.

See also:

Implemented By:

Module Name	Source File	Last Revised
mod_spidermonkey	mod_spidermonkey.c	5498

lcr_auto_route

Description needed! Please contribute one.

Usage:

Example needed! Please contribute one.

See also:

Implemented By:

Module Name	Source File	Last Revised
mod_lcr	mod_lcr.c	10510

lcr_route_count

Description needed! Please contribute one.

Usage:

Example needed! Please contribute one.

See also:

Implemented By:

Module Name	Source File	Last Revised
mod_lcr	mod_lcr.c	10510

left_hanging_extension

Description needed! Please contribute one.

Usage:

Example needed! Please contribute one.

See also:

Implemented By:

Module Name	Source File	Last Revised
mod_sofia	mod_sofia.c	7425

limit_id

Description needed! Please contribute one.

Usage:

Example needed! Please contribute one.

See also:

Implemented By:

Module Name	Source File	Last Revised
mod_db	mod_db.c

limit_max

Description needed! Please contribute one.

Usage:

Example needed! Please contribute one.

See also:

Implemented By:

Module Name	Source File	Last Revised
mod_db	mod_db.c

limit_rate

Description needed! Please contribute one.

Usage:

Example needed! Please contribute one.

See also:

Implemented By:

Module Name	Source File	Last Revised
mod_hash	mod_hash.c

limit_realm

Description needed! Please contribute one.

Usage:

Example needed! Please contribute one.

See also:

Implemented By:

Module Name	Source File	Last Revised
mod_db	mod_db.c

limit_usage

Description needed! Please contribute one.

Usage:

Example needed! Please contribute one.

See also:

Implemented By:

Module Name	Source File	Last Revised
mod_db	mod_db.c
mod_hash	mod_hash.c

local_media_ip

Description needed! Please contribute one.

Usage:

Example needed! Please contribute one.

See also:

Implemented By:

Module Name	Source File	Last Revised
mod_sofia	sofia_glue.c	9591

local_media_port

Description needed! Please contribute one.

Usage:

Example needed! Please contribute one.

See also:

Implemented By:

Module Name	Source File	Last Revised
mod_sofia	sofia_glue.c	9591

local_video_ip

Description needed! Please contribute one.

Usage:

Example needed! Please contribute one.

See also:

Implemented By:

Module Name	Source File	Last Revised
mod_sofia	sofia_glue.c	5114

local_video_port

Description needed! Please contribute one.

Usage:

Example needed! Please contribute one.

See also:

Implemented By:

Module Name	Source File	Last Revised
mod_sofia	sofia_glue.c	5114

loopback_leg

Description needed! Please contribute one.

Usage:

Example needed! Please contribute one.

See also:

Implemented By:

Module Name	Source File	Last Revised
mod_loopback	mod_loopback.c	11536

max_forwards

Contains the current Max-Forwards value provided in the originating request. The Max-Forwards value is decremented by one for each hop in a SIP call, when the Max-Forwards value is depleted the receiving agent must not pass the call onwards.

The max_forwards variable may be set on an outbound channel, some providers such as BT IP Exchange insist on a minimum value to faithfully terminate the call, 50 in BT's case.

Beware! If the max-Forwards value is reset it can cause potential cyclic calls between two operators who loop calls back to each other in error. Use wisely.

Usage:

Bridge an incoming call to a provider requiring a minimum Max-Forwards value.

<action application="bridge" data="{max_forwards=65}sofia/gateway/ipexchange/442920180123" />

See also:

Implemented By:

Module Name	Source File	Last Revised
mod_sofia	mod_sofia.c	7347
mod_sofia	sofia.c	4819
mod_sofia	sofia_glue.c	6974
core	switch_core_session.c	6976
core	switch_ivr.c	8103

mduration

Description needed! Please contribute one.

Usage:

Example needed! Please contribute one.

See also:

Implemented By:

Module Name	Source File	Last Revised
core	switch_channel.c	6917

media_webrtc

Used to instruct FS to generate an INVITE for a WebRTC call. For example, in case you need to originate a WebRTC call but you are not calling a SIP UA that is registered with FS (if the UA is registered with FS, FS knows it should originate a WebRTC call).

Usage:

# In the origination vars (commands originate or bridge) add

memory_debug

Description needed! Please contribute one.

Usage:

Example needed! Please contribute one.

See also:

Implemented By:

Module Name	Source File	Last Revised
core	switch_core_session.c	7718

monitor_ring_dispo

Description needed! Please contribute one.

Usage:

Example needed! Please contribute one.

See also:

Implemented By:

Module Name	Source File	Last Revised
core	switch_ivr_originate.c	11437

myid

Description needed! Please contribute one.

Usage:

Example needed! Please contribute one.

See also:

Implemented By:

Module Name	Source File	Last Revised
mod_voicemail	mod_voicemail.c	8689

NDLB_support_asterisk_missing_srtp_auth

Description needed! Please contribute one.

Usage:

Example needed! Please contribute one.

See also:

Implemented By:

Module Name	Source File	Last Revised
mod_sofia	sofia_glue.c	8908

new_name

Description needed! Please contribute one.

Usage:

Example needed! Please contribute one.

See also:

Implemented By:

Module Name	Source File	Last Revised
mod_sofia	sofia.c	9703

nonexistantvar

Description goes here

Usage:

Example needed

See also:

Implemented By:

original_caller_id_name

Description needed! Please contribute one.

Usage:

Example needed! Please contribute one.

See also:

Implemented By:

Module Name	Source File	Last Revised
core	switch_ivr_bridge.c	7083

original_caller_id_number

Description needed! Please contribute one.

Usage:

Example needed! Please contribute one.

See also:

Implemented By:

Module Name	Source File	Last Revised
core	switch_ivr_bridge.c	7083

original_destination_number

Description needed! Please contribute one.

Usage:

Example needed! Please contribute one.

See also:

Implemented By:

Module Name	Source File	Last Revised
core	switch_ivr_bridge.c	7083

originate_continue_on_timeout

Controls wether or not a bridge should continue after timing out. Default value is false. This variable resets the timeout after each | default is to die on first timeout

Usage:

See also:

Implemented By:

Module Name	Source File	Last Revised
core	switch_originate.c	15121

originate_retries

Number of retries before giving up on originating a call.

Usage:

Example needed! Please contribute one.

See also:

Implemented By:

Module Name	Source File	Last Revised
core	switch_ivr_originate.c	4796

originate_retry_sleep_ms

This will set how long FreeSWITCH is going to wait between sending invite messages to the receiving gateway. Using the value of 500 FreeSWITCH will wait 500ms between sending invite messages to the called gateway.

Usage:

<action application="set" data="originate_retry_sleep_ms=500" />

See also:

Implemented By:

Module Name	Source File	Last Revised
core	switch_ivr_originate.c	4796

origination_timeout

Controls the amount of time that the bridge or originate operation will remain in "originate" state. In effect, this is a control on how long the bridge/originate will spend in total when calling multiple endpoints.

Usage:

<action application="bridge" data="{origination_timeout=10}[leg_timeout=5]sofia/internal/foo1@bar1\|[leg_timeout=5]sofia/internal/foo2@bar2" />

See also:

Channel Variables

Implemented By:

Module Name	Source File	Last Updated
Core	switch_ivr_originate.c

IMPORTANT NOTE

This variable does not exist in freeswitch source, use originate_timeout instead.

origination_uuid

You can specify the uuid of an originated call using origination_uuid. This way you can hang up the call before it is answered, since you know the uuid. Just remember you need to use the create_uuid command to generate the uuid as 2 calls with the same uuid == bad!

Usage:

originate

See also:

Implemented By:

Module Name	Source File	Last Revised
core	switch_core_session.c	10239

originator

Holds the UUID of the channel that originated the call. It's used to notify a parent channel that the state of it's child has changed, hence interrupting any blocking reads on the parent. It's automatically set and read by FreeSWITCH internals, usually the user won't want to set it.

Usage:

Example needed! Please contribute one.

See also:

Implemented By:

Module Name	Source File	Last Revised
core	switch_channel.c	4795
core	switch_core_session.c	4798

originator_video_codec

Description needed! Please contribute one.

Usage:

Example needed! Please contribute one.

See also:

Implemented By:

Module Name	Source File	Last Revised
core	switch_core_session.c	11731

other_loopback_leg_uuid

Description needed! Please contribute one.

Usage:

Example needed! Please contribute one.

See also:

Implemented By:

Module Name	Source File	Last Revised
mod_loopback	mod_loopback.c	12905

pa_hold_file

Description needed! Please contribute one.

Usage:

Example needed! Please contribute one.

See also:

Implemented By:

Module Name	Source File	Last Revised
mod_alsa	mod_alsa.c	5023
mod_portaudio	mod_portaudio.c	4795

pa_ring_file

Description needed! Please contribute one.

Usage:

Example needed! Please contribute one.

See also:

Implemented By:

Module Name	Source File	Last Revised
mod_alsa	mod_alsa.c	5023
mod_portaudio	mod_portaudio.c	4795

playback_terminators

Allows you to set which DTMF tones, if pressed during the playback of a file, will stop it. The default terminator is *. Keyword 'none' disables on-key termination. keyword 'any' will terminate playback when any key (1234567890*#) is pressed.

Usage:

<action application="set" data="playback_terminators=#*" />

Note: Git Committed 38b3f43d:

add prefix chars to playback_terminators + means include the term in the string and x means include the char and return SWITCH_STATUS_RESTART eg #+* only includes the * if you type it but not the #

See also:

Implemented By:

Module Name	Source File	Last Revised
mod_dptools	mod_dptools.c	5888

pound_replace

Description needed! Please contribute one.

Usage:

Example needed! Please contribute one.

See also:

Implemented By:

Module Name	Source File	Last Revised
core	switch_ivr_play_say.c	5908

You can customize the fields in FreeSWITCH's internal channels table by using the global variable presence_data_cols

Use : to separate the column names in presence_data_cols

You must define the fields in the channels table first:

ALTER TABLE channels ADD COLUMN accountcode VARCHAR(256)

<X-PRE-PROCESS cmd="set" data="presence_data_cols=accountcode:domain_name"/>

presence_id

Will instruct mod_sofia to invoke a PRESENCE_IN event that will assert the BLF lamp for the subscribed entity. This only lasts while the channel still exists so it should not be used for persistent status (i.e. a user that is DND, or an agent logged into a queue).

Usage:

See also:

Implemented By:

Module Name	Source File	Last Revised
mod_sofia	mod_sofia.c	6849
mod_sofia	sofia.c	6808
core	switch_channel.c	9363

profile_start_epoch

Description needed! Please contribute one.

Usage:

Example needed! Please contribute one.

See also:

Implemented By:

Module Name	Source File	Last Revised
core	switch_channel.c	6917

profile_start_stamp

Description needed! Please contribute one.

Usage:

Example needed! Please contribute one.

See also:

Implemented By:

Module Name	Source File	Last Revised
core	switch_channel.c	6917

profile_start_uepoch

Description needed! Please contribute one.

Usage:

Example needed! Please contribute one.

See also:

Implemented By:

Module Name	Source File	Last Revised
core	switch_channel.c	6917

progress_epoch

Description needed! Please contribute one.

Usage:

Example needed! Please contribute one.

See also:

Implemented By:

Module Name	Source File	Last Revised
core	switch_channel.c	11186

progress_media_epoch

Description needed! Please contribute one.

Usage:

Example needed! Please contribute one.

See also:

Implemented By:

Module Name	Source File	Last Revised
core	switch_channel.c	11186

progress_media_stamp

Description needed! Please contribute one.

Usage:

Example needed! Please contribute one.

See also:

Implemented By:

Module Name	Source File	Last Revised
core	switch_channel.c	8568

progress_media_uepoch

Description needed! Please contribute one.

Usage:

Example needed! Please contribute one.

See also:

Implemented By:

Module Name	Source File	Last Revised
core	switch_channel.c	11186

progress_mediamsec

Description needed! Please contribute one.

Usage:

Example needed! Please contribute one.

See also:

Implemented By:

Module Name	Source File	Last Revised
core	switch_channel.c	8576

progress_mediasec

Description needed! Please contribute one.

Usage:

Example needed! Please contribute one.

See also:

Implemented By:

Module Name	Source File	Last Revised
core	switch_channel.c	8576

progress_mediausec

Description needed! Please contribute one.

Usage:

Example needed! Please contribute one.

See also:

Implemented By:

Module Name	Source File	Last Revised
core	switch_channel.c	8576

progress_stamp

Description needed! Please contribute one.

Usage:

Example needed! Please contribute one.

See also:

Implemented By:

Module Name	Source File	Last Revised
core	switch_channel.c	8568

progress_timeout

This is the maximum time we will wait before we get media (wether its early media, ringing or answer)

Usage:

<action application="set" data="progress_timeout=20" />

See also:

Early Media

Implemented By:

Module Name	Source File	Last Revised
core	switch_ivr_originate.c	11286

progress_uepoch

Description needed! Please contribute one.

Usage:

Example needed! Please contribute one.

See also:

Implemented By:

Module Name	Source File	Last Revised
core	switch_channel.c	11186

progressmsec

Description needed! Please contribute one.

Usage:

Example needed! Please contribute one.

See also:

Implemented By:

Module Name	Source File	Last Revised
core	switch_channel.c	8576

progresssec

Description needed! Please contribute one.

Usage:

Example needed! Please contribute one.

See also:

Implemented By:

Module Name	Source File	Last Revised
core	switch_channel.c	8576

progressusec

Description needed! Please contribute one.

Usage:

Example needed! Please contribute one.

See also:

Implemented By:

Module Name	Source File	Last Revised
core	switch_channel.c	8576

proxy_media

Proxy Media mode puts FreeSWITCH in a "transparent proxy mode" for the RTP streams. The RTP streams still pass through FreeSWITCH (unlike bypass media mode).

NOTE: Late Negotiation (<param name="inbound-late-negotiation" value="true"/>) must be enabled in SIP profile for this to work in the dialplan.

Usage:

<action application="set" data="proxy_media=true" />

See also:

Proxy Media

Implemented By:

Module Name	Source File	Last Revised
mod_dptools	mod_dptools.c	7717

rdnis

Description needed! Please contribute one.

Usage:

Example needed! Please contribute one.

See also:

Implemented By:

Module Name	Source File	Last Revised
core	switch_ivr_bridge.c	9170

read_rate

Description needed! Please contribute one.

Usage:

Example needed! Please contribute one.

See also:

Implemented By:

Module Name	Source File	Last Revised
core	switch_core_codec.c	9040

read_result

Description needed! Please contribute one.

Usage:

Example needed! Please contribute one.

See also:

Implemented By:

Module Name	Source File	Last Revised
core	switch_ivr_play_say.c	9225

read_terminator_used

Contains the digit that the caller used to terminate a playback.

Usage:

Example needed! Please contribute one.

See also:

Implemented By:

Module Name	Source File	Last Revised
core	switch_ivr_play_say.c	12840

recovery_profile_name

Taken from [http://lists.freeswitch.org/pipermail/freeswitch-users/2012-December/090291.html](http://lists.freeswitch.org/pipermail/freeswitch-users/2012-December/090291.html)

Its used internally so the core recovery engine knows which profile name goes with the call. The profile name to the core is just an arbitrary sub category of the call where to mod_sofia it means the sip profile name.

You personally don't really have much use for it.

Usage:

Example needed! Please contribute one.

remote_media_ip

Description needed! Please contribute one.

Usage:

Example needed! Please contribute one.

See also:

Implemented By:

Module Name	Source File	Last Revised
mod_esf	mod_esf.c	5057
mod_sofia	mod_sofia.c	8689
mod_sofia	sofia_glue.c	6370

remote_media_port

Description needed! Please contribute one.

Usage:

Example needed! Please contribute one.

See also:

Implemented By:

Module Name	Source File	Last Revised
mod_esf	mod_esf.c	5057
mod_sofia	mod_sofia.c	8689
mod_sofia	sofia_glue.c	6370

remote_video_ip

Description needed! Please contribute one.

Usage:

Example needed! Please contribute one.

See also:

Implemented By:

Module Name	Source File	Last Revised
mod_sofia	sofia_glue.c	5114

remote_video_port

Description needed! Please contribute one.

Usage:

Example needed! Please contribute one.

See also:

Implemented By:

Module Name	Source File	Last Revised
mod_sofia	sofia_glue.c	5114

ringback

This will set the unanswered aka (early media) calls. The following example uses the US ring tone defined in ~/autoload_configs/switch.conf.xml

Usage:

<action application="set" data="ringback=$${us-ring}" />

Implemented By:

Module Name	Source File	Last Revised
core	switch_ivr_originate.c	9095

rss_alt_config

Description needed! Please contribute one.

Usage:

Example needed! Please contribute one.

See also:

Implemented By:

Module Name	Source File	Last Revised
mod_rss	mod_rss.c	6309

rtp_adv_audio_ip

This channel variable explicitly sets the ip address in SDP on a one-off basis. The "right" way to do this is ext-rtp-ip in the sofia profile.

Usage: <action application="set" data="rtp_adv_audio_ip=127.0.0.1"></action>

Implemented By:

Module Name	Source File	Last Revised
mod_sofia	sofia_glue.c	9088

rtp_autoflush

Skip timer sleeps when the socket has data ready.

Usage: <action application="set" data="rtp_autoflush=true"/>

See also:

Implemented By:

Module Name	Source File	Last Revised
mod_sofia	sofia_glue.c	12854

rtp_hold_timeout_sec

Description needed! Please contribute one.

Usage:

Example needed! Please contribute one.

See also:

Implemented By:

Module Name	Source File	Last Revised
mod_sofia	mod_sofia.c	10566
mod_sofia	sofia_glue.c	7309

rtp_manual_rtp_bugs

Needed!!!

Usage:

<action application="set" data="rtp_manual_rtp_bugs=cisco_skip_mark_bit_2833" />

Implemented By:

Module Name	Source File	Last Revised

rtp_rewrite_timestamps

Description needed.

Usage:

Implemented By:

Module Name	Source File	Last Revised
mod_sofia	sofia_glue.c	5501

rtp_stun_ping

Description needed! Please contribute one.

Usage:

Example needed! Please contribute one.

See also:

Implemented By:

Module Name	Source File	Last Revised
mod_sofia	sofia_glue.c	9088

rtp_timeout_sec

Description needed! Please contribute one.

Usage:

Example needed! Please contribute one.

See also:

Implemented By:

Module Name	Source File	Last Revised
mod_sofia	mod_sofia.c	10566
mod_sofia	sofia_glue.c	6093

rtp_timer_name

Description needed! Please contribute one.

Usage:

Example needed! Please contribute one.

See also:

Implemented By:

Module Name	Source File	Last Revised
mod_sofia	sofia_glue.c	12676

signal_bridge_to

Description needed! Please contribute one.

Usage:

Example needed! Please contribute one.

See also:

Implemented By:

Module Name	Source File	Last Revised
core	switch_ivr.c	5738
core	switch_ivr_bridge.c	12671

sip

Description needed! Please contribute one.

Usage:

Example needed! Please contribute one.

See also:

Implemented By:

Module Name	Source File	Last Revised
mod_sofia	mod_sofia.c	8689

sip_auth_method

The request method used to authenticate with.

Known values are;

Usage: TODO

See also: TODO

Implemented By: TODO

sip_auth_realm

This should contain the domain/realm of the authenticated user that has triggered the event, if applicable.

Usage:

originate {sip_auth_username=

<action application="bridge" data="{sip_auth_realm=${sip_auth_realm}}sofia/gateway/external/2000" />

See also: TODO

Implemented By: TODO

sip_auto_answer

Tells the SIP phone to auto-answer the call, if supported. Can be used for intercom support.

Usage: From dialplan/default.xml:

<action application="export" data="sip_auto_answer=true" />

See also:

Implemented By:

Module Name	Source File	Last Revised
mod_sofia	mod_sofia.c	11069
mod_sofia	sofia.c	5817

sip_call_id

Description needed! Please contribute one.

Usage:

Example needed! Please contribute one.

See also:

Implemented By:

Module Name	Source File	Last Revised
mod_sofia	sofia.c	6481

sip_codec_negotiation

Description needed! Please Expand.

sip_codec_negotiation is basically a channel variable equivalent of inbound-codec-negotiation.

sip_codec_negotiation accepts "generous", "scrooge" & "greedy" as values.

on Freeswitch versions before Feb 3 ( 74a0cfd1e101413a3941c41d04ee01d8df2ae418 ) sip_codec_negotiation will always be over-written by the value of sip-codec-negotiation ( in the sofia profile ).

sine this same revision the behaviour was change so that the channel variable will always over-ride the value set in the sip profile. This means you can change codec negotiation on a per call basis.

Usage:

| | | |

Implemented By:

Module Name	Source File	Last Revised
mod_sofia	sofia_glue.c	11881

sip_contact_host

Hostname part of the 'Contact' SIP header.

Usage:

For example, if your request header contains:

Contact: <sip:gw+test@server.example.com:5060;transport=udp;gw=test>

Then the value for this field would be:

server.example.com

See also: TODO

Implemented By: TODO

sip_contact_port

Port part of the 'Contact' SIP header.

Usage:

For example, if your request header contains:

Contact: <sip:gw+test@server.example.com:5060;transport=udp;gw=test>

Then the value for this field would be:

5060

See also: TODO

Implemented By: TODO

sip_contact_user

Username part from the Contact SIP header.

Usage:

For example, if your request header contains:

Contact: <sip:gw+test@server.example.com:5060;transport=udp;gw=test>

Then the value for this field would be:

gw+test

See also:

Implemented By:

Module Name	Source File	Last Revised
mod_sofia	sofia_glue.c	9190

sip_copy_custom_headers

To pass some custom X-headers from B-leg to A-leg, add {sip_copy_custom_headers=true} to the dial string of the B-leg.

Usage: Set it to false to disable sending custom X- headers to your SIP gateway provider

<action application="set" data="sip_copy_custom_headers=false" />

See also:

Implemented By:

Module Name	Source File	Last Revised
mod_sofia	mod_sofia.c	4350:
mod_sofia	sofia.c	4237

sip_destination_url

Description needed! Please contribute one.

Usage:

Example needed! Please contribute one.

See also:

Implemented By:

Module Name	Source File	Last Revised
mod_sofia	mod_sofia.c	6451

sip_enable_soa

For per call basis which can be set to false to disable SIP SOA from sofia and most likely result in untouched exchange of SDP.

Usage:

<action application="set" data="bypass_media=true" />

See also:

Channel_Variables#sip_enable_soa

Implemented By:

Module Name	Source File	Last Updated

sip_exclude_contact

Description needed! Please contribute one.

Usage:

Example needed! Please contribute one.

See also:

Implemented By:

Module Name	Source File	Last Revised
mod_sofia	mod_sofia.c	7894

sip_force_video_fmtp

Description needed! Please contribute one.

Usage:

Example needed! Please contribute one.

See also:

Implemented By:

Module Name	Source File	Last Revised
mod_fsv	mod_fsv.c	7564
mod_sofia	sofia_glue.c	7565

sip_from_comment

Description needed! Please contribute one.

Usage:

Example needed! Please contribute one.

See also:

Implemented By:

Module Name	Source File	Last Revised
mod_sofia	sofia.c	7190

sip_from_host

Hostname part of the 'From' SIP header.

Usage:

For example, if your request header contains:

From: <sip:1000@server.example.com>;tag=0e70ba56

Then the value for this field would be:

server.example.com

See also: TODO

Implemented By: TODO

sip_from_port

Port part of the 'From' SIP header.

Usage:

For example, if your request header contains:

From: <sip:1000@server.example.com:5661>;tag=0e70ba56

Then the value for this field would be:

5661

However, if your request header does not contain a port, then the value for this field would be your default SIP port (usually 5060) - for example;

From: <sip:1000@server.example.com>;tag=0e70ba56

See also: TODO

Implemented By: TODO

sip_from_uri

The SIP URI of the endpoint sending the INVITE.

Usage:

<action application="export" data="sip_from_uri=${sip_from_uri}" />

Some SIP providers insist on handling caller ID in a non-normal way. In most cases you can use effective_caller_id_name and effective_caller_id_number to set the caller ID on the outbound (B leg) of a bridged call. In some cases the SIP provider doesn't like that and may just show "anonymous" or some other default caller ID information.

The above example takes the From URI of an incoming SIP call and exports that value to the B leg. This is handy when you have a scenario like this:

Alice ==> FreeSWITCH bridge ==> Bob

You want Alice's caller ID information to be sent to Bob instead of FreeSWITCH's caller ID information. Usually this "just works" but if it doesn't you can try exporting sip_from_uri as above.

See also:

Implemented By:

Module Name	Source File	Last Revised
mod_sofia	sofia_glue.c	10397

sip_from_user

Username part of the 'From' SIP header.

Usage:

For example, if your request header contains:

From: <sip:1000@server.example.com>;tag=0e70ba56

Then the value for this field would be:

1000

See also: TODO

Implemented By: TODO

sip_from_user_stripped

This is the same as sip_from_user, but has the "+" sign stripped from it.

Code snippet taken from src/mod/endpoints/mod_sofia/sofia.c:

if (!zstr(from_user)) { if (from_user == '+') { switch_channel_set_variable(channel, "sip_from_user_stripped", (const char ) (from_user + 1)); } else { switch_channel_set_variable(channel, "sip_from_user_stripped", from_user); }}

Usage:

Example needed! Please contribute one.

See also:

Implemented By:

Module Name	Source File	Last Revised
mod_sofia	sofia.c	4819

sip_gateway

Description needed! Please contribute one.

Usage:

Example needed! Please contribute one.

See also:

Implemented By:

Module Name	Source File	Last Revised
mod_sofia	sofia.c	11468

sip_gateway_name

If your call is outbound, and Leg B is routed via a gateway, then this variable will contain the gateway name.

Usage:

Call from user on internal profile, routed out via gateway 'ntl'

sip_profile_name=gatewaysip_gateway_name=ntlsofia_profile_name=internal

See also:

sip_profile_name

Implemented By:

Module Name	Source File	Last Revised
mod_sofia	mod_sofia.c	6489

sip_h_Referred-By

This is the line:

Referred-By

In the SIP message. It may be set on a transfer to a number, which then bridges somewhere else.

Description needed! Please contribute one.

Usage: You can SET it by using:

<action application="set" data="sip_h_referred-by=000@domain.com" />

If you wish to unset it, you can use:

<action application="unset" data="sip_h_referred-by" />

See also:

Implemented By:

Module Name	Source File	Last Revised
mod_sofia	sofia.c	7001

sip_header_name

Description needed! Please contribute one.

Usage:

Example needed! Please contribute one.

See also:

Implemented By:

Module Name	Source File	Last Revised
mod_sofia	sofia.c	7190

sip_history_info

Description needed! Please contribute one.

Usage:

Example needed! Please contribute one.

See also:

Implemented By:

Module Name	Source File	Last Revised
mod_sofia	sofia.c	10718

sip_ignore_remote_cause

Description needed! Please contribute one.

Usage:

Example needed! Please contribute one.

See also:

Implemented By:

Module Name	Source File	Last Revised
mod_sofia	mod_sofia.c	10090

sip_invite_call_id

SIP Call-ID to use when originating a call

Usage:

<action application="set" data="sip_invite_call_id=mycustomcallid" />

See also:

Implemented By:

Module Name	Source File	Last Revised
mod_sofia	sofia.c

sip_local_url

Description needed! Please contribute one.

Usage:

Example needed! Please contribute one.

See also:

Implemented By:

Module Name	Source File	Last Revised
mod_sofia	mod_sofia.c	6451

sip_looped_call

True if the call has been authenticated via means other than an ACL and the current request IP/port matches the profile IP/port (see src/mod/endpoints/mod_sofia/sofia.c)

Usage: in dialplan/public.xml:

<condition field="${

Example:

| | | |

See also:

Dialplan_XML

Implemented By:

Module Name	Source File	Last Revised
mod_sofia	sofia.c	7078

sip_nat_detected

Description needed! Please contribute one.

Usage:

Example needed! Please contribute one.

See also:

Implemented By:

Module Name	Source File	Last Revised
mod_sofia	mod_sofia.c	8395
mod_sofia	sofia.c	8488
mod_sofia	sofia_glue.c	8395

sip_outgoing_call_id

Replaced by Variable_sip_invite_call_id

Usage:

Example needed! Please contribute one.

See also:

Implemented By:

Module Name	Source File	Last Revised
mod_sofia	sofia_glue.c	10416

sip_p_rtp_stat

Description needed! Please contribute one.

Usage:

Example needed! Please contribute one.

See also:

Implemented By:

Module Name	Source File	Last Revised
mod_sofia	sofia.c	8821

sip_profile

Name of the SIP profile which the request was received on.

Usage:

Example needed! Please contribute one.

See also:

Implemented By:

Module Name	Source File	Last Revised
mod_dialplan_asterisk	mod_dialplan_asterisk.c	6214

sip_profile_name

If your call is outbound, then this variable will contain the profile name used for the outbound channel (Leg B)

If the outbound channel is a gateway, then this variable will be set to 'gateway', and you'd need to look at 'sip_gateway_name' to get the name of the gateway.

If the outbound channel is another profile, then this variable will be set to the name of that profile.

This should NOT be confused with sofia_profile_name, which is the name of the profile for Leg A.

Usage:

Call from user on internal profile, routed out via gateway 'ntl'

sip_profile_name=gatewaysip_gateway_name=ntlsofia_profile_name=internal

Call from user on internal profile, routed out via internal profile

sip_profile_name=internalsip_gateway_name=sofia_profile_name=internal

Call from user on external profile, routed out via internal profile

sip_profile_name=internalsip_gateway_name=sofia_profile_name=external

See also:

sip_gateway_name

Implemented By:

TODO

sip_received_ip

Description needed! Please contribute one.

Usage:

Example needed! Please contribute one.

See also:

Implemented By:

Module Name	Source File	Last Revised
mod_sofia	sofia.c	8395

sip_received_port

Description needed! Please contribute one.

Usage:

Example needed! Please contribute one.

See also:

Implemented By:

Module Name	Source File	Last Revised
mod_sofia	sofia.c	8395

sip_refer_reply

Description needed! Please contribute one.

Usage:

Example needed! Please contribute one.

See also:

Implemented By:

Module Name	Source File	Last Revised
mod_sofia	mod_sofia.c	10471
mod_sofia	sofia.c	10471

sip_referred_by_cid

Description needed! Please contribute one.

Usage:

Example needed! Please contribute one.

See also:

Implemented By:

Module Name	Source File	Last Revised
mod_sofia	sofia.c	7190

sip_referred_by_user_stripped

Description needed! Please contribute one.

Usage:

Example needed! Please contribute one.

See also:

Implemented By:

Module Name	Source File	Last Revised
mod_sofia	sofia.c	7190

sip_reply_host

Description needed! Please contribute one.

Usage:

Example needed! Please contribute one.

See also:

Implemented By:

Module Name	Source File	Last Revised
mod_sofia	sofia.c	10573

sip_request_host

Hostname part of the SIP header.

Usage:

For example, if your request header starts with:

REGISTER sip:172.16.44.8 SIP/2.0

Then the value of this field would be:

172.16.44.8

See also: TODO

Implemented By: TODO

sip_request_port

Port part of the SIP header.

Usage:

For example, if your request header starts with:

REGISTER sip:172.16.44.8:5061 SIP/2.0

Then the value of this field would be:

5061

However, if your request header does not contain a port, then the value for this field would be your default SIP port (usually 5060) - for example;

REGISTER sip:172.16.44.8 SIP/2.0

See also: TODO

Implemented By: TODO

sip_require_timer

FS requires timer by default on session refresh unless its t.38 re-invite. To disable the require timer on session refresh, set it to false.

Usage:

<action application="set" data="sip_require_timer=false" />

See also:

Implemented By:

Module Name	Source File	Last Revised

sip_route_uri

This sets where the INVITE packet should actually be sent to.

This should normally be needed in situations where the TO field contains a hostname which points back at itself, and would cause an endless loop if this variable is not set.

Usage:

Very simple usage, set during bridge immediately before the endpoint to bridge to.

bridge {sip_route_uri=sip

A real-life example where this was needed. It changes the Request-URI and sends the INVITE packet to the correct destination by looking up the contact details for the registered endpoint. This is used to send an incoming call to a registered endpoint (PBX in this case) but set a Request-URI so the call can be routed by the receiving party. Without the sip_route_uri variable set, the call would loop back to FreeSWITCH endlessly until the originating party cancels the call.

The incoming number routes to another extension, and exports the original dialled DDI (as dialled_ddi) so this can be used with this bridge command.

bridge {sip_route_uri=sip

See also:

Variable sip invite req uri

[ Bug Report FS-5349] (which isn't a bug as such, but gives solution to a scenario where this variable is required)

Implemented By:

Module Name	Source File	Last Revised
mod_sofia	sofia_glue.c	12408

sip_rtp_rxstat

Description needed! Please contribute one.

Usage:

Example needed! Please contribute one.

See also:

Implemented By:

Module Name	Source File	Last Revised
mod_sofia	sofia.c	7427

sip_rtp_txstat

Description needed! Please contribute one.

Usage:

Example needed! Please contribute one.

See also:

Implemented By:

Module Name	Source File	Last Revised
mod_sofia	sofia.c	7427

sip_sticky_contact

Description needed! Please contribute one.

Usage:

Example needed! Please contribute one.

See also:

Implemented By:

Module Name	Source File	Last Revised
mod_sofia	mod_sofia.c	8391
mod_sofia	sofia_glue.c	8888

sip_subject

Description needed! Please contribute one.

Usage:

Example needed! Please contribute one.

See also:

Implemented By:

Module Name	Source File	Last Revised
mod_sofia	sofia.c	5089

sip_term_cause

Description needed! Please contribute one.

Usage:

Example needed! Please contribute one.

See also:

Implemented By:

Module Name	Source File	Last Revised
mod_sofia	sofia.c	12676

sip_term_status

Description needed! Please contribute one.

Usage:

Example needed! Please contribute one.

See also:

Implemented By:

Module Name	Source File	Last Revised
mod_sofia	sofia.c	12676

sip_to_comment

Description needed! Please contribute one.

Usage:

Example needed! Please contribute one.

See also:

Implemented By:

Module Name	Source File	Last Revised
mod_sofia	sofia.c	7190

sip_to_host

Hostname part of the 'To' SIP header.

Usage:

For example, if your request header contains:

To: <sip:1000@server.example.com>;tag=0e70ba56

Then the value for this field would be:

server.example.com

See also:

Implemented By:

Module Name	Source File	Last Revised
mod_sofia	mod_sofia.c	7067
mod_sofia	sofia.c	6874

sip_to_port

Port part of the 'To' SIP header.

Usage:

For example, if your request header contains:

To: <sip:1000@server.example.com:5661>;tag=0e70ba56

Then the value for this field would be:

5661

However, if your request header does not contain a port, then the value for this field would be your default SIP port (usually 5060) - for example;

To: <sip:1000@server.example.com>;tag=0e70ba56

See also: TODO

Implemented By: TODO

sip_to_uri

Description needed! Please contribute one.

Usage:

Example needed! Please contribute one.

See also:

Implemented By:

Module Name	Source File	Last Revised
mod_sofia	mod_sofia.c	7347
mod_sofia	sofia.c	6874

sip_to_user

Username part of the 'To' SIP header.

Usage:

To set manually, use:

<action application="set" data="sip_to_user=whatevah" />

For example, if your request header contains:

To: <sip:1000@server.example.com>;tag=0e70ba56

Then the value for this field would be:

1000

See also:

Implemented By:

Module Name	Source File	Last Revised
mod_sofia	sofia.c	6874

sip_transport

Description needed! Please contribute one.

Usage:

Example needed! Please contribute one.

See also:

Implemented By:

Module Name	Source File	Last Revised
mod_sofia	sofia_glue.c	7268

sip_use_gateway

Description needed! Please contribute one.

Usage:

Example needed! Please contribute one.

See also:

Implemented By:

Module Name	Source File	Last Revised
mod_sofia	sofia.c	10222
mod_sofia	sofia_reg.c	9068

sip_user_agent

User agent part of the SIP header.

Usage: For example, if your request header contains:

User-Agent: FreeSWITCH-mod_sofia/1.2.0-rc2+git~20120713T162602Z~0afd7318bd+unclean~20120713T184029Z

Then the value of this field would be:

FreeSWITCH-mod_sofia/1.2.0-rc2+git~20120713T162602Z~0afd7318bd+unclean~20120713T184029Z

See also:

Implemented By:

Module Name	Source File	Last Revised
mod_sofia	sofia.c	9170

sip_via_host

Description needed! Please contribute one.

Usage:

Example needed! Please contribute one.

See also:

Implemented By:

Module Name	Source File	Last Revised
mod_sofia	sofia.c	4819

sip_via_port

Description needed! Please contribute one.

Usage:

Example needed! Please contribute one.

See also:

Implemented By:

Module Name	Source File	Last Revised
mod_sofia	sofia.c	4819

sip_via_protocol

Description needed! Please contribute one.

Usage:

Example needed! Please contribute one.

See also:

Implemented By:

Module Name	Source File	Last Revised
mod_sofia	sofia.c	9923

sip_via_rport

Description needed! Please contribute one.

Usage:

Example needed! Please contribute one.

See also:

Implemented By:

Module Name	Source File	Last Revised
mod_sofia	sofia.c	4819

sip_video_fmtp

Description needed! Please contribute one.

Usage:

Example needed! Please contribute one.

See also:

Implemented By:

Module Name	Source File	Last Revised
mod_sofia	sofia_glue.c	7546

sip_video_pt

Description needed! Please contribute one.

Usage:

Example needed! Please contribute one.

See also:

Implemented By:

Module Name	Source File	Last Revised
mod_sofia	sofia_glue.c	7582

socket_host

Description needed! Please contribute one.

Usage:

Example needed! Please contribute one.

See also:

Implemented By:

Module Name	Source File	Last Revised
mod_event_socket	mod_event_socket.c	7766

socket_path

Description needed! Please contribute one.

Usage:

Example needed! Please contribute one.

See also:

Implemented By:

Module Name	Source File	Last Revised
mod_event_socket	mod_event_socket.c	7749

SOFIA_CRYPTO_MANDATORY_VARIABLE

Description needed! Please contribute one.

Usage:

Example needed! Please contribute one.

See also:

Implemented By:

Module Name	Source File	Last Revised
mod_sofia	sofia_glue.c	7746

SOFIA_HAS_CRYPTO_VARIABLE

Description needed! Please contribute one.

Usage:

Example needed! Please contribute one.

See also:

Implemented By:

Module Name	Source File	Last Revised
mod_sofia	sofia_glue.c	7244

sofia_profile_domain_name

Description needed! Please contribute one.

Usage:

Example needed! Please contribute one.

See also:

Implemented By:

Module Name	Source File	Last Revised
mod_sofia	sofia.c	7313

sofia_profile_name

The name of the profile of which the call originated (Leg A).

This should NOT be confused with sip_profile_name.

Usage:

Call from user on external profile

sofia_profile_name=external

Call from user on internal profile

sofia_profile_name=internal

See also:

Implemented By:

Module Name	Source File	Last Revised
mod_sofia	sofia.c	5739
mod_sofia	sofia_glue.c	5739

sofia_record_file

Description needed! Please contribute one.

Usage:

Example needed! Please contribute one.

See also:

Implemented By:

Module Name	Source File	Last Revised
mod_sofia	sofia.c	6952

SOFIA_REFER_TO_VARIABLE

Description needed! Please contribute one.

Usage:

Example needed! Please contribute one.

See also:

Implemented By:

Module Name	Source File	Last Revised
mod_sofia	sofia.c	7008

SOFIA_REPLACES_HEADER

Description needed! Please contribute one.

Usage:

Example needed! Please contribute one.

See also:

Implemented By:

Module Name	Source File	Last Revised
mod_sofia	sofia.c	9530
mod_sofia	sofia_glue.c	6120

SOFIA_SECURE_MEDIA_CONFIRMED_VARIABLE

Description needed! Please contribute one.

Usage:

Example needed! Please contribute one.

See also:

Implemented By:

Module Name	Source File	Last Revised
mod_sofia	sofia_glue.c	7244

SOFIA_SECURE_MEDIA_VARIABLE

Description needed! Please contribute one.

Usage:

Example needed! Please contribute one.

See also:

Implemented By:

Module Name	Source File	Last Revised
mod_sofia	mod_sofia.c	10228
mod_sofia	sofia_glue.c	7779

SOFIA_SESSION_TIMEOUT

Description needed! Please contribute one.

Usage:

Example needed! Please contribute one.

See also:

Implemented By:

Module Name	Source File	Last Revised
mod_sofia	mod_sofia.c	6347
mod_sofia	sofia_glue.c	6003

sound_prefix

Directory prefix where the sounds lives.

Usage:

Example needed! Please contribute one.

See also:

Implemented By:

Module Name	Source File	Last Revised
core	switch_ivr_play_say.c	11108

star_replace

Description needed! Please contribute one.

Usage:

Example needed! Please contribute one.

See also:

Implemented By:

Module Name	Source File	Last Revised
core	switch_ivr_play_say.c	5908

start_epoch

Description needed! Please contribute one.

Usage:

Example needed! Please contribute one.

See also:

Implemented By:

Module Name	Source File	Last Revised
core	switch_channel.c	6917

start_stamp

Description needed! Please contribute one.

Usage:

Example needed! Please contribute one.

See also:

Implemented By:

Module Name	Source File	Last Revised
core	switch_channel.c	6917

start_uepoch

Description needed! Please contribute one.

Usage:

Example needed! Please contribute one.

See also:

Implemented By:

Module Name	Source File	Last Revised
core	switch_channel.c	6917

stream_prebuffer

Description needed! Please contribute one.

Usage:

Example needed! Please contribute one.

See also:

Implemented By:

Module Name	Source File	Last Revised
mod_spidermonkey	mod_spidermonkey.c	8689
core	switch_cpp.cpp	5374
core	switch_ivr_play_say.c	5302

supress_cng

Description needed! Please contribute one.

Usage:

Example needed! Please contribute one.

See also:

Implemented By:

Module Name	Source File	Last Revised
mod_sofia	sofia_glue.c	9648

SWITCH_PLAYBACK_TERMINATOR_USED

Description needed! Please contribute one.

Usage:

Example needed! Please contribute one.

See also:

Implemented By:

Module Name	Source File	Last Revised
mod_dptools	mod_dptools.c	10489

SWITCH_UUID_BRIDGE

Description needed! Please contribute one.

Usage:

Example needed! Please contribute one.

See also:

Implemented By:

Module Name	Source File	Last Revised
core	switch_ivr_bridge.c	8686

sip_auto_simplify

If FreeSWITCH detects that it can remove itself from the SIP and RTP path (such as the haripin scenario) then it will issue REINVITEs the necessary endpoints.

Usage:

<action application="set" data="timer_name=soft" />

See also: uuid_simplify

tone_detect_expires

Description needed! Please contribute one.

Usage:

Example needed! Please contribute one.

See also:

Implemented By:

Module Name	Source File	Last Revised
core	switch_ivr_async.c	11309

tone_detect_sleep

Description needed! Please contribute one.

Usage:

Example needed! Please contribute one.

See also:

Implemented By:

Module Name	Source File	Last Revised
core	switch_ivr_async.c	11309

transfer_fallback_extension

It's an extension the channel falls back to on failed transfer. Set it before the transfer. For example, if you transfer to some invalid or unavailable ext, it will then "fall back" to the ext set in the var.

Usage:

Example needed! Please contribute one.

See also:

Implemented By:

Module Name	Source File	Last Revised
mod_sofia	sofia.c	6908

transfer_history

Description goes here

Usage:

Example needed.

See also:

Implemented By:

Module Name	Source File	Last Revised

		?

transfer_ringback

This will set the ring tone for answered calls. This is any call that has been setup. One example would be the tone to play during transfer. The following example uses the French ringtone defined in ~/autoload_configs/switch.conf.xml

Usage:

<action application="set" data="transfer_ringback=$${fr-ring}" />

See also:

ringback

Implemented By:

Module Name	Source File	Last Revised
core	switch_ivr_originate.c	9095

transfer_source

Description goes here

Usage:

Example needed.

See also:

Implemented By:

Module Name	Source File	Last Revised

		?

tts_engine

Description needed! Please contribute one.

Usage:

Example needed! Please contribute one.

See also:

Implemented By:

Module Name	Source File	Last Revised
mod_dptools	mod_dptools.c	5876
core	switch_ivr_menu.c	12297
core	switch_ivr_play_say.c	8686

tts_voice

Description needed! Please contribute one.

Usage:

Example needed! Please contribute one.

See also:

Implemented By:

Module Name	Source File	Last Revised
mod_dptools	mod_dptools.c	5876
core	switch_ivr_menu.c	12297
core	switch_ivr_play_say.c	8686

uduration

Description needed! Please contribute one.

Usage:

Example needed! Please contribute one.

See also:

Implemented By:

Module Name	Source File	Last Revised
core	switch_channel.c	6917

UNIQUEID

Description needed! Please contribute one.

Usage:

Example needed! Please contribute one.

See also:

Implemented By:

Module Name	Source File	Last Revised
mod_dialplan_asterisk	mod_dialplan_asterisk.c	6205

user_context

Description needed! Please contribute one.

Usage:

Example needed! Please contribute one.

See also:

Implemented By:

Module Name	Source File	Last Revised
mod_dptools	mod_dptools.c	6641
mod_sofia	sofia.c	6641

user_name

Description needed! Please contribute one.

Usage:

Example needed! Please contribute one.

See also:

Implemented By:

Module Name	Source File	Last Revised
core	switch_ivr.c	9056

verbose_presence

Description needed! Please contribute one.

Usage:

Example needed! Please contribute one.

See also:

Implemented By:

Module Name	Source File	Last Revised
core	switch_core_session.c	9363

video_possible

Description needed! Please contribute one.

Usage:

Example needed! Please contribute one.

See also:

Implemented By:

Module Name	Source File	Last Revised
mod_sofia	sofia_glue.c	8542

video_read_codec

Description needed! Please contribute one.

Usage:

Example needed! Please contribute one.

See also:

Implemented By:

Module Name	Source File	Last Revised
core	switch_core_codec.c	7565

video_read_rate

Description needed! Please contribute one.

Usage:

Example needed! Please contribute one.

See also:

Implemented By:

Module Name	Source File	Last Revised
core	switch_core_codec.c	7565

video_write_codec

Description needed! Please contribute one.

Usage:

Example needed! Please contribute one.

See also:

Implemented By:

Module Name	Source File	Last Revised
core	switch_core_codec.c	7565

video_write_rate

Description needed! Please contribute one.

Usage:

Example needed! Please contribute one.

See also:

Implemented By:

Module Name	Source File	Last Revised
core	switch_core_codec.c	7565

vm_cc

Description needed! Please contribute one.

Usage:

Example needed! Please contribute one.

See also: Mod_voicemail#vm_cc

Implemented By:

Module Name	Source File	Last Revised
mod_voicemail	mod_voicemail.c	12850

vm_message_ext

Determines in which format the voicemail message is saved.

This variable is set in the dial plan.

Valid values are (not exhaustive):

Usage example:

See also:

vm-message-ext

Implemented By:

Module Name	Source File	Last Revised
mod_voicemail	mod_voicemail.c	9098

vmd_detect

Description needed! Please contribute one.

Usage:

Example needed! Please contribute one.

See also:

Implemented By:

Module Name	Source File	Last Revised
mod_vmd	mod_vmd.c	10643

vname

Description needed! Please contribute one.

Usage:

Example needed! Please contribute one.

See also:

Implemented By:

Module Name	Source File	Last Revised
core	switch_channel.c	8776

voicemail_account

Description needed! Please contribute one.

Usage:

Example needed! Please contribute one.

See also:

Implemented By:

Module Name	Source File	Last Revised
mod_voicemail	mod_voicemail.c	6984

voicemail_alternate_greet_id

Overrides the ID the voicemail application reads back. I.e. to say a phone number instead of the user ID.

Usage:

<user id="johnsmith" number-alias="1000">

See also:

Implemented By:

Module Name	Source File	Last Revised
mod_voicemail	mod_voicemail.c	10594

voicemail_authorized

Description needed! Please contribute one.

Usage:

Example needed! Please contribute one.

See also:

Implemented By:

Module Name	Source File	Last Revised
mod_voicemail	mod_voicemail.c	6804

voicemail_caller_id_name

Description needed! Please contribute one.

Usage:

Example needed! Please contribute one.

See also:

Implemented By:

Module Name	Source File	Last Revised
mod_voicemail	mod_voicemail.c	6984

voicemail_caller_id_number

Description needed! Please contribute one.

Usage:

Example needed! Please contribute one.

See also:

Implemented By:

Module Name	Source File	Last Revised
mod_voicemail	mod_voicemail.c	6984

voicemail_current_folder

Description needed! Please contribute one.

Usage:

Example needed! Please contribute one.

See also:

Implemented By:

Module Name	Source File	Last Revised
mod_voicemail	mod_voicemail.c	8689

voicemail_domain

Description needed! Please contribute one.

Usage: This variable will change the domain name in the sender email address, when mod_voicemail emails a voicemail message.

from conf/directory/default/1010.xml:

<include> <user id="1010" mailbox="1010"> <params> <param name="password" value="password" /> <param name="vm-password" value="1010" /> </params> <variables> <variable name="voicemail_domain" value="speakblast.com" /> </variables> </user></include>

See also:

Implemented By:

Module Name	Source File	Last Revised
mod_voicemail	mod_voicemail.c	6984

voicemail_domain_name

Description needed! Please contribute one.

Usage:

Example needed! Please contribute one.

See also:

Implemented By:

Module Name	Source File	Last Revised
mod_voicemail	mod_voicemail.c	6984

voicemail_email

Description needed! Please contribute one.

Usage:

Example needed! Please contribute one.

See also:

Implemented By:

Module Name	Source File	Last Revised
mod_voicemail	mod_voicemail.c	6984

voicemail_file_path

Description needed! Please contribute one.

Usage:

Example needed! Please contribute one.

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Implemented By:

Module Name	Source File	Last Revised
mod_voicemail	mod_voicemail.c	6984

voicemail_greeting_number

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Usage:

Example needed! Please contribute one.

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Implemented By:

Module Name	Source File	Last Revised
mod_voicemail	mod_voicemail.c	6984

voicemail_greeting_path

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Usage:

Example needed! Please contribute one.

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Implemented By:

Module Name	Source File	Last Revised
mod_voicemail	mod_voicemail.c	10569

voicemail_id

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Usage:

Example needed! Please contribute one.

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Implemented By:

Module Name	Source File	Last Revised
mod_voicemail	mod_voicemail.c	6984

voicemail_message_len

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Usage:

Example needed! Please contribute one.

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Implemented By:

Module Name	Source File	Last Revised
mod_voicemail	mod_voicemail.c	6984

voicemail_priority

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Usage:

Example needed! Please contribute one.

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Implemented By:

Module Name	Source File	Last Revised
mod_voicemail	mod_voicemail.c	6984

voicemail_profile_name

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Usage:

Example needed! Please contribute one.

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Implemented By:

Module Name	Source File	Last Revised
mod_voicemail	mod_voicemail.c	6984

voicemail_read_flags

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Usage:

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Implemented By:

Module Name	Source File	Last Revised
mod_voicemail	mod_voicemail.c	6984

voicemail_time

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Usage:

Example needed! Please contribute one.

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Implemented By:

Module Name	Source File	Last Revised
mod_voicemail	mod_voicemail.c	6984

voicemail_total_new_messages

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Usage:

Example needed! Please contribute one.

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Module Name	Source File	Last Revised
mod_voicemail	mod_voicemail.c	6984

voicemail_total_saved_messages

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Usage:

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Module Name	Source File	Last Revised
mod_voicemail	mod_voicemail.c	6984

voicemail_urgent_new_messages

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Module Name	Source File	Last Revised
mod_voicemail	mod_voicemail.c	6984

voicemail_urgent_saved_messages

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Usage:

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Module Name	Source File	Last Revised
mod_voicemail	mod_voicemail.c	6984

waitmsec

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Usage:

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Module Name	Source File	Last Revised
core	switch_channel.c	3603

waitsec

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Usage:

Example needed! Please contribute one.

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Module Name	Source File	Last Revised
core	switch_channel.c	3582

waitusec

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Usage:

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Module Name	Source File	Last Revised
core	switch_channel.c	3624

write_rate

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Usage:

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Module Name	Source File	Last Revised
core	switch_core_codec.c	10045

xfer_uuids

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Usage:

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Module Name	Source File	Last Revised
mod_dptools	mod_dptools.c	7885

xml_cdr_base

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Usage:

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Module Name	Source File	Last Revised
mod_xml_cdr	mod_xml_cdr.c	6708

https://freeswitch.org/confluence/display/FREESWITCH/Variables

About
Variable References
absolute_codec_string
accountcode
acl_token
alert_info
answer_epoch
answer_stamp
answer_uepoch
answermsec
answersec
answerusec
api_after_bridge
api_hangup_hook
api_on_answer
api_on_media
api_on_startup
asr_intercept_dtmf
auto_answer_destination
auto_hunt
billmsec
billsec
billusec
bind_meta_key
bridge_answer_timeout
bridge_channel
bridge_early_media
bridge_epoch
bridge_filter_dtmf
bridge_generate_comfort_noise
bridge_hangup_cause
bridge_pre_execute_aleg_app
bridge_pre_execute_aleg_data
bridge_pre_execute_bleg_app
bridge_pre_execute_bleg_data
bridge_stamp
bridge_terminate_key
bridge_to
bridge_uepoch
bridge_uuid
bypass_media
bypass_media_after_bridge
cache_speech_handles
call_clientcode
call_timeout
caller_id
caller_id_name
caller_id_number
campon
campon_announce_sound
campon_fallback_context
campon_fallback_dialplan
campon_hold_music
campon_fallback_exten
campon_retries
campon_sleep
campon_stop_key
campon_timeout
cdr_csv_base
CHANNEL
channel_name
codec_string
conference_auto_outcall_announce
conference_auto_outcall_caller_id_name
conference_auto_outcall_caller_id_number
conference_auto_outcall_flags
conference_auto_outcall_maxwait
conference_auto_outcall_prefix
conference_auto_outcall_profile
conference_auto_outcall_timeout
conference_controls
conference_enforce_security
conference_enter_sound
conference_exit_sound
conference_last_matching_digits
conference_member_id
conference_moderator
conference_name
conference_recording
conference_uuid
continue_on_fail
copy_xml_cdr
created_time
current_application
current_application_data
current_application_response
default_language
destination_number
detect_speech_result
dialed_domain
dialed_group
dialed_user
digits_dialed
direction
disable_app_log
rtp_disable_hold
disable_q850_reason
disable_radius_start
disable_radius_stop
disable_rtp_auto_adjust
DISPLACE_HANGUP_ON_ERROR
dl_cid_msg
dl_from_host
dl_from_user
dl_host
dl_user
domain_name
drop_dtmf
dtmf_type
dtmf_verbose
duration
easy_acctcode
easy_destnum
easy_dialstring
easy_group
easy_limit
eavesdrop_annnounce_macro
eavesdrop_announce_id
eavesdrop_group
eavesdrop_indicate_failed
eavesdrop_indicate_idle
eavesdrop_indicate_new
eavesdrop_require_group
effective_caller_id_name
effective_caller_id_number
enable_file_write_buffering
enable_heartbeat_events
end_epoch
end_stamp
end_uepoch
ep_codec_prefer_sdp
ep_codec_string
exec_after_bridge_app
exec_after_bridge_arg
execute_on_answer
execute_on_fax_detect
execute_on_fax_failure
execute_on_fax_result
execute_on_fax_success
execute_on_media
execute_on_media_timeout
execute_on_originate
execute_on_pre_answer
execute_on_ring
execute_on_sip_extra_headers
execute_on_sip_reinvite
export_vars
EXTEN
fail_on_single_reject
failed_xml_cdr_prefix
failure_causes
fax_bad_rows
fax_disable_v17
fax_document_total_pages
fax_ecm_requested
fax_ecm_used
fax_end_page
fax_filename
fax_force_caller
fax_ident
fax_image_resolution
fax_image_size
fax_local_station_id
fax_prefix
fax_remote_station_id
fax_result_code
fax_result_text
fax_success
fax_transfer_rate
fax_use_ecm
fax_v17_disabled
fax_verbose
fifo_announce
fifo_bridged
fifo_caller_consumer_import
fifo_caller_exit_key
fifo_chime_freq
fifo_chime_list
fifo_consumer_caller_import
fifo_consumer_exit_key
fifo_consumer_id
fifo_consumer_wrapup_key
fifo_consumer_wrapup_sound
fifo_consumer_wrapup_time
fifo_hold_music
fifo_manual_bridged
fifo_member_wait
fifo_orbit_announce
fifo_orbit_context
fifo_orbit_exten
fifo_outbound_uuid
fifo_override_announce
fifo_pop_order
fifo_position
fifo_record_template
fifo_role
fifo_timestamp
force_transfer_context
force_transfer_dialplan
group_confirm_cancel_timeout
group_confirm_file
group_confirm_key
hangup_after_bridge
hangup_after_conference
hangup_cause
hangup_cause_q850
hold_events
hold_hangup_xfer_exten
hold_music
ignore_display_updates
ignore_early_media
import
inherit_codec
intercept_unanswered_only
intercept_unbridged_only
ivr_menu_status
ivr_menu_terminator
jitterbuffer_msec
last_bridge_hangup_cause
last_bridge_proto_specific_hangup_cause
last_matching_digits
last_transferred_conference
leg_delay_start
leg_progress_timeout
leg_timeout
loopback_bowout_on_execute
loopback_export
media_bug_answer_req
min_dup_digit_spacing_ms
monitor_early_media_fail
monitor_early_media_ring
monitor_early_media_ring_total
monitor_fail_dispo
no_throttle_limits
originate_delay_start
originate_disposition
originate_timeout
originating_leg_uuid
origination_callee_id_name
origination_callee_id_number
origination_caller_id_name
origination_caller_id_number
origination_channel_name
origination_privacy
originator_codec
outbound_redirect_fatal
park_after_bridge
park_timeout
pass_rfc2833
passthru_ptime_mismatch
process_cdr
profile_created_time
progress_time
proto_specific_hangup_cause
read_codec
RECORD_APPEND
RECORD_ARTIST
RECORD_BRIDGE_REQ
RECORD_COMMENT
RECORD_COPYRIGHT
RECORD_DATE
RECORD_DISCARDED
record_fill_cng
RECORD_HANGUP_ON_ERROR
RECORD_MIN_SEC
record_ms
record_sample_rate
RECORD_READ_ONLY
record_restart_limit_on_dtmf
record_sample_rate
record_samples
RECORD_SOFTWARE
RECORD_STEREO
RECORD_STEREO_SWAP
RECORD_TITLE
record_waste_resources
RECORD_WRITE_ONLY
recording_follow_transfer
rtcp_octet_count
rtcp_packet_count
rtp_disable_vad_in
rtp_disable_vad_out
rtp_enable_vad_in
rtp_enable_vad_out
sdp_m_per_ptime
sdp_secure_savp_only
send_silence_when_idle
session_in_hangup_hook
signal_bond
sip_acl_authed_by
sip_acl_token
sip_auth_password
sip_auth_username
sip_authorized
sip_auto_simplify
sip_callee_id_name
sip_callee_id_number
sip_callee_id_number
sip_cid_in_1xx
sip_cid_type
sip_copy_multipart
sip_enable_soa
sip_execute_on_image
sip_force_audio_fmtp
sip_from_display
sip_hangup_disposition
sip_ignore_183nosdp
sip_ignore_reinvites
sip_invite_contact_params
sip_invite_domain
sip_invite_from_params
ip_invite_params
sip_invite_req_uri
sip_invite_route_uri
sip_invite_tel_params
sip_invite_to_params
sip_invite_to_uri
rtp_jitter_buffer_during_bridge
sip_jitter_buffer_plc
sip_local_sdp_str
sip_mirror_remote_audio_codec_payload
sip_network_destination
sip_recovery_break_rfc
sip_renegotiate_codec_on_reinvite
sip_wait_for_aleg_ack
skeleton
skip_cdr_causes
spandsp_dtmf_rx_filter_dialtone
spandsp_dtmf_rx_reverse_twist
spandsp_dtmf_rx_threshold
spandsp_dtmf_rx_twist
suppress_cng
switch_m_sdp
switch_r_sdp
temp_hold_music
timer_name
timezone
tod_tz_offset
transfer_after_bridge
transfer_on_fail
transfer_to
uuid_bridge_continue_on_cancel
verbose_sdp
write_codec
data
endpoint_disposition
fax_document_transferred_pages
fax_header
fax_start_page
fifo_music
fifo_priority
fifo_serviced_by
fifo_serviced_uuid
fifo_status
fifo_strategy
fifo_target
fire_asr_events
flow_billmsec
flow_billsec
flow_billusec
funny_stun
group_context
has_t38
holding_uuid
id
inbound_dialplan
instant_ringback
is_outbound
language
last_app
last_arg
last_dtmf_duration
last_file_position
lcr_auto_route
lcr_route_count
left_hanging_extension
limit_id
limit_max
limit_rate
limit_realm
limit_usage
local_media_ip
local_media_port
local_video_ip
local_video_port
loopback_leg
max_forwards
mduration
media_webrtc
memory_debug
monitor_ring_dispo
myid
NDLB_support_asterisk_missing_srtp_auth
new_name
nonexistantvar
original_caller_id_name
original_caller_id_number
original_destination_number
originate_continue_on_timeout
originate_retries
originate_retry_sleep_ms
origination_timeout
IMPORTANT NOTE
origination_uuid
originator
originator_video_codec
other_loopback_leg_uuid
pa_hold_file
pa_ring_file
playback_terminators
pound_replace
presence_id
profile_start_epoch
profile_start_stamp
profile_start_uepoch
progress_epoch
progress_media_epoch
progress_media_stamp
progress_media_uepoch
progress_mediamsec
progress_mediasec
progress_mediausec
progress_stamp
progress_timeout
progress_uepoch
progressmsec
progresssec
progressusec
proxy_media
rdnis
read_rate
read_result
read_terminator_used
recovery_profile_name
remote_media_ip
remote_media_port
remote_video_ip
remote_video_port
ringback
rss_alt_config
rtp_adv_audio_ip
rtp_autoflush
rtp_hold_timeout_sec
rtp_manual_rtp_bugs
rtp_rewrite_timestamps
rtp_stun_ping
rtp_timeout_sec
rtp_timer_name
signal_bridge_to
sip
sip_auth_method
sip_auth_realm
sip_auto_answer
sip_call_id
sip_codec_negotiation
sip_contact_host
sip_contact_port
sip_contact_user
sip_copy_custom_headers
sip_destination_url
sip_enable_soa
sip_exclude_contact
sip_force_video_fmtp
sip_from_comment
sip_from_host
sip_from_port
sip_from_uri
sip_from_user
sip_from_user_stripped
sip_gateway
sip_gateway_name
sip_h_Referred-By
sip_header_name
sip_history_info
sip_ignore_remote_cause
sip_invite_call_id
sip_local_url
sip_looped_call
sip_nat_detected
sip_outgoing_call_id
sip_p_rtp_stat
sip_profile
sip_profile_name
sip_received_ip
sip_received_port
sip_refer_reply
sip_referred_by_cid
sip_referred_by_user_stripped
sip_reply_host
sip_request_host
sip_request_port
sip_require_timer
sip_route_uri
sip_rtp_rxstat
sip_rtp_txstat
sip_sticky_contact
sip_subject
sip_term_cause
sip_term_status
sip_to_comment
sip_to_host
sip_to_port
sip_to_uri
sip_to_user
sip_transport
sip_use_gateway
sip_user_agent
sip_via_host
sip_via_port
sip_via_protocol
sip_via_rport
sip_video_fmtp
sip_video_pt
socket_host
socket_path
SOFIA_CRYPTO_MANDATORY_VARIABLE
SOFIA_HAS_CRYPTO_VARIABLE
sofia_profile_domain_name
sofia_profile_name
sofia_record_file
SOFIA_REFER_TO_VARIABLE
SOFIA_REPLACES_HEADER
SOFIA_SECURE_MEDIA_CONFIRMED_VARIABLE
SOFIA_SECURE_MEDIA_VARIABLE
SOFIA_SESSION_TIMEOUT
sound_prefix
star_replace
start_epoch
start_stamp
start_uepoch
stream_prebuffer
supress_cng
SWITCH_PLAYBACK_TERMINATOR_USED
SWITCH_UUID_BRIDGE
sip_auto_simplify
tone_detect_expires
tone_detect_sleep
transfer_fallback_extension
transfer_history
transfer_ringback
transfer_source
tts_engine
tts_voice
uduration
UNIQUEID
user_context
user_name
verbose_presence
video_possible
video_read_codec
video_read_rate
video_write_codec
video_write_rate
vm_cc
vm_message_ext
vmd_detect
vname
voicemail_account
voicemail_alternate_greet_id
voicemail_authorized
voicemail_caller_id_name
voicemail_caller_id_number
voicemail_current_folder
voicemail_domain
voicemail_domain_name
voicemail_email
voicemail_file_path
voicemail_greeting_number
voicemail_greeting_path
voicemail_id
voicemail_message_len
voicemail_priority
voicemail_profile_name
voicemail_read_flags
voicemail_time
voicemail_total_new_messages
voicemail_total_saved_messages
voicemail_urgent_new_messages
voicemail_urgent_saved_messages
waitmsec
waitsec
waitusec
write_rate
xfer_uuids
xml_cdr_base

__Variables

About​

Variable References​

absolute_codec_string​

accountcode​

acl_token​

alert_info​

answer_epoch​

answer_stamp​

answer_uepoch​

answermsec​

answersec​

answerusec​

api_after_bridge​

api_hangup_hook​

api_on_answer​

api_on_media​

api_on_startup​

asr_intercept_dtmf​

auto_answer_destination​

auto_hunt​

billmsec​

billsec​

billusec​

bind_meta_key​

bridge_answer_timeout​

bridge_channel​

bridge_early_media​

bridge_epoch​

bridge_filter_dtmf​

bridge_generate_comfort_noise​

bridge_hangup_cause​

bridge_pre_execute_aleg_app​

bridge_pre_execute_aleg_data​

bridge_pre_execute_bleg_app​

bridge_pre_execute_bleg_data​

bridge_stamp​

bridge_terminate_key​

bridge_to​

bridge_uepoch​

bridge_uuid​

bypass_media​

bypass_media_after_bridge​

cache_speech_handles​

call_clientcode​

call_timeout​

caller_id​

caller_id_name​

caller_id_number​

campon​

campon_announce_sound​

campon_fallback_context​

campon_fallback_dialplan​

campon_hold_music​

campon_fallback_exten​

campon_retries​

campon_sleep​

campon_stop_key​

campon_timeout​

cdr_csv_base​

CHANNEL​

channel_name​

codec_string​

conference_auto_outcall_announce​

conference_auto_outcall_caller_id_name​

conference_auto_outcall_caller_id_number​

conference_auto_outcall_flags​

conference_auto_outcall_maxwait​

conference_auto_outcall_prefix​

conference_auto_outcall_profile​

conference_auto_outcall_timeout​

conference_controls​

conference_enforce_security​

conference_enter_sound​

conference_exit_sound​

conference_last_matching_digits​

conference_member_id​

conference_moderator​

conference_name​

conference_recording​

About

Variable References

absolute_codec_string

accountcode

acl_token

alert_info

answer_epoch

answer_stamp

answer_uepoch

answermsec

answersec

answerusec

api_after_bridge

api_hangup_hook

api_on_answer

api_on_media

api_on_startup

asr_intercept_dtmf

auto_answer_destination

auto_hunt

billmsec

billsec

billusec

bind_meta_key

bridge_answer_timeout

bridge_channel

bridge_early_media

bridge_epoch

bridge_filter_dtmf

bridge_generate_comfort_noise

bridge_hangup_cause

bridge_pre_execute_aleg_app

bridge_pre_execute_aleg_data

bridge_pre_execute_bleg_app

bridge_pre_execute_bleg_data

bridge_stamp

bridge_terminate_key

bridge_to

bridge_uepoch

bridge_uuid

bypass_media

bypass_media_after_bridge

cache_speech_handles

call_clientcode

call_timeout

caller_id

caller_id_name

caller_id_number

campon

campon_announce_sound

campon_fallback_context

campon_fallback_dialplan

campon_hold_music

campon_fallback_exten

campon_retries

campon_sleep

campon_stop_key

campon_timeout

cdr_csv_base

CHANNEL

channel_name

codec_string

conference_auto_outcall_announce

conference_auto_outcall_caller_id_name

conference_auto_outcall_caller_id_number

conference_auto_outcall_flags

conference_auto_outcall_maxwait

conference_auto_outcall_prefix

conference_auto_outcall_profile

conference_auto_outcall_timeout

conference_controls

conference_enforce_security

conference_enter_sound

conference_exit_sound

conference_last_matching_digits

conference_member_id

conference_moderator

conference_name

conference_recording

conference_uuid