absolute_codec_string

Usage

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

accountcode

Usage

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

acl_token

alert_info

Usage

Load an external WAV ringtone 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"/>

answer_epoch

answer_stamp

answer_uepoch

answermsec

answersec

answerusec

api_after_bridge

Usage

Paging to PA System via Portaudio (with chime before and after announcement):

<extension name="paging-with-chime">
	<condition field="destination_number" expression="^7243$">
		<action application="set" data="bridge_pre_execute_bleg_app=playback"/>
		<action application="set" data="bridge_pre_execute_bleg_data=/tmp/2_tone_chime.wav"/>
		<action application="set" data="api_after_bridge=luarun post-chime.lua"/>
		<action application="bridge" data="portaudio/auto_answer"/>
	</condition>
</extension>

post-chime.lua:

api = freesswitch.API()
api:execute("pa", "play /tmp/2_tone_chime.wav")

api_hangup_hook

See also:

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

Usage

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

api_on_answer

Usage

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

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

api_on_originate

api_on_post_originate

api_on_pre_answer

api_on_pre_originate

api_on_ring

api_on_sip_extra_headers

api_on_startup

Usage

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

api_on_tone_detect

asr_intercept_dtmf

auto_answer_destination

auto_hunt

Usage

In conf/vars.xml:

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

<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>

base_dir

This defines the base directory for most other directory settings

billmsec

billsec

billusec

bind_meta_key

Usage

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

bridge_answer_timeout

See also: Early Media

Usage

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

bridge_channel

bridge_early_media

boolean 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. 

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

• you are using multiple targets in originate or bridge (e.g.: originate {globalVar=global}[channelVar=channel1]sofia/gateway/GW1/123456|[channelVar=channel2]sofia/gateway/GW2/123456) where you need to receive network generated Call Progress Tones (e.g.: ringtones and network announcements). You can combine this with continue_on_fail so you can fine-tune fallback on specific release causes only.

bridge_epoch

bridge_filter_dtmf

Usage

In the dialplan:

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

Dialstring:

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

bridge_generate_comfort_noise

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"/>

bridge_hangup_cause

Usage

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

bridge_pre_execute_aleg_app

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

bridge_pre_execute_aleg_data

bridge_pre_execute_bleg_app

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.

Arguments can be passed to the app with bridge_pre_execute_bleg_data.

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"/>

bridge_pre_execute_bleg_data

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"/>

bridge_stamp

bridge_terminate_key

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 dialplan.

bridge_to

bridge_uepoch

bridge_uuid

bypass_keep_codec

bypass_media

Usage

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

bypass_media_after_bridge

cache_dir

cache_speech_handles

call_clientcode

call_timeout

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.

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.

Default Value: 60

Usage

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

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})}"/>

callee_id_name

callee_id_number

caller_id_name

caller_id_number

campon

Default: false

Usage

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

campon_announce_sound

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

campon_fallback_dialplan

campon_fallback_exten

campon_hold_music

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_retries

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

Default: 10

Usage

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

campon_stop_key

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

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

certs_dir

CHANNEL

channel_name

codec_string

Usage

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

conf_dir

conference_auto_outcall_announce

Usage

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

conference_auto_outcall_caller_id_name

Usage

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

conference_auto_outcall_caller_id_number

Usage

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

conference_auto_outcall_flags

Usage

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

conference_auto_outcall_maxwait

Usage

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

conference_auto_outcall_prefix

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>

conference_auto_outcall_profile

Usage

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

conference_auto_outcall_timeout

Usage

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

conference_controls

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"/>

conference_enforce_security

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)

conference_enter_sound

Usage

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

conference_exit_sound

Usage

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

conference_last_matching_digits

Usage

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

conference_member_id

conference_moderator

conference_recording

conference_uuid

continue_on_fail

Usage

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

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

With Q.850 cause codes:

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

copy_xml_cdr

Usage

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

core_uuid

created_time

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 profile_created_time and variable_progress_time.

current_application

current_application_data

current_application_response

data

db_dir

default_language

Usage

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

deny_refer_requests

destination_number

detect_speech_result

dialed_domain

dialed_group

dialed_user

direction

Usage

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

disable_app_log

disable_q850_reason

Usage

<action application="set" data="disable_q850_reason=true"/>

disable_radius_start

disable_radius_stop

disable_rtp_auto_adjust

Usage

Add {disable_rtp_auto_adjust=true} in your dial string.

DISPLACE_HANGUP_ON_ERROR

Usage

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

dl_cid_msg

dl_from_host

dl_from_user

dl_host

dl_user

domain_name

drop_dtmf

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

TODO RFC 2833 is obsoleted by RFC 4733.

Usage

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

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

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

drop_dtmf_masking_digits

Usage

Each DTMF tone that is entered into the a-leg will be dropped and a "d" sent to the b-leg.

<action application="set" data="drop_dtmf=true"/>
<action application="set" data="drop_dtmf_masking_digits=d"/>

To drop and replace DTMF on the b-leg use:

<action application="export" data="nolocal:drop_dtmf=true"/>
<action application="export" data="nolocal:drop_dtmf_masking_digits=d"/>

drop_dtmf_masking_file

Usage

Each DTMF tone that is entered into the a-leg will be dropped and the file dtmf_masking_tone played to the b-leg.

<action application="export" data="drop_dtmf=true"/>
<action application="export" data="drop_dtmf_masking_file=/etc/freeswitch/Sounds/dtmf_masking_tone"/>

To drop and replace DTMF on the b-leg use

<action application="export" data="nolocal:drop_dtmf=true"/>
<action application="export" data="nolocal:drop_dtmf_masking_file=/etc/freeswitch/Sounds/dtmf_masking_tone" />

dtmf_type

Usage

<action application="set" data="dtmf_type=info"/>

<action application="set" data="dtmf_type=rfc2833"/>

TODO RFC 2833 is obsoleted by RFC 4733.

<action application="set" data="dtmf_type=none"/>

dtmf_verbose

Usage

<action application="set" data="dtmf_verbose=true"/>
<action application="spandsp_start_dtmf" />

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_ani

effective_aniii

effective_caller_id_name

Usage

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

effective_caller_id_number

Usage

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

effective_sip_cid_in_1xx

Usage

<action application="set" data="sip_cid_in_1xx=false"/>

enable_file_write_buffering

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"/>

enable_heartbeat_events

end_epoch

end_stamp

end_uepoch

endpoint_disposition

ep_codec_prefer_sdp

ep_codec_string

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"/>

exec_after_bridge_app

Usage

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

exec_after_bridge_arg

Usage

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

execute_on_answer

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}"/>

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>

execute_on_fax_detect

execute_on_fax_failure

execute_on_fax_result

execute_on_fax_success

execute_on_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 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}"/>

To wait 30 seconds for an answer without ignoring early media:

originate sofia/gateway/my_gateway/5551212 885551212

<extension name="exe_on_ans">
  <condition field="destination_number" expression="^88(\d+)$">
    <action application="set" data="execute_on_media=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>

execute_on_media_timeout

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"/>

execute_on_originate

Execute syntax: <app> <arg>

Asynchronous syntax: <app>::<arg>

Executes code on successful origination.

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

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}"/>

To wait 30 seconds for an answer without ignoring early media:

originate sofia/gateway/my_gw/5551212 885551212

<extension name="exe_on_preans">
  <condition field="destination_number" expression="^88(\d+)$">
    <action application="set" data="execute_on_pre_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>

execute_on_ring

Usage

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

execute_on_sip_extra_headers

execute_on_sip_reinvite

export_vars

This is useful to transfer information from a-leg (INVITE) to the future b-leg of REFER.

EXTEN

fail_on_single_reject

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

<action application="set" data="fail_on_single_reject=true"/>
<action application="bridge" data="sofia/$${profile}/$${kitchen}%$${domain},sofia/$${profile}/$${dining}%$${domain}"/>
<action application="javascript" data="answermachine.js"/>

<action application="set" data="fail_on_single_reject=USER_BUSY"/>

<action application="set" data="fail_on_single_reject=!NORMAL_CIRCUIT_CONGESTION"/>

To use a list:

<action application="set" data="fail_on_single_reject=^^:CALL_REJECTED:NORMAL_CLEARING:USER_BUSY"/>

For negated list:

<action application="set" data="fail_on_single_reject=!^^:ALLOTTED_TIMEOUT:NETWORK_OUT_OF_ORDER"/>

For use with the OR operator, consider the following example. Here the dialplan sends a call to multiple gateways simultaneously. In the case of GATEWAY_DOWN or INVALID_GATEWAY the bridge should not fail (i.e. negated fail_on_single_reject) and the dialplan should continue (i.e. continue_on_fail):

<action application="set" data="continue_on_fail=GATEWAY_DOWN,INVALID_GATEWAY"/>
<action application="set" data="fail_on_single_reject=!^^:GATEWAY_DOWN:INVALID_GATEWAY"/>
<action application="set" data="hangup_after_bridge=true"/>
<action application="bridge" data="sofia/gateway/gw1/$1|sofia/gateway/gw2/$1|sofia/gateway/gw3/$1"/>

failed_xml_cdr_prefix

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

<action application="set" data="failed_xml_cdr_prefix=failinggw" />

failure_causes

Usage

<action application="set" data="failure_causes=USER_BUSY,NO_ANSWER"/>

With Q.850 cause codes:

<action application="set" data="failure_causes=487"/>

fax_bad_rows

fax_disable_v17

fax_document_total_pages

fax_document_transferred_pages

fax_ecm_requested

fax_ecm_used

fax_end_page

fax_filename

fax_force_caller

fax_header

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_start_page

fax_success

fax_transfer_rate

fax_use_ecm

fax_v17_disabled

fax_verbose

fifo_announce

fifo_bridged

fifo_caller_consumer_import

Usage

<action application="set" data="fifo_caller_consumer_import=var1,var2"/>

fifo_caller_exit_key

fifo_chime_freq

fifo_chime_list

fifo_consumer_caller_import

Usage

<action application="set" data="fifo_consumer_caller_import=var1,var2"/>

fifo_consumer_exit_key

fifo_consumer_id

fifo_consumer_wrapup_key

fifo_consumer_wrapup_sound

fifo_consumer_wrapup_time

fifo_hold_music

fifo_manual_bridge

fifo_member_wait

If set to nowait, the consumer's leg will hangup when the caller hangs up.

Usage

<action application="set" data="result=${fifo_member(add MyQueueName {fifo_member_wait=nowait}user/1001@VoiceNetwork.ca )"/>

fifo_music

fifo_orbit_announce

fifo_orbit_context

Usage

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

fifo_orbit_exten

Syntax: exten:timeout

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

Before executing the fifo application:

<action application="set" data="fifo_orbit_exten=MyFIFOVoicemail:60"/>

Now after 60 seconds of waiting in the FIFO, the caller will be transferred to MyFIFOVoicemail.

<action application="fifo" data="MyFIFO in"/>

fifo_outbound_announce

fifo_outbound_uuid

fifo_override_announce

fifo_pop_order

Usage

<action application="set" data="fifo_pop_order=1,2" />

fifo_position

fifo_priority

fifo_record_template

fifo_role

fifo_serviced_by

fifo_serviced_uuid

fifo_status

fifo_strategy

fifo_target

fifo_timestamp

See Also

• mod_dptools: detect_speech
• mod_dptools: play_and_detect_speech

flow_billmsec

flow_billsec

flow_billusec

force_transfer_context

Usage

<action application="bridge" data="{force_transfer_context=some_context}sofia/gateway/gw_name/12345"/>

force_transfer_dialplan

funny_stun

grammar_dir

group_confirm_cancel_timeout

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

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

However, if you do

then the timeout will only apply to phase 1. So, once phase 1 is crossed, leg_timeout counter stops.

group_confirm_file

Usage

<action application="set" data="group_confirm_file=/usr/local/freeswitch/sounds/take_call_question.wav" />

group_confirm_key

Usage

<action application="set" data="group_confirm_key=1" />

group_context

hangup_after_bridge

Usage

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

hangup_after_conference

Usage

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

<!-- all regular memebers who dial "3500" will continue to the echo action after the "endconf" member below leaves conference -->
<!-- note that "hangup_after_true" is depenent on "endconf", one members (or more) must have endconf flag -->
<!-- if two or more managers join, call, participants will be kicked after laster member with "endconf" flag leaves -->
<!-- When a person calls into "3500-admin" and leaves, the partipants who came in via "3500" will continue to thier own echo app -->



    <extension name="cdquality_conferences">
      <condition field="destination_number" expression="^(35\d{2})$">
        <action application="answer"/>
        <action application="set" data="hangup_after_conference=false"/>
        <action application="conference" data="$1-${domain_name}@video-mcu-stereo"/>
        <action application="set" data="echo_encode_video=true"/>
        <action application="echo"/>
      </condition>
    </extension>


    <extension name="conference manager">
      <condition field="destination_number" expression="^(35\d{2})-admin$">
        <action application="set" data="conference_member_flags=endconf"/>
        <action application="conference" data="$1-${domain_name}@video-mcu-stereo"/>
      </condition>
    </extension>

hangup_cause

Usage

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

hangup_cause_q850

Usage

<action application="log" data="1 A-leg hangup Q850 cause: ${hangup_cause_q850}"/>

has_t38

hold_hangup_xfer_exten

Usage

<action application="set" data="hold_hangup_xfer_exten=1000:XML:default"/>

hold_music

Usage

<action application="set" data="hold_music=/sounds/holdmusic.wav" />

<action application="set" data="hold_music=silence" />

You can also use phrase: to indicate to use a phrase instead of a specific file:

<action application="set" data="hold_music=phrase:my_hold_music" />

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:

<action application="set" data="hold_music=indicate_hold" />

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:

<action application="bridge_export" data="hold_music=$${sounds_dir}/music/company-a.mp3"/>

holding_uuid

hostname

htdocs_dir

id

ignore_display_updates

Usage

From dialplan/default.xml, to set on A-Leg:

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

To set on B-Leg:

<action application="bridge" data="{ignore_display_updates=true}sofia/gateway/provider/18005551212"/>

ignore_early_media

Usage

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

You may also 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):

<action application="bridge" data="{ignore_early_media=true}sofia/test-int/1001@somebox,sofia/test-int/1000@somehost"/>

Setting the value to ring_ready will work the same as ignore_early_media=true but also send a SIP 180 to the inbound leg when the first SIP 183 is caught.

<action application="set" data="ignore_early_media=ring_ready"/>

import

Usage

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

inbound_dialplan

inherit_codec

Usage

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

initial_callee_id_name

Usage

An example of setting this to the caller id name of the number being dialed:

<action application="set" data="initial_callee_id_name='${user_data(${dialed_extension}@${domain_name} var effective_caller_id_name)}'"/>

instant_ringback

Usage

this may cause undesirable effects, like ringing for 3 seconds that turns into a busy signal.

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

intercept_unanswered_only

Default: false

Usage

<action application="set" data="intercept_unanswered_only=true"/>
<action application="intercept" data="myUUID"/>

intercept_unbridged_only

Default: false

Usage

<action application="set" data="intercept_unbridged_only=true"/>
<action application="intercept" data="myUUID"/>

ip_invite_params

is_outbound

ivr_menu_status

• success
• failure
• timeout

ivr_menu_terminator

Usage

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

jitterbuffer_msec

Usage

<action application="set" data="jitterbuffer_msec=60:200:20"/>
<action application="answer"/>

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.

<action application="export" data="nolocal:jitterbuffer_msec=60"/>
<action application="bridge" data="sofia/default/888@conference.freeswitch.org"/>

You can also activate the jitter buffer in the bridge as follows:

<action application="bridge" data="{jitterbuffer_msec=60}sofia/gateway/$1@gateway.com"/>

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.

language

last_app

last_arg

last_bridge_hangup_cause

Usage

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

last_bridge_proto_specific_hangup_cause

last_bridge_to

last_dtmf_duration

last_file_position

last_transferred_conference

Usage

<action application="log" data="INFO Last conference this person visited was [${last_transferred_conference}]"/>

lcr_auto_route

lcr_route_count

left_hanging_extension

leg_delay_start

Usage

<action application="bridge" data="sofia/profile/dest1,[leg_delay_start=10]sofia/profile/dest2,[leg_delay_start=15]sofia/profile/dest3"/>

leg_progress_timeout

Usage

Make all bridged calls fail over to the next in 6 seconds:

<action application="export" data="leg_progress_timeout=6"/>

leg_timeout

You can also use leg_progress_timeout to specify the maximum time we will wait before we get media (whether it's 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

<action application="bridge" data="[leg_timeout=15]user/hastoanswerquickly/some.domain.com,[leg_timeout=60]user/hasaminutetoanswer@some.domain.com"/>

limit_id

limit_ignore_transfer

limit_max

limit_rate

limit_realm

limit_usage

string

This global variable is set dynamically by the FreeSWITCH core by asking the underlying OS for the IP address of the server on which FreeSWITCH is running. If the server has multiple IP addresses this will be set to the first one returned by the OS. In the default configuration there are many settings that use this global variable. 

You cannot override the value of this global variable, because it will be reset by the FreeSWITCH core every minute. If your server has multiple IP addresses and you want to force FreeSWITCH to use a particular IP address, you can create a global variable force_local_ip_v4 and set it to the IP address you prefer. You must also set the local_ip_v4 variable to this IP address, you can do it by simply setting local_ip_v4 to use the value of force_local_ip_v4 

<X-PRE-PROCESS cmd="set" data="force_local_ip_v4=1.1.1.1"/>
<X-PRE-PROCESS cmd="set" data="local_ip_v4=$${force_local_ip_v4 }"/>

string

This global variable is set dynamically by the FreeSWITCH core by asking the underlying OS for the IPv6 address of the server on which FreeSWITCH is running (if it supports IPv6). If the server has multiple IPv6 addresses this will be set to the first one returned by the OS. In the default configuration this variable is used for setting up IPv6 SIP and Verto Profiles.

You cannot override the value of this global variable, because it will be reset by the FreeSWITCH core every minute. If your server has multiple IPv6 addresses and you want to force FreeSWITCH to use a particular IPv6 address, you can create a global variable force_local_ip_v6 and set it to the IP address you prefer. You must also set the local_ip_v6 variable to this IP address, you can do it by simply setting local_ip_v6 to use the value of force_local_ip_v6.

<X-PRE-PROCESS cmd="set" data="force_local_ip_v4=1.1.1.1"/>
<X-PRE-PROCESS cmd="set" data="local_ip_v4=$${force_local_ip_v4 }"/>

local_media_ip

local_media_port

local_video_ip

local_video_port

log_dir

loopback_bowout_on_execute

loopback_bowout_on_execute is only useful to single leg calls. You never have to set it: loopback_bowout is true by default. It tries to remove loopback by doing uuid_bridge.

When setting loopback_bowout=false, a bridged loopback call results 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 the call setup phase even with loopback_bowout=true there will always be 4 legs, however. loopback-a and loopback-b will be destroyed when a-leg and b-leg are successfully bridged.

Usage

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

loopback_export

loopback_leg

max_forwards

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.

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"/>

mduration

media_bug_answer_req

Usage

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

media_mix_inbound_outbound_codecs

FreeSWITCH used to take the codecs from the inbound A-leg and offer them out to the B-leg mixed with the configured list of outbound codecs. With large lists of codecs this can exceed the MTU of the UDP message and cause call setup failures.

As of FS-8321 - Getting issue details... STATUS the old behavior now requires the variable media_mix_inbound_outbound_codecs=true as the new default behavior is to offer the exact same codecs on the outbound B-leg that were offered on the inbound A-leg.

Usage

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

media_webrtc

Usage

{media_webrtc=true}

memory_debug

min_dup_digit_spacing_ms

Usage

<action application="set" data="min_dup_digit_spacing_ms=40"/>
<action application="spandsp_start_dtmf" />

mod_dir

monitor_early_media_fail

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

this variable only works when ignore_early_media is set to true.

Usage

<action application="bridge" data="{ignore_early_media=true,monitor_early_media_fail=user_busy:2:480+620!destination_out_of_order:2:1776.7}sofia/dial/string"/>

monitor_early_media_ring

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

this variable only works when ignore_early_media is set to true.

Usage

<action application="bridge" data="{monitor_early_media_ring_total=3,monitor_early_media_ring=usring:1:440.0+480.0!ukring:2:400+450}sofia/gateway/yourgateway/1239@conference.freeswitch.org"/>

monitor_early_media_ring_total

Usage

<action application="bridge" data="{ignore_early_media=true,monitor_early_media_ring_total=3,monitor_early_media_ring=us_ring:1:440.0+480.0}sofia/dial/string"/>

monitor_fail_dispo

Usage

<action application="bridge" data="{ignore_early_media=true,monitor_fail_dispo=strange_bleep_attack,monitor_early_media_fail=user_busy:2:1234}sofia/dial/string"/>

monitor_ring_dispo

myid

nat_private_addr

If you use auto-nat and a public IP address was detected, this variable will be set with the private IP address to which the detected public IP address is mapped.

nat_public_addr

If you use auto-nat and FreeSWITCH was able to identify the public IP address pointing to your server, this variable will be set with the auto detected public IP address.

string

If you use auto-nat FreeSWITCH will set this global variable to the type of auto nat device it detected this can be either pmp or upnp.

NDLB_support_asterisk_missing_srtp_auth

new_name

no_throttle_limits

Usage

{no_throttle_limits=true}

nonexistantvar

original_caller_id_name

original_caller_id_number

original_destination_number

originate_continue_on_timeout

Usage

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

originate_delay_start

Usage

<action application="bridge" data="sofia/profile/dest1:_:{originate_delay_start=10000}sofia/profile/dest2:_:{originate_delay_start=15000}sofia/profile/dest3"/>

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

<action application="bridge","users/1000:_:{originate_delay_start=8000}user/2302:_:{originate_delay_start=14000}sofia/gateway/flowroute/1231231234"/>

Assuming all users just let it ring:

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

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

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

originate_disposition

The value is updated after every bridge attempt, if the bridge is not successful.

originate_retries

originate_retry_sleep_ms

Usage

Wait 500ms between sending invite messages to the called gateway:

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

originate_timeout

Default Value: 60

Usage

Useful if you are using multiple gateways. In this example, FreeSWITCH will wait 2 seconds for 192.168.1.4 to respond to the invite message before trying the next gateway:

<action application="set" data="originate_timeout=2"/>
<action application="bridge" data="sofia/default/$1@192.168.1.4|sofia/default/$1@192.168.1.5"/>

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

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

originate_timeout

Default Value: 60

Usage

Useful if you are using multiple gateways. In this example, FreeSWITCH will wait 2 seconds for 192.168.1.4 to respond to the invite message before trying the next gateway:

<action application="set" data="originate_timeout=2"/>
<action application="bridge" data="sofia/default/$1@192.168.1.4|sofia/default/$1@192.168.1.5"/>

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

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

originating_leg_uuid

Usage

In A-leg CDR:

<uuid>cb5f5b90-75a0-11e0-873b-d1cba9e0f1b8</uuid>
<call_uuid>cb5f5b90-75a0-11e0-873b-d1cba9e0f1b8</call_uuid>

In B-leg CDR:

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

<uuid>cb8633aa-75a0-11e0-873d-d1cba9e0f1b8</uuid>
<call_uuid>cb5f5b90-75a0-11e0-873b-d1cba9e0f1b8</call_uuid>
<originating_leg_uuid>cb5f5b90-75a0-11e0-873b-d1cba9e0f1b8</originating_leg_uuid>

origination_callee_id_name

Usage

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

<action application="bridge" data="{origination_callee_id_name=Reginald}sofia/gateway/provider/<Reginald's cellphone number>" />

origination_callee_id_number

Usage

<action application="bridge" data="{origination_callee_id_name=Reginald,origination_callee_id_number=2332}sofia/gateway/provider/<Reginald's cellphone number>" />

origination_caller_id_name

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

Usage

<action application="set" data="origination_caller_id_name=Uncle Sam"/>

origination_caller_id_number

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

Usage

if you want to relay the Caller ID Number of an incoming PSTN call via FXO gateway, comment out this variable.

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

origination_cancel_key

Usage

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

origination_channel_name

Usage

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

origination_nested_vars

origination_privacy

Options:

• screen
• hide_name
• hide_number

Usage

<action application="set" data="origination_privacy=hide_name"/>

origination_timeout

origination_uuid

Bridge also uses the origination syntax so you can also pre-allocate the UUID for the new channel resulting from a bridge by using the {}/[] syntax and specifying origination_uuid there, too.

Usage

originate [origination_uuid=....]sofia/<profile>/<extension>

originator

originator_codec

Usage

<action application="set" data="originator_codec=PCMU"/>

originator_video_codec

other_loopback_leg_uuid

outbound_redirect_fatal

Usage

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

pa_hold_file

pa_ring_file

park_after_bridge

Usage

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

park_timeout

Usage

<action application="set" data="park_timeout=30"/>
<action application="park"/>

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

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

passthru_ptime_mismatch

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.

<action application="bridge" data="{passthru_ptime_mismatch=true}sofia/gateway/trunk/$1"/>

Globally in vars.xml

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

playback_last_offset_pos

playback_ms

playback_samples

playback_sleep_val

Usage

To play a list of short files one right after the other, with no pause in between:

<action application="set" data="playback_sleep_val=0"/>

playback_terminator_used

playback_terminators

Syntax: playback_terminators=123456789*0# | any | none

Allows you to set which DTMF tones, if pressed during the playback of a file or during mod_dptools: play_and_detect_speech, will terminate playback.

Digits not specified as terminators will be collected, but not interrupt playback.

• none - undefines the terminator digit, meaning no DTMF tone will interrupt playback
• any - any DTMF will interrupt playback

Modifiers:

• + (plus) - includes the following terminator digit along with the other DTMF digits in the returned string
• x (letter ex) - includes the following terminator digit in the returned string and also returns SWITCH_STATUS_RESTART

Usage

Either * (star) or # (bar/pound/square) will interrupt the subsequent playback, tone_stream, or say operation:

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

If DTMF 9 is dialed during playback, it will interrupt playback and return the digit 9 along with any other dialed digits; if * is dialed it will only interrupt playback without being included in the DTMF string returned:

<action application="set" data="playback_terminators=*+9"/>

playback_timeout_sec

Usage

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

pound_replace

presence_data_cols

Use : to separate the column names in presence_data_cols

You must define the field in the channels table first:

Usage

In conf/vars.xml:

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

presence_id

Usage

<action application="set" data="presence_id=${dialed_extension}@${domain_name}"/>

process_cdr

Options:

• false - do not process the record
• true - process records for both legs. This is the default.
• a_only - only process records on the a leg
• b_only - only process records on the b leg

This variable is unconditionally exported.

Usage

<action application="set" data="process_cdr=a_only"/>

profile_created_time

Usage

In the event that a call is transferred, this is the effective created_time for that transfer. For example, if you did this 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.

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_time

progress_timeout

I set my progress_timeout to 13 seconds to accommodate for slow-setup mobile connections, however, it can probably be shorter. -- Avi Marcus

Two scenarios:

1. If a carrier is having issues with their platform, all calls might timeout.
2. If it's not a carrier issue -- if a single carrier timed out at 13 seconds and the user actually waited, the call rarely connected via any other carrier (although they all expired at 13 seconds).

See also: Early Media

Usage

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

progress_uepoch

progressmsec

progresssec

progressusec

proto_specific_hangup_cause

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 dialed doesn't exist.

Usage

<extension name="nothing_left" continue="true">
  <condition break="always">
    <action application="set" data="proto_specific_hangup_cause=sip:503"/>
    <action application="hangup"/>
  </condition>
</extension>

Example SIP Response Map

<extension name="from_gw_to_internal">
  <condition field="destination_number" expression="^(.*)$">
    <action application="set" data="hangup_after_bridge=true"/>
    <action application="set" data="continue_on_fail=19"/>
    <action application="bridge" data="{sip_cid_type=none}sofia/gateway/gw/$1"/>
    <action application="transfer" data="480to503"/>
  </condition>
</extension>

<extension name="480to503">
  <condition field="${proto_specific_hangup_cause}" expression="sip:480">
    <action application="set" data="sip_ignore_remote_cause=true"/>
    <action application="respond" data="503"/>
    <action application="hangup" data="NORMAL_CIRCUIT_CONGESTION"/>
  </condition>
</extension>

proxy_media

See also: Proxy Media

Usage

Late Negotiation () must be enabled in sip profile for this to work in the dialplan.

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

rdnis

read_codec

read_rate

read_result

read_terminator_used

RECORD_APPEND

Usage

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

RECORD_ARTIST

Usage

<action application="set" data="RECORD_ARTIST=Unknown"/>

RECORD_BRIDGE_REQ

Usage

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

RECORD_COMMENT

Usage

<action application="set" data="RECORD_COMMENT=This is a blog spot"/>

RECORD_COPYRIGHT

Usage

<action application="set" data="RECORD_COPYRIGHT=(c)2007-me"/>

RECORD_DATE

Usage

<action application="set" data="RECORD_DATE=${strftime(%Y-%m-%d %H:%M)}"/>

RECORD_DISCARDED

record_fill_cng

Usage

<action application="set" data="record_fill_cng=1200"/>

RECORD_HANGUP_ON_ERROR

Usage

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

RECORD_MIN_SEC

Usage

<action application="set" data="RECORD_MIN_SEC=0"/>

record_ms

record_post_process_exec_api

Usage

<action application="set" data="record_post_process_exec_api=api_here:api_arg1 api_arg2 api_arg3" />

record_post_process_exec_app

Usage

<action application="set" data="record_post_process_exec_app=lua::vm_test.lua" />

RECORD_READ_ONLY

Usage

From the perspective of FreeSWITCH looking toward the endpoint

Aleg is recording, and only incoming audio read from the channel goes into file

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

or Bleg is recording, and only incoming audio read from the channel goes into file

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

See Also:

RECORD_WRITE_ONLY

record_restart_limit_on_dtmf

Usage

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

record_sample_rate

Usage

<action application="set" data="record_sample_rate=8000"/>

RECORD_SOFTWARE

Usage

<action application="set" data="RECORD_SOFTWARE=FreeSWITCH"/>

RECORD_STEREO

Usage

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

RECORD_STEREO_SWAP

RECORD_TITLE

Usage

<action application="set" data="RECORD_TITLE=Recording ${destination_number} ${caller_id_number} ${strftime(%Y-%m-%d %H:%M)}"/>

record_waste_resources

Usage

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

<action application="set" data="record_waste_resources=1300"/>

RECORD_WRITE_ONLY

boolean Record write stream only.

Usage

From the perspective of FreeSWITCH looking toward the endpoint

Aleg is recording, and only outgoing audio written to the channel goes into file

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

or Bleg is recording, and only outgoing audio written to the channel goes into file

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

See Also

RECORD_READ_ONLY

recording_follow_transfer

Usage

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

recordings_dir

recovery_profile_name

remote_media_ip

remote_media_port

remote_video_ip

remote_video_port

ringback

Syntax: <action application="set" data="ringback=[data]"/>

Where data is either the full path ( must be full path!) to an audio file or teletone generation markup language.

You may also combine this with the ring_ready application. In this case the caller hears the normal ringback tone for a fraction of a second, then he hears the defined ringback media.

Since r15025 local ringback tone is generated only when there is no early media. To force local ringback despite early media use the ignore_early_media channel variable.

Ringback will not be sent unless the variable is set and there's a ringing indication from the party that is being called. See instant_ringback if you want to fake ringback even when ringing indication hasn't been received yet.

Usage

Using the French ring tone defined in vars.xml:

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

US ring tone:

<action application="set" data="ringback=%(2000,4000,440.0,480.0)"/>

You can send ringback after the call is answered:

def handler(uuid):
    session = PySession(uuid)
    session.answer()
    session.speak("Hello, connecting your call to the first available agent")
    session.execute("set","call_timeout=30")
    session.execute("set","continue_on_fail=true")
    session.execute("set","hangup_after_bridge=true")
    session.execute("set","ringback=%(2000,4000,440.0,480.0)")
    session.execute("bridge","sofia/foo/bar@test.com,sofia/foo/baz@test2.com:5070")
    session.speak("doh!")

rss_alt_config

rtcp_audio_interval_msec

setting the rtp_timer_name to none will create all sorts of undesired side effects with RTCP (packets not sent, RTCP autoadjust failing, etc.).

rtcp_mux

rtcp_octet_count

rtcp_packet_count

rtp_adv_audio_ip

Usage

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

rtp_append_audio_sdp

It should/must be set before bridging.

Usage

<action application="export" data="rtp_append_audio_sdp=a=fmtp:18 annexb=no"/>

rtp_assume_rtcp

rtp_autoflush

It is worth it to set to true if you have crappy network conditions where you are hearing hiccups it's related to jitter. Sometimes you have the other side sending audio too fast, then this option set to false will smooth it out but if you have it set to false in jitter conditions it tricks it into moving too fast.

rtp_autoflush_during_bridge

rtp_disable_hold

Usage

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

rtp_disable_vad_in

rtp_disable_vad_out

rtp_enable_vad_in

rtp_enable_vad_out

rtp_force_audio_fmtp

rtp_hold_timeout_sec

rtp_jitter_buffer_during_bridge

Usage

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

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

rtp_manual_rtp_bugs

Usage

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

rtp_negotiate_near_match

Usage

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

rtp_rewrite_timestamps

rtp_sdes_suites

rtp_secure_media

Syntax: rtp_secure_media=<permission>[:<list of encryption suites>]

sip_secure_media is no longer implemented

permission:

• mandatory - Accept/Offer SAVP negotiation ONLY
• optional - Accept/Offer SAVP/AVP with SAVP preferred
• forbidden - More useful for inbound to deny SAVP negotiation
• deprecatedfalse - implies forbidden
• deprecatedtrue - implies mandatory
• default - accept SAVP inbound if offered

encryption suites:

• AEAD_AES_256_GCM_8
• AEAD_AES_128_GCM_8
• AES_CM_256_HMAC_SHA1_80
• AES_CM_192_HMAC_SHA1_80
• AES_CM_128_HMAC_SHA1_80
• AES_CM_256_HMAC_SHA1_32
• AES_CM_192_HMAC_SHA1_32
• AES_CM_128_HMAC_SHA1_32
• AES_CM_128_NULL_AUTH

Examples:

• rtp_secure_media=mandatory:AES_CM_256_HMAC_SHA1_80,AES_CM_256_HMAC_SHA1_32
• rtp_secure_media=true:AES_CM_256_HMAC_SHA1_80,AES_CM_256_HMAC_SHA1_32
• rtp_secure_media=optional:AES_CM_256_HMAC_SHA1_80
• rtp_secure_media=true:AES_CM_256_HMAC_SHA1_80

rtp_secure_media_inbound

Same syntax as rtp_secure_media: rtp_secure_media_inbound=<permission>[:<list of encryption suites>]

Examples:

rtp_secure_media_inbound=true:AEAD_AES_256_GCM_8,AES_CM_256_HMAC_SHA1_80,AES_CM_256_HMAC_SHA1_32

rtp_secure_media_inbound=mandatory:AEAD_AES_256_GCM_8

rtp_secure_media_outbound

Same syntax as rtp_secure_media: rtp_secure_media_outbound=<permission>[:<list of encryption suites>]

Examples:

rtp_secure_media_outbound=true:AEAD_AES_128_GCM_8

rtp_secure_media_outbound=optional:AEAD_AES_128_GCM_8

rtp_secure_media_suites

Syntax: rtp_secure_media_suites=<list of encryption suites>

rtp_stun_ping

rtp_timeout_sec

rtp_timer_name

run_dir

script_dir

sdp_m_per_ptime

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 it's 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_transcoding or absolute_codec_string as well.

Usage

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

sdp_secure_savp_only

Usage

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

send_silence_when_idle

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

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

Usage

<action application="set" data="send_silence_when_idle=400"/>

session_in_hangup_hook

Usage

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

signal_bond

signal_bridge_to

sip

sip_acl_authed_by

sip_acl_token

sip_auth_method

Known values:

• REGISTER

sip_auth_password

Usage

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

sip_auth_realm

Usage

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

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:

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

sip_auth_username

Usage

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

sip_authorized

sip_auto_answer

Usage

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

sip_auto_answer_detected

sip_auto_answer_notify

sip_auto_answer_suppress_notify

sip_auto_simplify

Usage

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

sip_call_id

sip_callee_id_name

Usage

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

<action application="set" data="sip_callee_id_name=Reginald" />
<action application="set" data="sip_callee_id_number=2332" />
<action application="bridge" data="sofia/gateway/provider/<Reginald's cellphone number>" />

sip_cid_in_1xx

Usage

<action application="set" data="sip_cid_in_1xx=false"/>

sip_cid_type

Usage

Send no extra caller id info (Caller ID will be in the SIP From):

this will not work for gateways. You must set caller-id-in-from=true in the gateway settings.

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

Send Remote-Party-ID (default):

{sip_cid_type=rpid}sofia/default/user@example.com

Send P-Asserted-Identity:

you must set privacy flag, otherwise P-Preferred-Identity will be inserted instead of P-Asserted-Identity.

{sip_cid_type=pid}sofia/default/user@example.com

Send RPID with chosen content and privacy flags (+ delimited, none to clear all flags):

{sip_cid_type=rpid,origination_caller_id_name=test,origination_caller_id_number=1234,origination_privacy=screen+hide_name+hide_number}sofia/default/user@example.com

Send Remote-Party-ID with chosen content:

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

sip_codec_negotiation

Options:

• generous
• scrooge
• greedy

Before 74a0cfd1e101413a3941c41d04ee01d8df2ae418, sip_codec_negotiation will always be overwritten by the sofia profile configuration param sip-codec-negotiation. As of this revision, the codec negotiation can be changed on a per-call basis.

Usage

<action application="set" data="sip_codec_negotiation=greedy"/>

sip_contact_host

Usage

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

sip_contact_port

Usage

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

sip_contact_user

Usage

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

sip_copy_custom_headers

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"/>

sip_copy_multipart

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.

Usage

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

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

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

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

sip_destination_url

sip_enable_soa

Usage

<action application="set" data="bypass_media=true"/>
<action application="export" data="sip_enable_soa=false"/>

sip_exclude_contact

sip_execute_on_image

sip_force_audio_fmtp

sip_force_full_from

sip_force_full_to

sip_force_video_fmtp

sip_from_comment

sip_from_display

Usage

Clear the user part of the SIP message sent to the Leg B call:

<action application="bridge" data="{sip_from_display=_undef_}user/1001"/>

sip_from_host

Usage

If your request header contains:

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

Then the value for this field would be:

server.example.com

sip_from_port

Usage

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

sip_from_uri

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.

sip_from_user

Usage

If your request header contains:

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

Then the value for this field would be:

1000

sip_from_user_stripped

sip_gateway

sip_gateway_name

Usage

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

sip_profile_name=gateway
sip_gateway_name=ntl
sofia_profile_name=internal

sip_h_Referred-By

Usage

<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"/>

sip_handle_full_from

sip_handle_full_to

sip_hangup_disposition

<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:

sip_has_crypto

sip_header_name

sip_history_info

sip_ignore_183nosdp

Usage

<action application="set" data="sip_ignore_183nosdp=true"/>

sip_ignore_reinvites

Usage

Don't allow any re-INVITEs once bridged:

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

sip_ignore_remote_cause

sip_invite_call_id

Usage

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

sip_invite_contact_params

sip_invite_domain

Usage

<action application="bridge" data="{sip_invite_domain=${sip_from_host}}sofia/gateway/gw1/$1@domain.org"/>

sip_invite_from_params

Usage

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

Returns:

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

sip_invite_from_uri

sip_invite_full_from

sip_invite_full_to

sip_invite_params

sip_invite_record_route

sip_invite_req_uri

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

Usage

<action application="bridge" data="{sip_invite_req_uri=sip:11112222@test1.com}sofia/external/33334444%192.168.4.6"/>

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>

sip_invite_route_uri

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>

sip_invite_tel_params

Usage

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

<action application="bridge" data="{sip_invite_tel_params=npdi=yes;rn=555000001,sip_invite_params=user=phone}sofia/gateway/test_gw/555000002"/>

sip_invite_to_params

sip_invite_to_uri

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>

sip_jitter_buffer_plc

Usage

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

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

sip_local_sdp_str

sip_local_url

sip_looped_call

Usage

In conf/dialplan/public.xml:

<condition field="${sip_looped_call}" expression="^true$">

<extension name="unloop">
  <condition field="${unroll_loops}" expression="^true$"/>
  <condition field="${sip_looped_call}" expression="^true$">
    <action application="deflect" data="${destination_number}"/>
  </condition>
</extension>

sip_mirror_remote_audio_codec_payload

This variable can be set globally or per channel.

sip_nat_detected

sip_network_destination

Usage

<action application="bridge" data="{sip_network_destination=sip:5551234567@66.243.109.243:10005}sofia/external/5551234567@172.16.110.45:5060"/>

sip_outgoing_call_id

sip_p_rtp_stat

sip_profile

sip_profile_name

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=gateway
sip_gateway_name=ntl
sofia_profile_name=internal

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

sip_profile_name=internal
sip_gateway_name=
sofia_profile_name=internal

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

sip_profile_name=internal
sip_gateway_name=
sofia_profile_name=external

sip_received_ip

sip_received_port

sip_recover_contact

sip_recover_via

sip_recovery_break_rfc

This variable can be set globally or per channel

sip_refer_reply

sip_referred_by_cid

sip_referred_by_user_stripped

sip_renegotiate_codec_on_reinvite

Usage

<action application="bridge" data="{sip_renegotiate_codec_on_reinvite=true}sofia/gateway/trunk/$1"/>

Globally in vars.xml:

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

In a SIP profile:

<param name="renegotiate-codec-on-reinvite" value="true"/>

sip_reply_host

sip_request_host

Usage

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

sip_request_port

Usage

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):

REGISTER sip:172.16.44.8 SIP/2.0

sip_require_timer

Usage

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

sip_route_uri

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.

See also: FS-5349 - Getting issue details... STATUS

Usage

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

bridge {sip_route_uri=sip:USER@HOSTNAME.or.IP.ADDRESS}

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:${sofia_contact(${dialed_extension}@${domain_name})},sip_invite_req_uri=sip:${dialled_ddi}@${domain_name}}user/${user_data(${dialed_extension}@${domain_name} attr id)}@${domain_name}

sip_rtp_rxstat

sip_rtp_txstat

sip_secure_media

sip_sticky_contact

sip_subject

sip_term_cause

sip_term_status

sip_to_comment

sip_to_host

Usage

If your request header contains:

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

Then the value for this field would be:

server.example.com

sip_to_port

Usage

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):

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

sip_to_uri

sip_to_user

Usage

To set manually, use:

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

Then the value for this field would be:

1000

As options you can make call:

bridge(sofia/external/333@biloxi.com^1234)

or make call:

bridge(sofia/external/333^1234@biloxi.com)

sip_transport

sip_use_gateway

sip_user_agent

Usage

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

sip_via_host

sip_via_port

sip_via_protocol

sip_via_rport

sip_video_fmtp

sip_video_pt

sip_wait_for_aleg_ack

Usage

<action application="bridge" data="{sip_wait_for_aleg_ack=true}sofia/internal/foo@bar.com"/>

skeleton

Usage

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

skip_cdr_causes

sleep_eat_digits

this is a change in default behavior as the sleep application previously ate DTMFs without exception. Be sure to set sleep_eat_digits to true in order to preserve the previous behavior.

Usage

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

socket_host

socket_path

SOFIA_CRYPTO_MANDATORY_VARIABLE

SOFIA_HAS_CRYPTO_VARIABLE

sofia_profile_domain_name

sofia_profile_name

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

sofia_record_file

SOFIA_REFER_TO_VARIABLE

SOFIA_REPLACES_HEADER

SOFIA_SECURE_MEDIA_CONFIRMED_VARIABLE

SOFIA_SECURE_MEDIA_VARIABLE

SOFIA_SESSION_TIMEOUT

sound_prefix

sounds_dir

spandsp_dtmf_rx_filter_dialtone

Usage

<action application="set" data="spandsp_dtmf_rx_filter_dialtone=true"/>
<action application="spandsp_start_dtmf" />

spandsp_dtmf_rx_reverse_twist

Usage

<action application="set" data="spandsp_dtmf_rx_reverse_twist=6"/>
<action application="spandsp_start_dtmf" />

spandsp_dtmf_rx_threshold

Usage

<action application="set" data="spandsp_dtmf_rx_threshold=-42"/>
<action application="spandsp_start_dtmf" />

spandsp_dtmf_rx_twist

Usage

<action application="set" data="spandsp_dtmf_rx_twist=8"/>
<action application="spandsp_start_dtmf" />

star_replace

start_epoch

start_stamp

start_uepoch

storage_dir

stream_prebuffer

suppress_cng

Usage

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

switch_m_sdp

SWITCH_PLAYBACK_TERMINATOR_USED

switch_r_sdp

Usage

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.

<action application="set"><![CDATA[switch_r_sdp=(sdp here)
]]>
</action>

switch_serial

SWITCH_UUID_BRIDGE

temp_dir

temp_hold_music

Usage

<action application="set" data="temp_hold_music=local_stream://alternate_moh"/>

timer_name

Usage

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

timezone

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

See also: Time of Day and Holiday Routing.

Usage

You can set the time zone globally for Freeswitch in /conf/vars.xml

<X-PRE-PROCESS cmd="set" data="timezone=America/Toronto"/>

Specify the timezone in the dialplan:

<action application="set" data="timezone=Asia/Seoul"/>

<action application="set" data="timezone=GMT0"/>

<action application="set" data="timezone=America/New_York"/>

In the directory:

<param name="timezone" value="America/New_York"/>

tod_tz_offset

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

Usage

To set the offset global in conf/vars.xml:

<X-PRE-PROCESS cmd="set" data="tod_tz_offset=5"/>

<action application="set" data="tod_tz_offset=5"/>

tone_detect_expires

tone_detect_sleep

transfer_after_bridge

Variable gets cleaned after bridge is destroyed and leg is transferred to the specified dialplan.

Usage

<action application="set" data="transfer_after_bridge=1000"/>

Note the : separator

<action application="set" data="transfer_after_bridge=1000:XML:default"/>

transfer_fallback_extension

transfer_history

transfer_on_fail

you must also set failure_causes

Argument syntax: hangupcauses destination dialplan context

Usage

<action application="set" data="failure_causes=UNALLOCATED_NUMBER"/>
<action application="set" data="transfer_on_fail=UNALLOCATED_NUMBER"/>

<action application="set" data="transfer_on_fail=1"/>

transfer_ringback

Usage

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

transfer_source

transfer_to

See Also

• mod_tts_commandline
• mod_flite
• tts_commandline

tts_voice

uduration

UNIQUEID

user_context

user_name

uuid_bridge_continue_on_cancel

Usage

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

verbose_presence

verbose_sdp

By default FreeSWITCH sets verbose_sdp=false which doesn't include an rtpmap for static payload types. If your equipment doesn't support this (shame on them) set verbose_sdp=true which will include rtpmaps for any static payload types (previous default behavior).

View more details about this in the mod_conference page 

mod_conference#SettableChannelVariables

video_mirror_input

video_possible

video_read_codec

video_read_rate

video_write_codec

video_write_rate

vm_cc

vm_message_ext

This variable is set in the dial plan.

Default available options:

• mp3
• wav

Usage

<action application="set" data="vm_message_ext=mp3"/>

vmd_detect

vname

voicemail_account

voicemail_alternate_greet_id

Usage

<user id="johnsmith" number-alias="1000">
   <variables>
       <variable name="mailbox" value="1000"/>
       <variable name="effective_caller_id_name" value="1000"/>
       <variable name="effective_caller_id_number" value="1000"/>
       <variable name="voicemail_alternate_greet_id" value="1000"/>
   </variables>
</user>

voicemail_authorized

voicemail_caller_id_name

voicemail_caller_id_number

voicemail_current_folder

voicemail_domain

Usage

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>

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_codec

write_rate

xfer_uuids

xml_cdr_base

zrtp_enabled

This global variable is set by the FreeSWITCH core to the value of the global rtp-enable-zrtp setting in switch.conf.xml. For more information see the SIP TLS page.