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__Channel Variables

About​

There are a number of channel variables that can be set in the dialplan or your application to affect the progress or settings for a call. Channel variables can also be set in dialstrings.

This page parallels the __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.

Channel Variables in the XML Dialplan​

Channel variables are set, appropriately enough, with the set application like this:

<action application="set" data="var_name=var value"/>

Reading channel variables requires the ${} syntax:

<action application="log" data="INFO The value in the var_name chan var is ${var_name}"/>
<condition field="${var_name}" expression="some text">
Scoped Variables​

- Channel variables used to be global to the session.
- With Scoped variables, you can now set variables that only exist for the duration of that single application execution and any subsequent applications under it.
- Applications can leverage this concept to use named input params

As of June 15 (git commit b2c3199f) you may now have scoped variables that will stay in effect only for the duration of a given dialplan application. Example:

<action application="log" data="INFO myvar is '${myvar}'"/>
<action application="log" data="%[myvar=Hello]INFO myvar is '${myvar}'"/>
<action application="log" data="INFO myvar is '${myvar}'"/>
<action application="myapp" data="%[var1=val1,var2=val2]mydata"/>

Channel Variables in Dial strings​

The syntax of using { } versus < > versus [] is explained below.

* {foo=bar} is **only** valid at the beginning of the dial string. It will set the same variables on **every** channel, but does not do so for enterprise bridging/originate.
* <foo=bar> is **only** valid at the beginning of the dial string. It will set the same variables on **every** channel, including all those in an enterprise bridging/originate.
* \[foo=bar\] goes before each individual dial string and will set the variable values specified for only this channel.
* The following example sets the foo variable for all channels implemented and chan=1 is specific to blah, while chan=2 is specific to blah2

{foo=bar}[chan=1]sofia/default/blah@baz.com,[chan=2]sofia/default/blah2@baz.com

  • Setting multiple variables can be accomplished by comma delimiting:

[var1=abc,var2=def,var3=ghi]sofia/default/blah@baz.com

  • If you would like to have variables in [] override the same variables set in {}, you may set 'local_var_clobber=true' inside {}. For example:

{local_var_clobber=true,rtp_secure_media=true}sofia/default/blah1@baz.com|sofia/default/johndoe@example.com|[rtp_secure_media=false]sofia/default/janedoe@acme.com

You must also set this local_var_clobber variable to true when you want to override channel variables that have been exported to your b-legs in your dialplan.

In this example, the legs for blah1@baz.com and johndoe@example.com would be set to offer SRTP (RTP/SAVP) while janedoe@acme.com would not receive an SRTP offer (she would see RTP/AVP instead).

Handling Variables With Commas​

Since commas are the default delimiter inside of {} and [] it is necessary to have an alternate means of delimiting values that actually contain commas.

To set a variable with a "," inside of it (e.g. the absolute_codec_string), you must tell FreeSWITCH what delimiter to use instead for THIS value.

The syntax is to put a prefix of ^^: at the start of the value and replace the commas in the value with ":". Then, FreeSWITCH will substitute commas back in when setting the value for the channel.

You can use (almost) any character you like though, it uses the one after the initial "^^"

{absolute_codec_string=^^:PCMA@8000h@20i@64000b:PCMU@8000h@20i@64000b:G729@8000h@20i@8000b,leg_time_out=10,process_cdr=b_only}

This approach however does not work when you are setting sip_h_, sip_rh_ and sip_ph headers. If you want to pass a comma into the contents of a private header then you must escape the comma with a preceding backslash:

{sip_h_X-My-Header=one\,two\,three,leg_time_out=10,process_cdr=b_only}

Exporting Channel Variables in Bridge Operations​

You can export a variable from one call leg (A) to the other call leg (B) by using the 'export_vars' variable which is a comma separated list of variables that you wish to propagate across calls. Usage:

<action application="set" data="export_vars=myvar,myvar2,foo,bar"/>

You can also set a variable on the (A) leg and add it to the export list in one step with the export application:

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

Using Channel Variables in Dialplan Condition Statements​

Channel variables can be used in conditions:

Custom Channel Variables​

Additionally, you may set any number of unique channel variables for your own purposes and even elect to log them to the CDR. In order to set any channel variable, use the application "set" to effect the setting. You can also set channel variables on the command issued via XML-RPC or Event Socket by adding {myvar=myval,myvar1=myval1} e.g.

originate {ignore_early_media=true}sofia/mydomain.com/18005551212@1.2.3.4 15555551212

If the value you are setting has a space in it then you will need to enclose the value in quotes. For example:

originate {fax_ident=1231231234,fax_header='Fax Test'}sofia/gateway/outbound.fax/1004 &txfax(/tmp/fax.tiff)

Channel Variable Manipulation​

Channel variables can be manipulated for varied results. For example, you can trim bits and pieces of a channel variable, perhaps to get the first three digits of a phone number. See Manipulating Channel Variables for more information.

Channel Variable Scope w/ Example​

Consider the following example that shows where each application sets it's variable :

  <extension name="test" continue="false">
     <condition field="destination_number" expression="^test([0-9]+)$">
       <action application="set" data="fruit=tomato" /> <!-- Set variable in local channel -->
       <action application="export" data="veggie=tomato" /> <!-- Set variable in local channel and export it to new channels we bridge to -->
       <action application="bridge" data="{meat=tomato}sofia/gateway/testaccount/1234" /><!-- bridge new channel and set variable only in new channel -->
     </condition>
   </extension>

Leg A ( the channel that called the dial plan ) will have the following variables set :

fruit: tomato veggie: tomato

Leg B ( the channel created with sofia/gateway/testaccount/1234 ) will have the following variables set :

veggie: tomato meat: tomato

Accessing Channel Variables in Other Environments​

In addition to the dialplan, you can get at channel variables in other environments as well.

In a FreeSWITCH module, written in C, you would use:

  switch_channel_set_variable(channel,"name","value");  
  char* result = switch_channel_get_variable(channel,"name");    
  char* result = switch_channel_get_variable_partner(channel,"name");

In the console (or fs_cli, implemented in mod_commands), you can use:

uuid_getvar <uuid> <name>
uuid_setvar <uuid> <name> [<value>]
uuid_setvar_multi <uuid> <name>=<value>[;<name>=<value>[;...]]

Alternatively, you can call uuid_dump to get all the variables, or you can use the eval command, adding the prefix variable_ to your key:

  uuid_dump <uuid>  
  eval uuid:<uuid> ${variable_<name>}

In an event socket, just extend the above with the 'api' prefix.

  api uuid_getvar <uuid> <name>

In Lua, there are several ways to interact with variables. In the freeswitch.Session() invocation that creates a new Session object, variables go in square brackets:

  s = freeswitch.Session("[myname=myvars]sofia/localhost/1003");

Once you have a Session object you can call the

  result = session:getVariable(“name");  
  session:setVariable(“name","value");

methods. The variables actually belong to the channel object that is bonded to the session, but since there is a one-to-one binding, Lua treats sessions and channels as a single object.

Info Application Variable Names (variable_xxxx)​

Some variables, as shown from the info app, may have variable_ in front of their names. For example, if you pass a header variable called 'type' from the proxy server, it will get displayed as 'variable_sip_h_type' in FreeSWITCH. To access that variable, you should strip off the variable_, and just do ${sip\_h\_type}. Other variables shown in the info app are prepended with channel, which should be stripped as well. The example below show a list of info app variables and the corresponding channel variable names:

Info variable namechannel variable nameDescription
Channel-StatestateCurrent state of the call
Channel-State-Numberstate_numberInteger
Channel-Namechannel_nameChannel name
Unique-IDuuiduuid of this channel's call leg
Call-DirectiondirectionInbound or Outbound
Answer-Statestate-
Channel-Read-Codec-Nameread_codecthe read codec variable mean the source codec
Channel-Read-Codec-Rateread_ratethe source rate
Channel-Write-Codec-Namewrite_codecthe destination codec same to write_codec if not transcoded
Channel-Write-Codec-Ratewrite_ratedestination rate same to read rate if not transcoded
Caller-Usernameusername.
Caller-Dialplandialplanuser dialplan like xml, lua, enum, lcr
Caller-Caller-ID-Namecaller_id_name.
Caller-Caller-ID-Numbercaller_id_number.
Caller-ANIaniANI of caller, frequently the same as caller ID number
Caller-ANI-IIaniiiANI II Digits (OLI - Originating Line Information), if available. Refer Here
Caller-Network-Addrnetwork_addrIP address of calling party
Caller-Destination-Numberdestination_numberDestination (dialed) number
Caller-Unique-IDuuidThis channel's uuid
Caller-SourcesourceSource module, i.e. mod_sofia, mod_openzap, etc.
Caller-ContextcontextDialplan context
Caller-RDNISrdnisRedirected DNIS info. See mod_dptools: transfer application
Caller-Channel-Namechannel_name.
Caller-Profile-Indexprofile_index.
Caller-Channel-Created-Timecreated_time.
Caller-Channel-Answered-Timeanswered_time.
Caller-Channel-Hangup-Timehangup_time.
Caller-Channel-Transfer-Timetransfer_time.
Caller-Screen-Bitscreen_bit.
Caller-Privacy-Hide-Nameprivacy_hide_name.
Caller-Privacy-Hide-Numberprivacy_hide_numberThis variable tells you if the inbound call is asking for CLIR[Calling Line ID presentation Restriction] (either with anonymous method or Privacy:id method)
initial_callee_id_nameSets the callee id name during the 183. This allows the phone to see a name of who they are calling prior to the phone being answered. An example of setting this to the caller id name of the number being dialled: <action application="set" data="initial\_callee\_id\_name='${user\_data(${dialed\_extension}@${domain\_name} var effective\_caller\_id\_name)}'"/>
variable_sip_received_ipsip_received_ip.
variable_sip_received_portsip_received_port.
variable_sip_authorizedsip_authorized.
variable_sip_mailboxsip_mailbox.
variable_sip_auth_usernamesip_auth_username.
variable_sip_auth_realmsip_auth_realm.
variable_mailboxmailbox.
variable_user_nameuser_name.
variable_domain_namedomain_name.
variable_record_stereorecord_stereo.
variable_accountcodeaccountcodeAccountcode for the call. This is an arbitrary value. It can be defined in the user variables in the directory, or it can be set/modified from dialplan. The accountcode may be used to force a specific CDR CSV template for the call.
variable_user_contextuser_context.
variable_effective_caller_id_nameeffective_caller_id_name.
variable_effective_caller_id_numbereffective_caller_id_number.
variable_caller_domaincaller_domain.
variable_sip_from_usersip_from_user.
variable_sip_from_urisip_from_uri.
variable_sip_from_hostsip_from_host.
variable_sip_from_user_strippedsip_from_user_stripped.
variable_sip_from_tagsip_from_tag.
variable_sofia_profile_namesofia_profile_name.
variable_sofia_profile_domain_namesofia_profile_domain_name.
variable_sip_full_routesip_full_routeThe complete contents of the Route: header.
variable_sip_full_viasip_full_viaThe complete contents of the Via: header.
variable_sip_full_fromsip_full_fromThe complete contents of the From: header.
variable_sip_full_tosip_full_toThe complete contents of the To: header.
variable_sip_req_paramssip_req_params.
variable_sip_req_usersip_req_user.
variable_sip_req_urisip_req_uri.
variable_sip_req_hostsip_req_host.
variable_sip_to_paramssip_to_params.
variable_sip_to_tagsip_to_tag.
variable_sip_to_usersip_to_user.
variable_sip_to_urisip_to_uri.
variable_sip_to_hostsip_to_host.
variable_sip_contact_paramssip_contact_params.
variable_sip_contact_usersip_contact_user.
variable_sip_contact_portsip_contact_port.
variable_sip_contact_urisip_contact_uri.
variable_sip_contact_hostsip_contact_host.
variable_sip_invite_domainsip_invite_domain.
variable_channel_namechannel_name.
variable_sip_call_idsip_call_idSIP header Call-ID
variable_sip_user_agentsip_user_agent.
variable_sip_via_hostsip_via_host.
variable_sip_via_portsip_via_port.
variable_sip_via_rportsip_via_rport.
variable_presence_idpresence_id.
variable_sip_h_P-Key-Flagssip_h_p-key-flagsThis will contain the optional P-Key-Flags header(s) that may be received from calling endpoint.
variable_switch_r_sdpswitch_r_sdpThe whole SDP received from calling endpoint.
variable_remote_media_ipremote_media_ip.
variable_remote_media_portremote_media_port.
variable_write_codecwrite_codec.
variable_write_ratewrite_rate.
variable_endpoint_dispositionendpoint_disposition.
variable_dialed_extdialed_ext.
variable_transfer_ringbacktransfer_ringback.
variable_call_timeoutcall_timeout.
variable_hangup_after_bridgehangup_after_bridge.
variable_continue_on_failcontinue_on_fail.
variable_dialed_userdialed_user.
variable_dialed_domaindialed_domain.
variable_sip_redirect_contact_user_0sip_redirect_contact_user_0.
variable_sip_redirect_contact_host_0sip_redirect_contact_host_0.
variable_sip_h_Referred-Bysip_h_referred-by.
variable_sip_refer_tosip_refer_toThe full SIP URI received from a SIP Refer-To: response
variable_max_forwardsmax_forwards.
variable_originate_dispositionoriginate_disposition.
variable_read_codecread_codec.
variable_read_rateread_rate.
variable_openopen.
variable_use_profileuse_profile.
variable_current_applicationcurrent_application.
variable_ep_codec_stringep_codec_stringThis variable is only available if late negotiation is enabled on the profile. It's a readable string containing all the codecs proposed by the calling endpoint. This can be easily parsed in the dialplan.
variable_rtp_disable_holdrtp_disable_holdThis variable when set will disable the hold feature of the phone.
variable_sip_acl_authed_bysip_acl_authed_byThis variable holds what ACL rule allowed the call.
variable_curl_response_datacurl_response_dataThis variable stores the output from the last curl made.
variable_drop_dtmfdrop_dtmfSet on a channel to drop DTMF events on the way out.
variable_drop_dtmf_masking_filedrop_dtmf_masking_fileIf drop_dtmf is true play specified file for every tone received.
variable_drop_dtmf_masking_digitsdrop_dtmf_masking_digitsIf drop_dtmf is true play specified tone for every tone received.
sip_codec_negotiationsip_codec_negotiationsip_codec_negotiation is basically a channel variable equivalent of inbound-codec-negotiation.sip_codec_negotiation accepts "scrooge" & "greedy" as values.This means you can change codec negotiation on a per call basis.
Caller-Callee-ID-Name--
Caller-Callee-ID-Number--
Caller-Channel-Progress-Media-Time--
Caller-Channel-Progress-Time--
Caller-Direction--
Caller-Profile-Created-Timeprofile_created-
Caller-Transfer-Source--
Channel-Call-State--
Channel-Call-UUID--
Channel-HIT-Dialplan--
Channel-Read-Codec-Bit-Rate--
Channel-Write-Codec-Bit-Rate--
Core-UUID--
Event-Calling-File--
Event-Calling-Function--
Event-Calling-Line-Number--
Event-Date-GMT--
Event-Date-Local--
Event-Date-Timestamp--
Event-Name--
Event-Sequence--
FreeSWITCH-Hostname--
FreeSWITCH-IPv4--
FreeSWITCH-IPv6--
FreeSWITCH-Switchname--
Hunt-ANI--
Hunt-Callee-ID-Name--
Hunt-Callee-ID-Number--
Hunt-Caller-ID-Name--
Hunt-Caller-ID-Number--
Hunt-Channel-Answered-Time--
Hunt-Channel-Created-Time--
Hunt-Channel-Hangup-Time--
Hunt-Channel-Name--
Hunt-Channel-Progress-Media-Time--
Hunt-Channel-Progress-Time--
Hunt-Channel-Transfer-Time--
Hunt-Context--
Hunt-Destination-Number--
Hunt-Dialplan--
Hunt-Direction--
Hunt-Network-Addr--
Hunt-Privacy-Hide-Name--
Hunt-Privacy-Hide-Number--
Hunt-Profile-Created-Timeprofile_created-
Hunt-Profile-Index--
Hunt-RDNIS--
Hunt-Screen-Bit--
Hunt-Source--
Hunt-Transfer-Source--
Hunt-Unique-ID--
Hunt-Username--
Presence-Call-Direction--
variable_DIALSTATUS--
variable_absolute_codec_string--
variable_advertised_media_ip--
variable_bridge_channel--
variable_bridge_hangup_cause--
variable_bridge_uuid--
variable_call_uuid--
variable_current_application_response--
variable_direction--
variable_inherit_codec--
variable_is_outbound--
variable_last_bridge_to--
variable_last_sent_callee_id_name--
variable_last_sent_callee_id_number--
variable_local_media_ip--
variable_local_media_port--
variable_number_alias--
variable_originate_early_media--
variable_originating_leg_uuid--
variable_originator--
variable_originator_codec--
variable_outbound_caller_id_number--
variable_recovery_profile_name--
variable_rtp_use_ssrc--
variable_session_id--
variable_sip_2833_recv_payload--
variable_sip_2833_send_payload--
variable_sip_P-Asserted-Identity--
variable_sip_Privacy--
variable_sip_audio_recv_pt--
variable_sip_cid_type--
variable_sip_cseq--
variable_sip_destination_url--
variable_sip_from_displaysip_from_display'User' element of SIP From: line
variable_sip_from_port--
variable_sip_gateway--
variable_sip_gateway_name--
variable_sip_h_P-Charging-Vector--
variable_sip_local_network_addr--
variable_sip_local_sdp_str--
variable_sip_network_ip--
variable_sip_network_port--
variable_sip_number_alias--
variable_sip_outgoing_contact_uri--
variable_sip_ph_P-Charging-Vector--
variable_sip_profile_name--
variable_sip_recover_contact--
variable_sip_recover_via--
variable_sip_reply_host--
variable_sip_reply_port--
variable_sip_req_port--
variable_sip_to_port--
variable_sip_use_codec_name--
variable_sip_use_codec_ptime--
variable_sip_use_codec_rate--
variable_sip_use_pt--
variable_sip_via_protocol--
variable_switch_m_sdp--
variable_transfer_history--
variable_transfer_source--
variable_uuid--

${variable} vs. $${variable}​

${variables} are channel variables that may be set in the dialplan, application or directory that affect call progress or settings.

$${variables} are expanded when the master freeswitch.xml is compiled. This happens when you start FreeSWITCH, or when reloadxml is called.

The key is that $${variables} are evaluated *once* by the pre-processor, and ${variables} are evaluated fresh each time they are executed (for example when a call connects).

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:

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

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"/>

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}}]

hangup_complete_with_xml​

Variable hangup complete with xml

skip_cdr_causes​

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

transfer_to​

Description needed

Example: Example needed

Hangup Causes​

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 [old wiki].

Usage:

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

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:

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

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:

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

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:

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

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.

ValueIncomingOutgoing
send_byeFS sent BYE to the caller (we hung up)FS sent BYE to the endpoint (we hung up)
recv_byeFS received BYE from the caller (they hung up)FS received BYE from the endpoint (they hung up)
send_refuseFS rejected the call (e.g. 4xx or 5xx)Endpoint rejected the call (e.g. 4xx or 5xx)
send_canceln/aFS aborted the call (we sent CANCEL)

Usage:

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

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:

<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="<a href="sip:480">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>

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:

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

last_bridge_proto_specific_hangup_cause​

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

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:

<action application="set" data="pass_rfc2833=true"/>

dtmf_type​

For inband DTMF, mod_dptools: start_dtmf must be used in the dialplan.

Usage:

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

or,

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

or,

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

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"/>

drop_dtmf_masking_digits​

If drop_dtmf is true play specified tone for every tone received.

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=/etc/freeswitch/Sounds/dtmf_masking_tone" />

drop_dtmf_masking_file​

If drop_dtmf is true play specified file for every tone received.

Usage:

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

sent to the b-leg.

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

<action application="set" 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" />

Media Handling​

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_nameuser defined name for the error condition
number_of_hitsthe number of times the tone must be heard before considering it a fail
tone_detect_frequenciesthe 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:

<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​

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_nameOptional? user-defined name for the error condition
number_of_hitsthe 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_frequenciesthe 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:

<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"/>

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"/>

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:

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

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:

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

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:

<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.

Usage:

<action application="set" data="park\_timeout=30:MEDIA\_TIMEOUT"/>

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:

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

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,

<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" />

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"/>

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:

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

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"/>

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:

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

or,

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

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:

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

api_after_bridge​

Execute an API command after bridge.

Usage:

Paging to PA System via Portaudio (w/ chime before and after announcement) [1]

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>

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_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_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

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"/>

transfer_on_fail​

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

Please note, you will need to set failure_causes variable too.

Usage:

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

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,

<action application="set" data="transfer_on_fail=<hangupcauses> <destination> <dialplan> <context>"/>

or,

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

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).

Related discussion; Link

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"/>

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:

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

or Q.850 cause codes,

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

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:

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

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.

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:

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

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:

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

last_bridge_to​

Variable last bridge to

loopback_bowout_on_execute​

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

Usage:

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

Remarks:

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

loopback_export​

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

Usage:

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:

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

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:

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

Example

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

park_after_bridge​

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

Default: false

Usage:

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

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:

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}

See also:

rtp_jitter_buffer_during_bridge​

Enables/disables the jitter buffer during bridge.

Usage:

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

or,

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

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:

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

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"/>

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}"/>

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}"/>

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"/>

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>

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"/>

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"/>

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"/>

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"/>

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}"/>

last_transferred_conference​

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

Usage:

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

conference_member_id​

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

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.

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:

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

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)

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.

bridge_pre_execute_aleg_data​

Arguments to be used with bridge_pre_execute_aleg_app.

Usage:

Example needed! Please contribute one.

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"/>

bridge_pre_execute_bleg_data​

Arguments to be used with bridge_pre_execute_bleg_app

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"/>

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"/>

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"/>

origination_nested_vars​

Description needed! Please contribute one.

The execute_on family​

The execute_on_xx variables will execute dialplan applications on various conditions. As of April 1, 2011 these variables support a new syntax that allows multiple applications for a single condition. Example:

Example

<action application="set" data="execute_on_answer_1=app1 arg"/>
<action application="set" data="execute_on_answer_2=app2 arg"/>
<action application="set" data="execute_on_answer_3=app3 arg"/>

Note: please be careful not to abuse this feature! There are cases where you could end up blocking the session thread and bad things could happen. If you find that you are trying to do lots and lots of things with these then consider using an execute_extension app and doing your stuff in a dialplan extension.

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>
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}"/>

or, 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​

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"/>
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}"/>

or, 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

Execute a command when the called party rings.

Usage:

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

Execute a command when SIP Reinvite.

Usage:

The api_on family​

The api_on_xx variables will execute FreeSWITCH APIs on various conditions.

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"/>
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}"/>
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_tone_detect

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:

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

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. See below for an example of using fail_on_single_reject with the "|" OR operator.

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"/>

or,

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

or,

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

or to use a list,

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

or 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"/>

intercept_unbridged_only​

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

Default: false

Usage:

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

intercept_unanswered_only​

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

Default: false

Usage:

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

session_in_hangup_hook​

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

Usage:

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

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

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.

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

Informational Tip

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"/>

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"/>

sip_cid_type​

  • Modify how the Caller ID will show up in SIP header of the outbound leg.

Examples:

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

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

Note: for gateways (rather than sip calls on a profile), this will not work. You need caller-id-in-from=true in the gateway settings.

Send Remote-Party-ID (default)

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

Send P-Asserted-Identity

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

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

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

Also the privacy app on the inbound leg controls the remaining contents of the RPID and Privacy headers.

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

effective_sip_cid_in_1xx​

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

Usage:

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

initial_callee_id_name​

Sets the callee id name during the 183. This allows the phone to see the name of the called party prior to the phone being answered.

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)}'"/>

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_diplay_updates which affect the processing of these variables.

Usage:

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

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_diplay_updates which affect the processing of these variables.

Usage:

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

These variables all relate to call recording. See also Misc._Dialplan_Tools_record_session

Audio File Metadata​

RECORD_TITLE​
RECORD_SOFTWARE​
RECORD_ARTIST​
RECORD_COMMENT​
RECORD_DATE​
RECORD_STEREO​

record_fill_cng​

Description need please.

Usage:

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

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:

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

RECORD_DISCARDED​

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

Usage:

N/A

record_post_process_exec_api​

and

record_post_process_exec_app​

These two variables allow the postprocessing of recorded audio. The reason this is required is if the A leg hangs up first in a call, the dialplan stops being processed, and then you aren't able to take action on the file that was recorded. These variables take the form of:

<action application="set" data="record_post_process_exec_api=some_api_app:api_app args" />

Example usage:

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

record_restart_limit_on_dtmf​

Record restart limit on dtmf

record_sample_rate​

Specify the sampling rate of the recorded file.

Usage:

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

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:

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

recording_follow_transfer​

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

Example​

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

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"/>

codec_string​

Sets the base codec string to use.

Usage:

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

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:

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

media_mix_inbound_outbound_codecs​

Set which list of codecs is offered to the B-leg.

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

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 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.

read_codec​

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

write_codec​

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

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.

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

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

sip_renegotiate_codec_on_reinvite​

Allow SDP codec change with re-INVITE

Usage:

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

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

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)

suppress_cng​

Sets a=silenceSupp: off in the sdp to disable silence suppression while making an outbound call.

Usage:

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

ivr_menu_terminator​

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

Usage:

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

detect_speech_result​

The result of play_and_detect_speech.

Usage:

This value is read-only.

rtp_disable_hold​

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

Usage:

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

sip_acl_authed_by​

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

Usage:

None.

sip_acl_token​

Contains the ACL auth token for the current call.

Usage:

N/A

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):

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

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:

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

sip_invite_params​

Description needed! Please contribute one.

Usage:

Example needed! Please contribute one.

sip_invite_domain​

Set the from domain in leg (B).

Usage:

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

sip_from_display​

The 'User' element of the SIP From: line

Usage:

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

This will clear the user part of the SIP message sent to the Leg B call.

Implemented By:

Module NameSource FileLast Revised
mod_sofiasofia_glue.c12566

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

sip_invite_to_params​

Description needed! Please contribute one.

Usage:

Example needed! Please contribute one.

sip_invite_contact_params​

Description needed! Please contribute one.

Usage:

Example needed! Please contribute one.

sip_invite_tel_params​

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

Usage:

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

Result:

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

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:

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

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

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

sip_auto_answer​

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

Usage:

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

sip_auto_answer_detected​

Description needed! Please contribute one.

Usage:

Example needed! Please contribute one.

sip_auto_answer_notify​

Description needed! Please contribute one.

Usage:

Example needed! Please contribute one.

sip_auto_answer_suppress_notify​

Description needed! Please contribute one.

Usage:

Example needed! Please contribute one.

auto_answer_destination​

In an inbound call, the remote side may send a SIP NOTIFY message with the event type "talk", telling Sofia endpoint to answer the call. In this case, the variable "auto_answer_destination" is set to the channel's "destination_number", and the call is transferred to the extension "auto_answer" within current dialplan and context.

Usage:

Example needed! Please contribute one.

See Also:

Link

sip_auto_simplify​

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

Usage:

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

sip_call_id​

SIP header Call-ID.

sip_callee_id_name​

Error

DEPRECATED. Use variable origination_callee_id_name instead

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

Usage:

<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>" />

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

rtp_force_audio_fmtp​

Set the audio fmtp.

Usage:

Please add example if you have one.

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:

<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_record_route​

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>

sip_invite_full_from​

sip_invite_full_to​

sip_handle_full_from​

sip_handle_full_to​

sip_force_full_from​

sip_force_full_to

sip_recover_contact

sip_recover_via

sip_invite_from_uri

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>

sip_ignore_reinvites​

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

Usage:

Don't allow any re-INVITEs once bridged.

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

sip_has_crypto​

rtp_sdes_suites​

rtp_secure_media​

rtp_secure_media _inbound​

rtp_secure_media _outbound​

Usage:

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

NOTICE: 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
  • false - implies forbidden, deprecated
  • true - implies mandatory, deprecated
  • default - accept SAVP inbound if offered.

list of 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

Additionally you can restrict this list to either inbound or outbound by specifying thus:

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=true:AEAD_AES_128_GCM_8

rtp_secure_media_outbound=optional:AEAD_AES_128_GCM_8

See also: $CONFDIR/vars.xml

rtp_secure_media_suites​

Usage: rtp_secure_media_suites=<list of encryption suites>

An alternative, optional variable that specifies a comma-separated list of available encryption suites. If this is specified, then rtp_secure_media need only specify the permission flag without the list of suites.

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:

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

timer_name​

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

Usage:

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

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

<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"/>

deny_refer_requests

If this variable is set to true on either leg of a bridged SIP call, and the other end sends a REFER request, this will be denied by FreeSWITCH.

SDP Manipulation​

When set, these variable change the SDP headers. (not for the faint of heart)

rtp_append_audio_sdp​

This may be used to append audio parameters to the SDP sent to B-leg.

It should/must be set before bridging.

Usage:

<action application="export" data="rtp\_append\_audio\_sdp=a=fmtp:18 annexb=no"/>

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:

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

switch_r_sdp​

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

Usage:

Rewriting SDP:

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

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.

switch_l_sdp​

  • (Local SDP)
  • Although this variable is defined in switch_types.h it is not used anywhere in the FreeSwitch code. (It should be removed for the source and from this page.)

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")

sip_ignore_183nosdp​

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

Usage:

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

verbose_sdp​

The RFCs require an rtpmap for IANA dynamic payload types. An rtpmap with codec name and payload type is not required for well known static payloads - (PCMU, PCMA, G729, etc).

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).

sip_local_sdp_str​

Description goes here

Usage:

Example needed.

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_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.

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"/>
<action application="export" data="sip_enable_soa=false"/>

sdp_secure_savp_only​

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

Usage:

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

fifo_bridged​

Description goes here

Usage:

Example needed

fifo_caller_consumer_import​

Import list of variables from the caller to the consumer.

Usage:

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

fifo_consumer_caller_import​

Import list of variables from the consumer to the caller

Usage:

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

fifo_manual_bridged​

Description goes here

Usage:

Example needed

fifo_position​

Description goes here

Usage:

Example needed

fifo_role​

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

Usage:

None

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.

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

Usage:

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

or,

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

(Note the : separator)

playback_terminators​

playback_terminators=123456789*0# | any | none

Allows you to set which DTMF tones, if pressed during the playback of a file or during play_and_detect_speech, will terminate playback. The default terminator is * (star). Digits not specified as terminators will be collected, but not interrupt playback.

none - undefines the terminator digit, meaning no DTMF tone will be to 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

This variable applies not only to playback of audio files, but also to the record, tone_stream, and say() applications as well.

Examples:

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

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

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

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 ontly interrupt playback without being included in the DTMF string returned.

sound_prefix​

Directory prefix where the sounds lives.

playback_terminator_used​

Contains the digit that the caller used to terminate a playback. Is undef when a new playback is called.

playback_ms​

Contains the number of milliseconds of the length of the audio file just played back. There is also a variable playback_seconds which stores the length in whole seconds.

playback_samples​

Contains the number of samples in the audio file just played back.

playback_last_offset_pos​

Contains the file offset of the file that just played back. This can be used to resume playback at that position at a later time.

playback_sleep_val​

How long to pause after a file is played. Default is 250 milliseconds.

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_delimiter​

When set allows playback of multiple files in sequence, by separating them with the delimiter.

For example, setting playback_delimiter to the following:

<action application="set" data="playback_delimiter=&"/>

Permits the streaming of files foo.wav and bar.wav one right after the other:

<action application="playback" data="foo.wav&bar.wav"/>

sleep_eat_digits​

When set to true, the sleep application will consume DTMFs which will, for example, prevent a caller from exiting out of an IVR. The default behavior is not to eat DTMF digits. NOTE: 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.

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

This variable was added in SVN rev 14102.

playback_timeout_sec​

Set timeout for playback. This is very useful if you want to play short excerpts of a file that could be very long.

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

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

leg_delay_start​

Specifies a wait time in seconds before each leg is called in a forked dial scenario. Can be used in per-leg [], but not in global {} for the dialstring.
Usage:

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

originate_disposition​

This is the originate disposition or hangup cause that is returned. (LEG B)

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

originate_retries​

Number of retries before giving up on originating a call (default is 0).

originate_retry_sleep_ms​

This will set how long FreeSWITCH is going to wait between sending invite messages to the receiving gateway.

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

Using the value of 500 FreeSWITCH will wait 500ms between sending invite messages to the called gateway.

originate_timeout​

Determines how long FreeSWITCH is going to wait for a response from the invite message sent to the gateway.

<action application="set" data="originate_timeout=2"/>

This is very useful if you are using multiple gateways within your dial plan.. so if your dial string is:

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

In this example FreeSWITCH will wait 2 seconds for the 192.168.1.4 to respond to the invite message before trying the next gateway.

originating_leg_uuid​

Shows the uuid of the originating leg on an outbound channel

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:

<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>

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.

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

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.

Usage:

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

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:

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

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

origination_cancel_key​

Implemented in revision 14650 Used with attended transfer function. Allows you to set a DTMF key that will cancel the att_xfer and re-connect to the call on hold.

It'll also cancel a bridge that hasn't been bridged as yet (and thus can't be terminated with a bridge_terminate_key)

Usage:

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

origination_privacy​

Sets privacy profile for caller. Options are "screen", "hide_name", "hide_number".

Usage:

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

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!

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

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.

originator​

Holds the UUID of the channel that originated the call. It's used to notify a parent channel that the state of its 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.

originator_codec​

Sets the codec for calls originated from LEG A (setting the codec for LEG B)

Usage:

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

bypass_media​

When set, the media (RTP) from the originating endpoint is sent directly to the destination endpoint and vice versa. The signaling (SIP) for both endpoints still goes through FreeSWITCH, but the media is point-to-point.

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

See also: Bypass Media Overview

bypass_media_after_bridge​

Same as bypass_media but will handle media for a call until it has reached the answered state (and has processed a few RTP frames.) At this point FreeSWITCH will use a ReInvite to take itself out of the media path. This helps reduce audio latency and take some load off FreeSWITCH. Especially useful for UACs behind Coned NAT as it gives RTP Auto-Adjust enough time to determine the correct ports to avoid one-way audio.

bypass_keep_codec​

By doing export bypass_keep_codec=true to force re-invite with the same codec it was using previously.

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), however it is lighter on the CPU because freeswitch never even parses the packets or processes them in any way, it simply forwards them onwards.

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

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

See also: Proxy Media

rtp_autoflush​

When set to true (default if not present), it will skip timer waits when the socket already has data on read. When set to false, it will always sleep one timer interval. When a packet is too late with this setting, it would be saved for next time in the udp stack and we would place a filler packet into the core to keep it moving that is flagged as CNG so you know there is no audio in it. If you have it set to false, you end up with delay if the other side is sending the audio at a different speed (can be tiny difference but it would build up).

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​

The same as rpt_autoflush, but is set during the bridge.

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 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.

Usage:

Add {disable_rtp_auto_adjust=true} in your dial string.

progress_timeout​

This is the maximum time we will wait before we get media (whether its early media, ringing or answer) e.g. maximum PDD before abandoning the route -- the time from Sending the INVITE to receiving the first ringing response for example: a SIP/2.0 180 Ringing or 183.

Typical fixed lines have a PDD of under 3 seconds, and mobiles of under 8 seconds.

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

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).

Usage

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

See also Early Media

bridge_answer_timeout​

  • Introduced in build 15057

Timeout in seconds how long to tolerate a bridge that is in early media without being answered (can be set on either leg)

Usage

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

See also Early Media

ignore_early_media​

Controls if the call returns on early media or not. Default is false.

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"/>

See also Early Media

ringback​

This addition lets you set artificial ringback on a channel that is waiting for an originated call to be answered. This will specify the audio to play to the A leg on unanswered aka (early media) calls.

The syntax is:

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

Where data is either the full path (important! 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.

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

The following example uses the US ring tone defined in ~/vars.xml:

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

Example #2:

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

Example #3:

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!")

Ringback will not be sent unless the variable is set *and* there's a ringing indication from the party that is being called.

If you want to use fake ringback via the ringback variable and you want it to happen instantly, you can also do:

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

And FS will send ringback without waiting for a ringing indication. Note that this may cause undesirable effects, like ringing for 3 seconds that turns into a busy signal.

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"/>

transfer_ringback​

This is the sound that will play if a call has already been answered, and it is then transferred to another endpoint.

Usage:

<action application="set" data="transfer_ringback=[data]"/>

Where data is either the full path to an audio file or teletone generation markup language.

The following example uses the French ringtone defined in ~/vars.xml:

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

rtp_disable_hold​

This is the channel variable for disable-hold.

Usage:

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

rtp_negotiate_near_match​

Setting this to true will cause FreeSWITCH to consider codec PTIME on inbound codec negotiation as well. If you receive a call with PCMU@30i and only have PCMU@20i on the list, it will prefer anything else that matches codec and ptime by default. This changes behavior and matches PCMU@30i despite the PTIME mismatch. This is only valid if you have other matching codecs on the endpoints lists, of course.

Usage:

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

rtcp_packet_count​

Contains number of RTCP packets collected during the call.

Usage:

N/A

rtcp_octet_count​

Contains number of RTCP octets collected during the call.

Usage:

N/A

rtcp_audio_interval_msec​

Channel variable to set the interval in msec between each RTCP SR paquet.
WARNING: setting the rtp_timer_name to none will create all sorts of undesired side effects with RTCP ( paquets not sent, RTCP autoadjust failing, etc... ).

rtcp_mux​

Channel variable to force rtcp-mux to be used on the outgoing INVITE.

rtp_assume_rtcp​

Channel variable to force freeswitch to assume rtcp is at rtp port + 1 all the time.

Camping is used when bridging a call. It will basically park (aka camp) your channel to music on hold while it attempts to bridge the call.

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_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_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"/>

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"/>

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_fallback_dialplan​

Controls camping during bridge app (testing needed)

Usage:

<action application="set" data="campon"/>
<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_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_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_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"/>

Answer confirmation variables​

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.

<action application="set" data="group_confirm_file=/usr/local/freeswitch/sounds/take_call_question.wav" />
<action application="set" data="group_confirm_key=1" />

Through 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.

group_confirm_key​

see group_confirm_file

group_confirm_cancel_timeout​

If set, cancels leg_timeout after the call is answered. originate_timeout still remains active.

More detail:

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

Normally, a timeout on the leg will apply to phases 1 and 2.

However, if you do

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

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

See Also:

voicemail_alternate_greet_id​

voicemail_greeting_number​

vm_message_ext​

vm_cc​

skip_greeting​

skip_instructions​

voicemail_authorized​

Events​

fire_asr_events​

If set, fire an event when speech is detected.

base_dir​

This variable is used to leverage the installation directory in order to avoid hard-coding it the xml.

Usage:

<action application="system" data="$${base_dir}/scripts/my_script.sh"/>

openzap_span_number​

openzap span number

openzap_chan_number​

openzap channel number

Comments:​

This page needs the sip_*_media variables to be updated to rtp_*_media variables. Posted by boteman at Jun 21, 2014 19:07
An alternative, optional variable that specifies a comma-separated list of available encryption suites. If this is specified, then rtp_secure_media need only specify the permission flag without the list of suites.need to change comma-separated to colon-separated and add examplertp_secure_media_suites=AES_CM_128_HMAC_SHA1_32:AES_CM_128_HMAC_SHA1_80 Posted by chusov at Feb 02, 2015 01:19
Also: sip_append_audio_sdp should be rtp_append_audio_sdp Posted by stephalnet at Nov 30, 2015 08:08
Found this in old wiki for playback_terminators variable: Git Committed 38b3f43d: add prefix chars to playback_terminators '+' means include the term in the stringand x means include the char and return SWITCH_STATUS_RESTARTeg #+* only includes the * if you type it but not the # Posted by tomasb at Dec 01, 2015 14:01
same for sip_force_audio_fmtp . It should be rtp_force_audio_fmtp Posted by dragos_oancea at Sep 23, 2016 14:01
Please add channel variable sip_reason Posted by livem at Nov 08, 2016 06:25
I have added you to the Confluence-editors group so you can add that variable, but what is the explanation of sip_reason?Thank you. Posted by boteman at Nov 14, 2016 14:15
Thank you very much. ^_^The sip header ReasonThe Reason Header Field for SIP Initially, the Reason header field defined here appears to be most useful for BYE and CANCEL requests, but it can appear in any request within a dialog, in any CANCEL request and in any response whose status code explicitly allows the presence of this header field.from Here Posted by livem at Nov 18, 2016 05:41
also at wiki we have about playback_terminators ---- keyword 'any' will terminate playback when any key (1234567890*#) is pressed. Posted by Pechen at Jan 09, 2017 06:26
Not all execute_o_xxx described. There are execute_on_originate very useful variable situated Posted by bordmi at Feb 10, 2017 07:32