Channel Variables

About

Channel variables are used to manipulate dialplan execution, to control call progress, and to provide options to applications. They play a pervasive role, as FreeSWITCH™ frequently consults channel variables as a way to customize processing prior to a channel's creation, during call progress, and after the channel hangs up. 

We rely on variable expansion to create flexible, reusable dialplans:

  • $${variable} is expanded once when FreeSWITCH™ first parses the configuration on startup or after invoking reloadxml. It is suitable for variables that do not change, such as the domain of a single-tenant FreeSWITCH™ server. That is why $${domain} is referenced so frequently in the vanilla dialplan examples
  • ${variable} is expanded 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 are set, appropriately enough, with the set application:

Reading channel variables requires the ${} syntax:

Channel variables used to be global to the session. As of b2c3199f, it is possible to set variables that only exist within a single application execution and any subsequent applications under it. For example, applications can use scoped variables for named input params:

The variable assignment syntax for dial strings differs depending on which scope they should apply to:

  • {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 a dial string. It will set the same variables on every channel, including all thos in an enterprise bridging/originate.
  • [foo=bar] goes before each individual dial string and will set the variable values specified for only this channel.

Examples

Set foo variable for all channels implemented and chan=1 will only be set for blah, while chan=2 will only be set for blah2:

Set multiple variables by delimiting with commas:

To have variables in [] override variables in {}, set local_var_clobber=true inside {}. You must also set local_var_clobber=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):

Commas are the default delimiter inside variable assignment tags. In some cases (like in absolute_codec_string), we may need to define variables whose values contain literal commas that should not be interpreted as delimiters. 

We can redefine the delimiter for a variable using ^^ followed by the desired delimiter:

To set absolute_codec_string=PCMA@8000h@20i@64000b,PCMU@8000h@20i@64000b,G729@8000h@20i@8000b in a dial string:

This approach does not work when setting sip_h_*, sip_rh_*, and sip_ph headers. To pass a comma into the contents of a private header, escape the comma with a backslash:

Variables from one call leg (A) can be exported to the other call leg (B) by using the export_vars variable. Its value is a comma separated list of variables that should propagate across calls.

To set a variable on the A-leg and add it to the export list, use the export application:

Channel variables can be used in conditions, refer to XML Dialplan Conditions for more information.

Some channel variables may not be set during the dialplan parsing phrase. See Inline Actions.

We are not constrained to the channel variables that FreeSWITCH™, its modules, and applications define. It is possible to set any number of unique channel variables for any purpose. They can also be logged in CDR

The set application can be used to set any channel variable:

In a command issued via mod_xml_rpc or mod_event_socket:

Values with spaces must be enclosed by quotes:

Channel variables can be manipulated for varied results. For example, a channel variable could be trimmed to get the first three digits of a phone number. Manipulating Channel Variables discusses this in detail.

Consider this example:

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

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

In addition to the dialplan, channel variables can be set in other environments as well.

In a  FreeSWITCH™ module, written in C:

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

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

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

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:

With the new Session object s:

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 name channel variable name Description
Channel-State state Current state of the call
Channel-State-Number state_number Integer
Channel-Name channel_name Channel name
Unique-ID uuid uuid of this channel's call leg
Call-Direction direction Inbound or Outbound
Answer-State state -
Channel-Read-Codec-Name read_codec the read codec variable mean the source codec
Channel-Read-Codec-Rate read_rate the source rate
Channel-Write-Codec-Name write_codec the destination codec same to write_codec if not transcoded
Channel-Write-Codec-Rate write_rate destination rate same to read rate if not transcoded
Caller-Username username .
Caller-Dialplan dialplan user dialplan like xml, lua, enum, lcr
Caller-Caller-ID-Name caller_id_name .
Caller-Caller-ID-Number caller_id_number .
Caller-ANI ani ANI of caller, frequently the same as caller ID number
Caller-ANI-II aniii ANI II Digits (OLI - Originating Line Information), if available. Refer to: http://www.nanpa.com/number_resource_info/ani_ii_digits.html
Caller-Network-Addr network_addr IP address of calling party
Caller-Destination-Number destination_number Destination (dialed) number
Caller-Unique-ID uuid This channel's uuid
Caller-Source source Source module, i.e. mod_sofia, mod_openzap, etc.
Caller-Context context Dialplan context
Caller-RDNIS rdnis Redirected DNIS info. See mod_dptools: transfer application
Caller-Channel-Name channel_name .
Caller-Profile-Index profile_index .
Caller-Channel-Created-Time created_time .
Caller-Channel-Answered-Time answered_time .
Caller-Channel-Hangup-Time hangup_time .
Caller-Channel-Transfer-Time transfer_time .
Caller-Screen-Bit screen_bit .
Caller-Privacy-Hide-Name privacy_hide_name .
Caller-Privacy-Hide-Number privacy_hide_number This 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_name Sets 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_ip sip_received_ip .
variable_sip_received_port sip_received_port .
variable_sip_authorized sip_authorized .
variable_sip_mailbox sip_mailbox .
variable_sip_auth_username sip_auth_username .
variable_sip_auth_realm sip_auth_realm .
variable_mailbox mailbox .
variable_user_name user_name .
variable_domain_name domain_name .
variable_record_stereo record_stereo .
variable_accountcode accountcode Accountcode 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_context user_context .
variable_effective_caller_id_name effective_caller_id_name .
variable_effective_caller_id_number effective_caller_id_number .
variable_caller_domain caller_domain .
variable_sip_from_user sip_from_user .
variable_sip_from_uri sip_from_uri .
variable_sip_from_host sip_from_host .
variable_sip_from_user_stripped sip_from_user_stripped .
variable_sip_from_tag sip_from_tag .
variable_sofia_profile_name sofia_profile_name .
variable_sofia_profile_domain_name sofia_profile_domain_name .
variable_sip_full_route sip_full_route The complete contents of the Route: header.
variable_sip_full_via sip_full_via The complete contents of the Via: header.
variable_sip_full_from sip_full_from The complete contents of the From: header.
variable_sip_full_to sip_full_to The complete contents of the To: header.
variable_sip_req_params sip_req_params .
variable_sip_req_user sip_req_user .
variable_sip_req_uri sip_req_uri .
variable_sip_req_host sip_req_host .
variable_sip_to_params sip_to_params .
variable_sip_to_tag sip_to_tag .
variable_sip_to_user sip_to_user .
variable_sip_to_uri sip_to_uri .
variable_sip_to_host sip_to_host .
variable_sip_contact_params sip_contact_params .
variable_sip_contact_user sip_contact_user .
variable_sip_contact_port sip_contact_port .
variable_sip_contact_uri sip_contact_uri .
variable_sip_contact_host sip_contact_host .
variable_sip_invite_domain sip_invite_domain .
variable_channel_name channel_name .
variable_sip_call_id sip_call_id SIP header Call-ID
variable_sip_user_agent sip_user_agent .
variable_sip_via_host sip_via_host .
variable_sip_via_port sip_via_port .
variable_sip_via_rport sip_via_rport .
variable_presence_id presence_id .
variable_sip_h_P-Key-Flags sip_h_p-key-flags This will contain the optional P-Key-Flags header(s) that may be received from calling endpoint.
variable_switch_r_sdp switch_r_sdp The whole SDP received from calling endpoint.
variable_remote_media_ip remote_media_ip .
variable_remote_media_port remote_media_port .
variable_write_codec write_codec .
variable_write_rate write_rate .
variable_endpoint_disposition endpoint_disposition .
variable_dialed_ext dialed_ext .
variable_transfer_ringback transfer_ringback .
variable_call_timeout call_timeout .
variable_hangup_after_bridge hangup_after_bridge .
variable_continue_on_fail continue_on_fail .
variable_dialed_user dialed_user .
variable_dialed_domain dialed_domain .
variable_sip_redirect_contact_user_0 sip_redirect_contact_user_0.
variable_sip_redirect_contact_host_0 sip_redirect_contact_host_0.
variable_sip_h_Referred-By sip_h_referred-by .
variable_sip_refer_to sip_refer_to The full SIP URI received from a SIP Refer-To: response
variable_max_forwards max_forwards .
variable_originate_disposition originate_disposition .
variable_read_codec read_codec .
variable_read_rate read_rate .
variable_open open .
variable_use_profile use_profile .
variable_current_application current_application .
variable_ep_codec_string ep_codec_string This 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_hold rtp_disable_hold This variable when set will disable the hold feature of the phone.
variable_sip_acl_authed_by sip_acl_authed_by This variable holds what ACL rule allowed the call.
variable_curl_response_data curl_response_data This variable stores the output from the last curl made.
variable_drop_dtmf drop_dtmf Set on a channel to drop DTMF events on the way out.
variable_drop_dtmf_masking_file drop_dtmf_masking_file If drop_dtmf is true play specified file for every tone received.
variable_drop_dtmf_masking_digits drop_dtmf_masking_digits If drop_dtmf is true play specified tone for every tone received.
sip_codec_negotiation sip_codec_negotiation sip_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-Time profile_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-Time profile_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_display sip_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 - -
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  • freeswitch/dp/channel_variables.txt
  • Последние изменения: 2018/10/10