FreeSwitch Channel variable 对照表

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 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 IDpresentation Restriction] (either with anonymous method or Privacy:id method)
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_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	.
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	.
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_disable_hold	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.
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.