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. |
FreeSwitch Channel variable 对照表
最新推荐文章于 2023-11-15 17:54:16 发布