freeswitch 通道变量

1  Introduction

• 1.1 Channel
  Variables in the XML Dialplan
  • 1.1.1 Scoped Variables
• 1.2 Channel
  Variables in Dial strings
  • 1.2.1 Handling
    Variables With Commas
• 1.3 Exporting
  Channel Variables in Bridge Operations
• 1.4 Using
  Channel Variables in Dialplan Condition Statements
• 1.5 Custom Channel
  Variables
• 1.6 Channel Variable
  Manipulation

2  Info
Application Variable Names (variable_xxxx) 3  ${variable}
vs. $${variable} 4  CDR related

• 4.1 process_cdr
• 4.2 accountcode
• 4.3 hangup_complete_with_xml

5  Hangup Causes

• 5.1 bridge_hangup_cause
• 5.2 disable_q850_reason
• 5.3 hangup_cause
• 5.4 hangup_cause_q850
• 5.5 sip_hangup_disposition
• 5.6 proto_specific_hangup_cause

6  DTMF Related

• 6.1 pass_rfc2833
• 6.2 dtmf_type

7  Media Handling

• 7.1 monitor_early_media_fail
• 7.2 monitor_early_media_ring

8  Timeout Related

• 8.1 call_timeout
• 8.2 leg_timeout
• 8.3 originate_continue_on_timeout
• 8.4 park_timeout

9  Music On Hold Related

• 9.1 hold_music
• 9.2 temp_hold_music

10  Locale Related

• 10.1 default_language
• 10.2 timezone

11  Bridge Related

• 11.1 api_after_bridge
• 11.2 auto_hunt
• 11.3 bridge_early_media
• 11.4 bridge_terminate_key
• 11.5 continue_on_fail
• 11.6 transfer_on_fail
• 11.7 enable_file_write_buffering
• 11.8 failure_causes
• 11.9 force_transfer_context
• 11.10 force_transfer_dialplan
• 11.11 hangup_after_bridge
• 11.12 hold_hangup_xfer_exten
• 11.13 last_bridge_to
• 11.14 loopback_bowout_on_execute
• 11.15 outbound_redirect_fatal
• 11.16 originate_timeout
• 11.17 park_after_bridge
• 11.18 signal_bond
• 11.19 sip_jitter_buffer_during_bridge
• 11.20 uuid_bridge_continue_on_cancel

12  Conference Related

• 12.1 conference_auto_outcall_announce
• 12.2 conference_auto_outcall_caller_id_name
• 12.3 conference_auto_outcall_caller_id_number
• 12.4 conference_auto_outcall_flags
• 12.5 conference_auto_outcall_prefix
• 12.6 conference_auto_outcall_timeout
• 12.7 conference_auto_outcall_maxwait
• 12.8 conference_controls
• 12.9 conference_enter_sound
• 12.10 conference_last_matching_digits
• 12.11 last_transferred_conference
• 12.12 conference_member_id
• 12.13 conference_uuid
• 12.14 hangup_after_conference

13  Code Execution Related

• 13.1 api_hangup_hook
• 13.2 bridge_pre_execute_aleg_app
• 13.3 bridge_pre_execute_aleg_data
• 13.4 bridge_pre_execute_bleg_app
• 13.5 bridge_pre_execute_bleg_data
• 13.6 exec_after_bridge_app
• 13.7 exec_after_bridge_arg
• 13.8 The execute_on family
  • 13.8.1 execute_on_answer
  • 13.8.2 execute_on_media
  • 13.8.3 execute_on_preanswer
  • 13.8.4 execute_on_ring
  • 13.8.5 execute_on_sip_reinvite
• 13.9 failed_xml_cdr_prefix
• 13.10 fail_on_single_reject
• 13.11 intercept_unbridged_only
• 13.12 intercept_unanswered_only
• 13.13 session_in_hangup_hook

14  Caller ID Related

• 14.1 caller_id_name
• 14.2 caller_id_number
• 14.3 effective_caller_id_name
• 14.4 effective_caller_id_number
• 14.5 sip_cid_type
• 14.6 effective_sip_cid_in_1xx

15  Callee ID Related

• 15.1 sip_callee_id_name
• 15.2 sip_callee_id_number

16  Call Recording Related

• 16.1 Audio File Metadata
  • 16.1.1 RECORD_TITLE
  • 16.1.2 RECORD_COPYRIGHT
  • 16.1.3 RECORD_SOFTWARE
  • 16.1.4 RECORD_ARTIST
  • 16.1.5 RECORD_COMMENT
  • 16.1.6 RECORD_DATE
  • 16.1.7 RECORD_STEREO
• 16.2 record_fill_cng
• 16.3 RECORD_HANGUP_ON_ERROR
• 16.4 RECORD_DISCARDED
• 16.5 record_post_process_exec_api
• 16.6 record_post_process_exec_app
• 16.7 record_restart_limit_on_dtmf
• 16.8 record_sample_rate
• 16.9 record_waste_resources

17  Codec Related

• 17.1 absolute_codec_string
• 17.2 codec_string
• 17.3 inherit_codec
• 17.4 read_codec
• 17.5 write_codec
• 17.6 passthru_ptime_mismatch
• 17.7 sip_renegotiate_codec_on_reinvite
• 17.8 conference_enforce_security
• 17.9 suppress-cng

18  IVR related

• 18.1 ivr_menu_terminator
• 18.2 detect_speech_result

19  SIP related variables

• 19.1 disable_hold
• 19.2 sip_acl_authed_by
• 19.3 sip_acl_token
• 19.4 sip_copy_multipart
• 19.5 sip_invite_params
• 19.6 sip_invite_domain
• 19.7 sip_invite_from_params
• 19.8 sip_invite_to_params
• 19.9 sip_invite_contact_params
• 19.10 sip_network_destination
• 19.11 sip_auth_username
• 19.12 sip_auth_password
• 19.13 sip_auto_simplify
• 19.14 sip_callee_id_name
• 19.15 sip_callee_id_number
• 19.16 sip_force_audio_fmtp
• 19.17 sip_invite_req_uri
• 19.18 sip_invite_from_uri
• 19.19 sip_invite_to_uri
• 19.20 sip_ignore_reinvites
• 19.21 sip_has_crypto
• 19.22 sip_secure_media
• 19.23 timer_name
• 19.24 ignore_display_updates
• 19.25 deny_refer_requests

20  SDP Manipulation

• 20.1 sdp_m_per_ptime
• 20.2 switch_r_sdp
• 20.3 switch_l_sdp
• 20.4 switch_m_sdp
• 20.5 sip_append_audio_sdp
• 20.6 sip_ignore_183nosdp
• 20.7 verbose_sdp
• 20.8 sip_local_sdp_str
• 20.9 sip_recovery_break_rfc
• 20.10 sip_enable_soa

21  FIFO related variables

• 21.1 fifo_bridged
• 21.2 fifo_caller_consumer_import
• 21.3 fifo_consumer_caller_import
• 21.4 fifo_manual_bridged
• 21.5 fifo_position
• 21.6 fifo_role
• 21.7 transfer_after_bridge

22  Playback related
variables

• 22.1 playback_terminators
• 22.2 sound_prefix
• 22.3 playback_terminator_used
• 22.4 playback_ms
• 22.5 playback_samples
• 22.6 playback_sleep_val
• 22.7 playback_delimiter
• 22.8 sleep_eat_digits
• 22.9 playback_timeout_sec

23  Originate related
variables

• 23.1 execute_on_originate
• 23.2 leg_delay_start
• 23.3 originate_disposition
• 23.4 originate_retries
• 23.5 originate_retry_sleep_ms
• 23.6 originate_timeout
• 23.7 originating_leg_uuid
• 23.8 origination_channel_name
• 23.9 origination_caller_id_name
• 23.10 origination_caller_id_number
• 23.11 origination_cancel_key
• 23.12 origination_privacy
• 23.13 origination_uuid
• 23.14 originator
• 23.15 originator_codec

24  RTP/media
related variables

• 24.1 bypass_media
• 24.2 bypass_media_after_bridge
• 24.3 proxy_media
• 24.4 rtp_autoflush
• 24.5 rtp_autoflush_during_bridge
• 24.6 disable_rtp_auto_adjust
• 24.7 progress_timeout
• 24.8 bridge_answer_timeout
• 24.9 ignore_early_media
• 24.10 ringback
• 24.11 instant_ringback
• 24.12 transfer_ringback