我建了一个 Freeswitch 内核研究 交流群， 45211986， 欢迎加入, 另外，提供基于SIP的通信服务器及客户端解决方案。
freeswitch有大量通道变量挂载在每一路通话中，这些变量的值控制了本路通话的大多行为。
下面根据类型分类的通道变量，慢慢翻译。。。
- 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
- 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
- 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
- 12.1 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
- 13.1 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
- 14.1 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
- 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
- 17.1 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
- 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
- 20.1 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
- 21.1 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
- 22.1 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
- 23.1 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
- 24.1 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
- 24.13 disable_hold
- 26.1 campon
- 26.2 campon_retries
- 26.3 campon_timeout
- 26.4 campon_sleep
- 26.5 campon_fallback_exten
- 26.6 campon_fallback_dialplan
- 26.7 campon_fallback_context
- 26.8 campon_hold_music
- 26.9 campon_stop_key
- 26.10 campon_announce_sound
- 28.1 voicemail_alternate_greet_id
- 28.2 voicemail_greeting_number
- 28.3 vm_message_ext
- 28.4 vm_cc
- 28.5 skip_greeting
- 28.6 skip_instructions
- 28.7 voicemail_authorized
