__Variables
About
This page is a reference to all FreeSWITCH variables.
This page parallels the __Channel Variables page. Consult both while the pages are being reconciled.
Click here to expand Table of Contents
- 1 Variable References
- 2 absolute_codec_string
- 3 accountcode
- 4 acl_token
- 5 alert_info
- 6 answer_epoch
- 7 answer_stamp
- 8 answer_uepoch
- 9 answermsec
- 10 answersec
- 11 answerusec
- 12 api_after_bridge
- 13 api_hangup_hook
- 14 api_on_answer
- 15 api_on_media
- 16 api_on_startup
- 17 asr_intercept_dtmf
- 18 auto_answer_destination
- 19 auto_hunt
- 20 billmsec
- 21 billsec
- 22 billusec
- 23 bind_meta_key
- 24 bridge_answer_timeout
- 25 bridge_channel
- 26 bridge_early_media
- 27 bridge_epoch
- 28 bridge_filter_dtmf
- 29 bridge_generate_comfort_noise
- 30 bridge_hangup_cause
- 31 bridge_pre_execute_aleg_app
- 32 bridge_pre_execute_aleg_data
- 33 bridge_pre_execute_bleg_app
- 34 bridge_pre_execute_bleg_data
- 35
- 36 bridge_stamp
- 37 bridge_terminate_key
- 38 bridge_to
- 39 bridge_uepoch
- 40 bridge_uuid
- 41 bypass_media
- 42 bypass_media_after_bridge
- 43 cache_speech_handles
- 44 call_clientcode
- 45 call_timeout
- 46 caller_id
- 47 caller_id_name
- 48 caller_id_number
- 49 campon
- 50 campon_announce_sound
- 51 campon_fallback_context
- 52 campon_fallback_dialplan
- 53 campon_hold_music
- 54 campon_fallback_exten
- 55 campon_retries
- 56 campon_sleep
- 57 campon_stop_key
- 58 campon_timeout
- 59 cdr_csv_base
- 60 CHANNEL
- 61 channel_name
- 62 codec_string
- 63 conference_auto_outcall_announce
- 64 conference_auto_outcall_caller_id_name
- 65 conference_auto_outcall_caller_id_number
- 66 conference_auto_outcall_flags
- 67 conference_auto_outcall_maxwait
- 68 conference_auto_outcall_prefix
- 69 conference_auto_outcall_profile
- 70 conference_auto_outcall_timeout
- 71 conference_controls
- 72 conference_enforce_security
- 73 conference_enter_sound
- 74 conference_exit_sound
- 75 conference_last_matching_digits
- 76 conference_member_id
- 77 conference_moderator
- 78 conference_name
- 79 conference_recording
- 80 conference_uuid
- 81 continue_on_fail
- 82 copy_xml_cdr
- 83 created_time
- 84 current_application
- 85 current_application_data
- 86 current_application_response
- 87 default_language
- 88 destination_number
- 89 detect_speech_result
- 90 dialed_domain
- 91 dialed_group
- 92 dialed_user
- 93 digits_dialed
- 94 direction
- 95 disable_app_log
- 96 rtp_disable_hold
- 97
- 98 disable_q850_reason
- 99 disable_radius_start
- 100 disable_radius_stop
- 101 disable_rtp_auto_adjust
- 102 DISPLACE_HANGUP_ON_ERROR
- 103 dl_cid_msg
- 104 dl_from_host
- 105 dl_from_user
- 106 dl_host
- 107 dl_user
- 108 domain_name
- 109 drop_dtmf
- 110 dtmf_type
- 111 dtmf_verbose
- 112 duration
- 113 easy_acctcode
- 114 easy_destnum
- 115 easy_dialstring
- 116 easy_group
- 117 easy_limit
- 118 eavesdrop_annnounce_macro
- 119 eavesdrop_announce_id
- 120 eavesdrop_group
- 121 eavesdrop_indicate_failed
- 122 eavesdrop_indicate_idle
- 123 eavesdrop_indicate_new
- 124 eavesdrop_require_group
- 125 effective_caller_id_name
- 126 effective_caller_id_number
- 127 enable_file_write_buffering
- 128 enable_heartbeat_events
- 129 end_epoch
- 130 end_stamp
- 131 end_uepoch
- 132 ep_codec_prefer_sdp
- 133 ep_codec_string
- 134 exec_after_bridge_app
- 135 exec_after_bridge_arg
- 136 execute_on_answer
- 137 execute_on_fax_detect
- 138 execute_on_fax_failure
- 139 execute_on_fax_result
- 140 execute_on_fax_success
- 141 execute_on_media
- 142 execute_on_media_timeout
- 143
- 144 execute_on_originate
- 145 execute_on_pre_answer
- 146 execute_on_ring
- 147 execute_on_sip_extra_headers
- 148 execute_on_sip_reinvite
- 149 export_vars
- 150 EXTEN
- 151 fail_on_single_reject
- 152 failed_xml_cdr_prefix
- 153 failure_causes
- 154 fax_bad_rows
- 155 fax_disable_v17
- 156 fax_document_total_pages
- 157 fax_ecm_requested
- 158 fax_ecm_used
- 159 fax_end_page
- 160 fax_filename
- 161 fax_force_caller
- 162 fax_ident
- 163 fax_image_resolution
- 164 fax_image_size
- 165 fax_local_station_id
- 166 fax_prefix
- 167 fax_remote_station_id
- 168 fax_result_code
- 169 fax_result_text
- 170 fax_success
- 171 fax_transfer_rate
- 172 fax_use_ecm
- 173 fax_v17_disabled
- 174 fax_verbose
- 175 fifo_announce
- 176 fifo_bridged
- 177 fifo_caller_consumer_import
- 178 fifo_caller_exit_key
- 179 fifo_chime_freq
- 180 fifo_chime_list
- 181 fifo_consumer_caller_import
- 182 fifo_consumer_exit_key
- 183 fifo_consumer_id
- 184 fifo_consumer_wrapup_key
- 185 fifo_consumer_wrapup_sound
- 186 fifo_consumer_wrapup_time
- 187 fifo_hold_music
- 188 fifo_manual_bridged
- 189 fifo_member_wait
- 190 fifo_orbit_announce
- 191 fifo_orbit_context
- 192 fifo_orbit_exten
- 193 fifo_outbound_uuid
- 194 fifo_override_announce
- 195 fifo_pop_order
- 196 fifo_position
- 197 fifo_record_template
- 198 fifo_role
- 199 fifo_timestamp
- 200 force_transfer_context
- 201 force_transfer_dialplan
- 202 group_confirm_cancel_timeout
- 203 group_confirm_file
- 204 group_confirm_key
- 205 hangup_after_bridge
- 206 hangup_after_conference
- 207 hangup_cause
- 208 hangup_cause_q850
- 209 hold_events
- 210 hold_hangup_xfer_exten
- 211 hold_music
- 212 ignore_display_updates
- 213 ignore_early_media
- 214 import
- 215 inherit_codec
- 216 intercept_unanswered_only
- 217 intercept_unbridged_only
- 218 ivr_menu_status
- 219 ivr_menu_terminator
- 220 jitterbuffer_msec
- 221 last_bridge_hangup_cause
- 222 last_bridge_proto_specific_hangup_cause
- 223 last_matching_digits
- 224 last_transferred_conference
- 225 leg_delay_start
- 226 leg_progress_timeout
- 227 leg_timeout
- 228 loopback_bowout_on_execute
- 229 loopback_export
- 230 media_bug_answer_req
- 231 min_dup_digit_spacing_ms
- 232 monitor_early_media_fail
- 233 monitor_early_media_ring
- 234 monitor_early_media_ring_total
- 235 monitor_fail_dispo
- 236 no_throttle_limits
- 237 originate_delay_start
- 238 originate_disposition
- 239 originate_timeout
- 240 originating_leg_uuid
- 241 origination_callee_id_name
- 242 origination_callee_id_number
- 243 origination_caller_id_name
- 244 origination_caller_id_number
- 245 origination_channel_name
- 246 origination_privacy
- 247 originator_codec
- 248 outbound_redirect_fatal
- 249 park_after_bridge
- 250 park_timeout
- 251 pass_rfc2833
- 252 passthru_ptime_mismatch
- 253 process_cdr
- 254 profile_created_time
- 255 progress_time
- 256 proto_specific_hangup_cause
- 257 read_codec
- 258 RECORD_APPEND
- 259 RECORD_ARTIST
- 260 RECORD_BRIDGE_REQ
- 261 RECORD_COMMENT
- 262 RECORD_COPYRIGHT
- 263 RECORD_DATE
- 264 RECORD_DISCARDED
- 265 record_fill_cng
- 266 RECORD_HANGUP_ON_ERROR
- 267 RECORD_MIN_SEC
- 268 record_ms
- 269 record_sample_rate
- 270 RECORD_READ_ONLY
- 271 record_restart_limit_on_dtmf
- 272 record_sample_rate
- 273 record_samples
- 274 RECORD_SOFTWARE
- 275 RECORD_STEREO
- 276 RECORD_STEREO_SWAP
- 277 RECORD_TITLE
- 278 record_waste_resources
- 279 RECORD_WRITE_ONLY
- 280 recording_follow_transfer
- 281 rtcp_octet_count
- 282 rtcp_packet_count
- 283 rtp_disable_vad_in
- 284 rtp_disable_vad_out
- 285 rtp_enable_vad_in
- 286 rtp_enable_vad_out
- 287 sdp_m_per_ptime
- 288 sdp_secure_savp_only
- 289 send_silence_when_idle
- 290 session_in_hangup_hook
- 291 signal_bond
- 292 sip_acl_authed_by
- 293 sip_acl_token
- 294 sip_auth_password
- 295 sip_auth_username
- 296 sip_authorized
- 297 sip_auto_simplify
- 298 sip_callee_id_name
- 299 sip_callee_id_number
- 300 sip_callee_id_number
- 301 sip_cid_in_1xx
- 302 sip_cid_type
- 303 sip_copy_multipart
- 304 sip_enable_soa
- 305 sip_execute_on_image
- 306 sip_force_audio_fmtp
- 307 sip_from_display
- 308 sip_hangup_disposition
- 309 sip_ignore_183nosdp
- 310 sip_ignore_reinvites
- 311 sip_invite_contact_params
- 312 sip_invite_domain
- 313 sip_invite_from_params
- 314 ip_invite_params
- 315 sip_invite_req_uri
- 316 sip_invite_route_uri
- 317 sip_invite_tel_params
- 318 sip_invite_to_params
- 319 sip_invite_to_uri
- 320 rtp_jitter_buffer_during_bridge
- 321 sip_jitter_buffer_plc
- 322 sip_local_sdp_str
- 323 sip_mirror_remote_audio_codec_payload
- 324 sip_network_destination
- 325 sip_recovery_break_rfc
- 326 sip_renegotiate_codec_on_reinvite
- 327 sip_wait_for_aleg_ack
- 328 skeleton
- 329 skip_cdr_causes
- 330 spandsp_dtmf_rx_filter_dialtone
- 331 spandsp_dtmf_rx_reverse_twist
- 332 spandsp_dtmf_rx_threshold
- 333 spandsp_dtmf_rx_twist
- 334 suppress_cng
- 335 switch_m_sdp
- 336 switch_r_sdp
- 337 temp_hold_music
- 338 timer_name
- 339 timezone
- 340 tod_tz_offset
- 341 transfer_after_bridge
- 342 transfer_on_fail
- 343 transfer_to
- 344 uuid_bridge_continue_on_cancel
- 345 verbose_sdp
- 346 write_codec
- 347 data
- 348 endpoint_disposition
- 349 fax_document_transferred_pages
- 350 fax_header
- 351 fax_start_page
- 352 fifo_music
- 353 fifo_priority
- 354 fifo_serviced_by
- 355 fifo_serviced_uuid
- 356 fifo_status
- 357 fifo_strategy
- 358 fifo_target
- 359 fire_asr_events
- 360 flow_billmsec
- 361 flow_billsec
- 362 flow_billusec
- 363 funny_stun
- 364 group_context
- 365 has_t38
- 366 holding_uuid
- 367 id
- 368 inbound_dialplan
- 369 instant_ringback
- 370 is_outbound
- 371 language
- 372 last_app
- 373 last_arg
- 374 last_dtmf_duration
- 375 last_file_position
- 376 lcr_auto_route
- 377 lcr_route_count
- 378 left_hanging_extension
- 379 limit_id
- 380 limit_max
- 381 limit_rate
- 382 limit_realm
- 383 limit_usage
- 384 local_media_ip
- 385 local_media_port
- 386 local_video_ip
- 387 local_video_port
- 388 loopback_leg
- 389 max_forwards
- 390 mduration
- 391 media_webrtc
- 392 memory_debug
- 393 monitor_ring_dispo
- 394 myid
- 395 NDLB_support_asterisk_missing_srtp_auth
- 396 new_name
- 397 nonexistantvar
- 398 original_caller_id_name
- 399 original_caller_id_number
- 400 original_destination_number
- 401 originate_continue_on_timeout
- 402 originate_retries
- 403 originate_retry_sleep_ms
- 404 origination_timeout
- 405 IMPORTANT NOTE
- 406 origination_uuid
- 407 originator
- 408 originator_video_codec
- 409 other_loopback_leg_uuid
- 410 pa_hold_file
- 411 pa_ring_file
- 412 playback_terminators
- 413 pound_replace
- 414 presence_id
- 415 profile_start_epoch
- 416 profile_start_stamp
- 417 profile_start_uepoch
- 418 progress_epoch
- 419 progress_media_epoch
- 420 progress_media_stamp
- 421 progress_media_uepoch
- 422 progress_mediamsec
- 423 progress_mediasec
- 424 progress_mediausec
- 425 progress_stamp
- 426 progress_timeout
- 427 progress_uepoch
- 428 progressmsec
- 429 progresssec
- 430 progressusec
- 431 proxy_media
- 432 rdnis
- 433 read_rate
- 434 read_result
- 435 read_terminator_used
- 436 recovery_profile_name
- 437 remote_media_ip
- 438 remote_media_port
- 439 remote_video_ip
- 440 remote_video_port
- 441 ringback
- 442 rss_alt_config
- 443 rtp_adv_audio_ip
- 444 rtp_autoflush
- 445 rtp_hold_timeout_sec
- 446 rtp_manual_rtp_bugs
- 447 rtp_rewrite_timestamps
- 448 rtp_stun_ping
- 449 rtp_timeout_sec
- 450 rtp_timer_name
- 451 signal_bridge_to
- 452 sip
- 453 sip_auth_method
- 454 sip_auth_realm
- 455 sip_auto_answer
- 456 sip_call_id
- 457 sip_codec_negotiation
- 458 sip_contact_host
- 459 sip_contact_port
- 460 sip_contact_user
- 461 sip_copy_custom_headers
- 462 sip_destination_url
- 463 sip_enable_soa
- 464 sip_exclude_contact
- 465 sip_force_video_fmtp
- 466 sip_from_comment
- 467 sip_from_host
- 468 sip_from_port
- 469 sip_from_uri
- 470 sip_from_user
- 471 sip_from_user_stripped
- 472 sip_gateway
- 473 sip_gateway_name
- 474 sip_h_Referred-By
- 475 sip_header_name
- 476 sip_history_info
- 477 sip_ignore_remote_cause
- 478 sip_invite_call_id
- 479 sip_local_url
- 480 sip_looped_call
- 481 sip_nat_detected
- 482 sip_outgoing_call_id
- 483 sip_p_rtp_stat
- 484 sip_profile
- 485 sip_profile_name
- 486 sip_received_ip
- 487 sip_received_port
- 488 sip_refer_reply
- 489 sip_referred_by_cid
- 490 sip_referred_by_user_stripped
- 491 sip_reply_host
- 492 sip_request_host
- 493 sip_request_port
- 494 sip_require_timer
- 495 sip_route_uri
- 496 sip_rtp_rxstat
- 497 sip_rtp_txstat
- 498 sip_sticky_contact
- 499 sip_subject
- 500 sip_term_cause
- 501 sip_term_status
- 502 sip_to_comment
- 503 sip_to_host
- 504 sip_to_port
- 505 sip_to_uri
- 506 sip_to_user
- 507 sip_transport
- 508 sip_use_gateway
- 509 sip_user_agent
- 510 sip_via_host
- 511 sip_via_port
- 512 sip_via_protocol
- 513 sip_via_rport
- 514 sip_video_fmtp
- 515 sip_video_pt
- 516 socket_host
- 517 socket_path
- 518 SOFIA_CRYPTO_MANDATORY_VARIABLE
- 519 SOFIA_HAS_CRYPTO_VARIABLE
- 520 sofia_profile_domain_name
- 521 sofia_profile_name
- 522 sofia_record_file
- 523 SOFIA_REFER_TO_VARIABLE
- 524 SOFIA_REPLACES_HEADER
- 525 SOFIA_SECURE_MEDIA_CONFIRMED_VARIABLE
- 526 SOFIA_SECURE_MEDIA_VARIABLE
- 527 SOFIA_SESSION_TIMEOUT
- 528 sound_prefix
- 529 star_replace
- 530 start_epoch
- 531 start_stamp
- 532 start_uepoch
- 533 stream_prebuffer
- 534 supress_cng
- 535 SWITCH_PLAYBACK_TERMINATOR_USED
- 536 SWITCH_UUID_BRIDGE
- 537 sip_auto_simplify
- 538 tone_detect_expires
- 539 tone_detect_sleep
- 540 transfer_fallback_extension
- 541 transfer_history
- 542 transfer_ringback
- 543 transfer_source
- 544 tts_engine
- 545 tts_voice
- 546 uduration
- 547 UNIQUEID
- 548 user_context
- 549 user_name
- 550 verbose_presence
- 551 video_possible
- 552 video_read_codec
- 553 video_read_rate
- 554 video_write_codec
- 555 video_write_rate
- 556 vm_cc
- 557 vm_message_ext
- 558 vmd_detect
- 559 vname
- 560 voicemail_account
- 561 voicemail_alternate_greet_id
- 562 voicemail_authorized
- 563 voicemail_caller_id_name
- 564 voicemail_caller_id_number
- 565 voicemail_current_folder
- 566 voicemail_domain
- 567 voicemail_domain_name
- 568 voicemail_email
- 569 voicemail_file_path
- 570 voicemail_greeting_number
- 571 voicemail_greeting_path
- 572 voicemail_id
- 573 voicemail_message_len
- 574 voicemail_priority
- 575 voicemail_profile_name
- 576 voicemail_read_flags
- 577 voicemail_time
- 578 voicemail_total_new_messages
- 579 voicemail_total_saved_messages
- 580 voicemail_urgent_new_messages
- 581 voicemail_urgent_saved_messages
- 582 waitmsec
- 583 waitsec
- 584 waitusec
- 585 write_rate
- 586 xfer_uuids
- 587 xml_cdr_base
Variable References
You will see references to variables in the dialplan of the form ${variable} as well as $${variable}.
$${variable} is evaluated once and becomes a static reference to the variable and is therefore suitable for variables that do not change, such as the domain of the FreeSWITCH server. That is why you see $${domain} referenced so frequently in the Vanilla dialplan examples and as pre-processor variables which are evaluated once at startup.
${variable} is evaluated 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.
absolute_codec_string
Sets the absolute codec string to use (nothing will be appended).
Usage:
<action application="set" data="absolute_codec_string=PCMU,GSM"/>
<action application="bridge" data="sofia/gateway/myprovider/5551231234"/>
Per-leg Dialstring:
<action application="bridge" data=[leg_timeout=60,origination_caller_id_number=918039251,
absolute_codec_string=^^:PCMA:PCMU]sofia/gateway/gateway1/${destination_number}|[leg_timeout=60,
absolute_codec_string=PCMA]sofia/gateway/gateway2/${destination_number}"/>
Global Dialstring:
<action application="bridge" data="{absolute_codec_string=PCMA}sofia/gateway/gateway2/${destination_number}"/>
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_opal | mod_opal.cpp | 10567 |
mod_sofia | sofia_glue.c | 12700 |
core | switch_ivr_originate.c | 12817 |
accountcode
Account code is mostly an arbitrary value that you can assign on a per leg basis. An important feature of accountcode is that if its value matches one of the CDR CSV templates defined in cdr_csv.conf.xml then that CDR template will be used when generating a CSV CDR.
Usage:
<action application="set" data="accountcode=custom"/>
acl_token
Description needed! Please contribute one.
Usage:
Example needed! Please contribute one.
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_sofia | sofia.c | 9056 |
## alert_info
Add an Alert-Info Header like Snom and other Phones need for Ring or Ringtone informations.
Usage:
Load an external Wav Rington on all Calls.
<action application="export" data="alert_info=http://192.168.181.5/sounds/ctu.wav"/>
Use the External Ringer (Change the Ringtone for the alert-external, alert-group and alert-internal in Snom under Setup > Preferences > Alert-Info Ringer)
<action application="export" data="alert_info=http://www.notused.com;info=alert-external;x-line-id=0"/>
(Set under See also: http://wiki.snom.com/Web%5FInterface/V8/Preferences/Alert-Info%5FRinger
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_sofia | sofia.c | 6350 |
mod_sofia | sofia_glue.c | 4819 |
answer_epoch
Description needed! Please contribute one.
Usage:
Example needed! Please contribute one.
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
core | switch_channel.c | 6917 |
answer_stamp
Description needed! Please contribute one.
Usage:
Example needed! Please contribute one.
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
core | switch_channel.c | 7750 |
answer_uepoch
Description needed! Please contribute one.
Usage:
Example needed! Please contribute one.
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
core | switch_channel.c | 8686 |
answermsec
Description needed! Please contribute one.
Usage:
Example needed! Please contribute one.
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
core | switch_channel.c | 3546 |
answersec
Description needed! Please contribute one.
Usage:
Example needed! Please contribute one.
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
core | switch_channel.c | 3546 |
answerusec
Description needed! Please contribute one.
Usage:
Example needed! Please contribute one.
See also:
Module Name | Source File | Last Revised |
---|---|---|
core | switch_channel.c | 3546 |
api_after_bridge
Execute an API command after bridge.
Usage:
Paging to PA System via Portaudio (w/ chime before and after announcement)
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
core | switch_ivr_bridge.c | 7885 |
api_hangup_hook
Execute an API command on hangup.
Usage:
<action application="set" data="api_hangup_hook=jsrun cleanup.js ${uuid}"/>
See also:
- session_in_hangup_hook
- api_reporting_hook - like api_hangup_hook but after reporting state (both honor session_in_hangup_hook)
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
core | switch_core_state_machine.c | 7885 |
api_on_answer
Execute an api (not an application) when the called party answers. To set an application, use execute_on_answer.
Usage:
<action application="export" data="nolocal:api_on_answer=uuid_broadcast ${uuid} beep.wav both"/>
Or,
<action application="bridge" data="{api_on_answer='uuid_broadcast ${uuid} beep.wav both'}sofia/gateway/provider/5551231234"/>
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
core | switch_channel.c | 15207 |
api_on_media
Execute a FreeSWITCH API when the far end sends media, i.e. ringing or 183/SDP.
The command is executed only on channels that are not already answered. Just use export or export with nolocal: prefix to make sure it is executed when b-leg answers.
In the second usage example below, we have originated an outbound call to a local extension, where we will wait 30 seconds without ignoring media. In this case we use 'set' and not 'export'.
Usage:
<action application="export" data="nolocal:execute_on_media=lua incrInUse.lua ${uuid}"/>
Implemented By:
Module Name | Source File | Last Updated |
---|---|---|
Core | switch_channel.c | 850f2e3 |
api_on_startup
Execute an api (not an application) when the switch starts up.
Usage:
<X-PRE-PROCESS cmd="set" data="api_on_startup=sofia recover"/>
Implemented By:
Module Name | Source File | Last Revised | date |
---|---|---|---|
core | switch_channel.c | e164b76caf0a47b6b33891eea80b1928067c9dd1 | 2011-12-15 |
asr_intercept_dtmf
Description needed! Please contribute one.
Usage:
Example needed! Please contribute one.
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
core | switch_ivr_async.c | 8912 |
auto_answer_destination
Description needed! Please contribute one.
Usage:
Example needed! Please contribute one.
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_sofia | sofia.c | 10525 |
auto_hunt
Setting auto_hunt to "true" will alter the normal sequential processing of dialplan extensions. auto_hunt will cause the dialplan to 'jump' to a specific extension name, not processing any other extension. The destination_number and extension name must be the same in order for this to work. The condition must still match, but the extension name is the operative element.
In the example below, there is no way to reach extension 333 without auto_hunt.
Usage: In vars.xml:
<X-PRE-PROCESS cmd="set" data="auto_hunt=true"/>
Example:
<extension name="do_xfer">
<condition field="destination_number" expression="^.*$">
<action application="set" data="auto_hunt=true"/>
<action application="transfer" data="333"/>
</condition>
</extension>
<extension name="333">
<condition field="destination_number" expression="^333$">
<action application="info"/>
</condition>
</extension>
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_dialplan_xml | mod_dialplan_xml.c | 12144 |
billmsec
Description needed! Please contribute one.
Usage:
Example needed! Please contribute one.
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
core | switch_channel.c | 6917 |
billsec
Description needed! Please contribute one.
Usage:
Example needed! Please contribute one.
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
core | switch_channel.c | 6917 |
billusec
Description needed! Please contribute one.
Usage:bypass_media
Example needed! Please contribute one.
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
core | switch_channel.c | 6917 |
bind_meta_key
Selects the meta key to use with the mod_dptools: bind_meta_app dialplan application.
Usage:
<action application="set" data="bind_meta_key=#"/>
Implemented By:
bridge_answer_timeout
Timeout in seconds how long to tolerate a bridge that is in early media without being answered (can be set on either leg). Useful when you want to pass early media from b-leg to a-leg but also use ${call_timeout}. This will consider the bridge failed if a 200 OK is not received before the bridge_answer_timeout.
Usage:
<action application="set" data="bridge_answer_timeout=20"/>
tags bridge answer timeout ringer ring timer voicemail
bridge_channel
Description needed! Please contribute one.
Usage:
Example needed! Please contribute one.
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
core | switch_ivr_bridge.c | 6537 |
bridge_early_media
By default this is false. Set to true, this makes the bridge use the live audio from the b-leg as ringback to the a-leg. Setting bridge_early_media=true means the early media will be buffered.
Consider setting this to true if you are using a loopback channel to execute a bridge to an endpoint which sends back early media and the received early media's audio is degraded. The buffering resulting from setting bridge_early_media=true brings with it a higher resource cost (than bridge_early_media=false), but may improve the sound quality of the early media.
Usage: Set bridge_early_media before the bridge, or in the dial string for the bridge.
bridge_epoch
Description needed! Please contribute one.
Usage:
Example needed! Please contribute one.
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
core | switch_channel.c | 3492 |
bridge_filter_dtmf
Setting this variable to true will prevent DTMF digits received on this channel when bridged from being sent to the other channel.
Usage:
You can set this variable in the dialplan before answering or inline as part of a dialstring.
Example dialplan usage:
<action application="set" data="bridge_filter_dtmf=true" />
Example dialstring usage:
{bridge_filter_dtmf=true}sofia/default/blah@baz.com
bridge_generate_comfort_noise
Generates comfort noise when bridged session is receiving discontinuous audio frames (silence suppression).
When greater than 0, this variable is set to the divisor of the silence generating function. 400 or 1400 are common values set, but you may experiment with other choices to pick one that sounds best.
When true, FreeSWITCH will pick a default comfort noise value.
When -1, FreeSWITCH will transmit silence without comfort noise. (As of 2012-10-25)
Usage:
Leg A wants a continuous stream of audio, but leg B is using silence suppression:
<action application="set" data="bridge_generate_comfort_noise=true"/>
<action application="bridge" data="sofia/user/1000"/>
Leg A uses silence suppression, but leg B wants a continuous stream of audio:
<action application="bridge" data="{bridge_generate_comfort_noise=true}sofia/user/1000"/>
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
core | switch_ivr_bridge.c | 9543 |
bridge_hangup_cause
This is set to the hangup cause of the last bridged B leg of the call. If you have continue_on_fail=true and hangup_after_bridge=false you can do checks on this to see what "really" happened to the call. You can for instance do execute_extension after bridge, do a condition check on ${bridge_hangup_cause} to see if it contains MEDIA_TIMEOUT and then trigger a redial of the call or transfer to a cell phone. For a list of hangup causes, see Hangup Causes.
Usage:
<action application="log" data="1 B-leg hangup cause: ${bridge_hangup_cause}"/>
bridge_pre_execute_aleg_app
Command or api to be executed on the A leg before bridging the two channels.
Note: this is executed AFTER the call is setup but BEFORE the media (audio) is bridged.
Usage:
Example needed! Please contribute one
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
core | switch_ivr_bridge.c | 8108 |
bridge_pre_execute_aleg_data
Arguments to be used with bridge_pre_execute_aleg_app.
Usage:
Example needed! Please contribute one.
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
core | switch_ivr_bridge.c | 8108 |
bridge_pre_execute_bleg_app
Command or api to be executed on the B leg before bridging the two channels. Useful when originating a call from the event socket, CLI or XML-RPC.
It could for instance be used to do a HTTP GET with a script or mod_http to the IP address of a Snom phone to increase the ringer volume if you need to do a wakeup call.
Can also be used to bind a dtmf to an app on the b leg of a call so that it can survive a transfer.
Note: this is executed AFTER the call is setup but BEFORE the media (audio) is bridged.
Usage:
<action application="set" data="bridge_pre_execute_bleg_app=bind_meta_app"/>
<action application="set" data="bridge_pre_execute_bleg_data=1 a s att_xfer::sofia/profile/destination"/>
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
core | switch_ivr_bridge.c | 8108 |
bridge_pre_execute_bleg_data
Arguments to be used with bridge_pre_execute_bleg_app
Usage:
<action application="set" data="bridge_pre_execute_bleg_app=bind_meta_app"/> <action application="set" data="bridge_pre_execute_bleg_data=1 a s att_xfer::sofia/profile/destination"/>
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
core | switch_ivr_bridge.c | 8108 |
bridge_stamp
Description needed! Please contribute one.
Usage:
Example needed! Please contribute one.
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
core | switch_channel.c | 3443 |
bridge_terminate_key
Allows you to bind a key and the bridge will terminate if the dtmf matches
Usage: you can set bridge_terminate_key on either or both legs which will end the bridge, if it hangs up or not is decided by hangup_after_bridge=false or what is next in your dp
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_dptools | mod_dptools.c | 12479 |
bridge_to
Description needed! Please contribute one.
Usage:
Example needed! Please contribute one.
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_commands | mod_commands.c | 7421 |
core | switch_ivr.c | 5738 |
core | switch_ivr_async.c | 4798 |
core | switch_ivr_bridge.c | 12671 |
bridge_uepoch
Description needed! Please contribute one.
Usage:
Example needed! Please contribute one.
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
core | switch_channel.c | 3494 |
bridge_uuid
Description needed! Please contribute one.
Usage:
Example needed! Please contribute one.
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
core | switch_ivr_bridge.c | 9603 |
bypass_media
When set, all the media address of the far end of the originating leg will be passed to the far end of the new call leg and vice versa so the signaling goes through FreeSWITCH but the media is point-to-point.
Usage:
<action application="set" data="bypass_media=true"/>
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_dptools | mod_dptools.c | 12479 |
mod_esf | mod_esf.c | 5115 |
bypass_media_after_bridge
This is useful for bypassing media after bridging has happened.
Usage:
<action application="set" data="bypass_media_after_bridge=true"/>
It could be useful for an already established channel and doing a later bypass.
uuid_setvar <uuid> bypass_media_after_bridge true
uuid_broadcast <uuid> bridge::<endpoint>
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_dptools | mod_dptools.c | 12479 |
mod_sofia | sofia.c | 9591 |
core | switch_ivr_bridge.c | 9591 |
cache_speech_handles
Description needed! Please contribute one.
Usage:
Example needed! Please contribute one.
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
core | switch_ivr_play_say.c | 5876 |
call_clientcode
Description needed! Please contribute one.
Usage:
Example needed! Please contribute one.
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_sofia | sofia.c | 6987 |
call_timeout
Controls how long (in seconds) to ring the B leg of a call when using the bridge application. The timeout is set on the A leg, and applies to any bridges that happen in the channel.
If you need to set a timeout on a call that has no A leg, use originate_timeout
If you need to set a timeout with enterprise bridging/originate, use originate_timeout
If you need to set the timeout on a per leg basis (i.e., a different timeout for each destination), use the leg_timeout variable.
Default Value: 60
Usage:
<action application="set" data="call_timeout=20"/>
Notes:
If a call timeout is to be specified against a group_call() list, use the following format:
<action application="bridge" data="{originate_timeout=24}${group_call(sales@$${domain})}"/>
Beware that if you are not using {ignore_early_media=true} call_timeout is no longer applicable as soon as early media signal is received.
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_dialplan_asterisk | mod_dialplan_asterisk.c | 6208 |
mod_dptools | mod_dptools.c | 10917 |
core | switch_ivr_originate.c | 9120 |
core | switch_swig.c | 4795 |
caller_id
Description needed! Please contribute one.
Usage:
Example needed! Please contribute one.
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
core | switch_channel.c | 6917 |
caller_id_name
The caller id name set by the inbound call, not a real variable. Practically it is read only.
caller_id_number
The caller id phone number set by the inbound call, not a real variable. Practically it is read only. From sofia.c, the values used (in precedence) are the user parts from: P-Preferred-Identity, P-Asserted-Identity, Remote-Party-ID, and the From header.
campon
Controls whether camping is enabled or not.
Default: false
Usage:
<action application="set" data="campon=true"/>
<action application="bridge" data="sofia/gateway/myprovider/5551231234"/>
## campon_announce_sound
File to play back after the first bridge fails (rg to announce what key to press to skip to fallback extension)
Default: none
Usage:
<action application="set" data="campon=true"/>
<action application="set" data="campon_stop_key=1"/>
<action application="set" data="campon_announce_sound=press_one_to_stop.wav"/>
<action application="bridge" data="sofia/gateway/myprovider/5551231234"/>
campon_fallback_context
Controls camping during bridge app (testing needed)
Usage:
<action application="set" data="campon"/>
<action application="bridge" data="sofia/gateway/myprovider/5551231234"/>
campon_fallback_dialplan
Controls camping during bridge app (testing needed)
Usage:
<action application="set" data="campon"/>
<action application="bridge" data="sofia/gateway/myprovider/5551231234"/>
campon_hold_music
If you don't set the hold_music variable, this variable controls hold music while camping.
Usage:
<action application="set" data="campon=true"/>
<action application="set" data="campon_hold_music=/data/campmusic/RelaxingCampSounds.wav"/>
<action application="bridge" data="sofia/gateway/myprovider/5551231234"/>
campon_fallback_exten
Controls camping during bridge app (testing needed)
Usage:
<action application="set" data="campon"/>
<action application="bridge" data="sofia/gateway/myprovider/5551231234"/>
campon_retries
Controls how many times the bridge will be retried while camping.
Default: 100
Usage:
<action application="set" data="campon=true"/>
<action application="set" data="campon_retries=13"/>
<action application="bridge" data="sofia/gateway/myprovider/5551231234"/>
## campon_sleep
Controls how long to wait before starting a retry.
Default: 10
Usage:
<action application="set" data="campon=true"/>
<action application="set" data="campon_sleep=30"/>
<action application="bridge" data="sofia/gateway/myprovider/5551231234"/>
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_dptools | mod_dptools.c | 15201 |
campon_stop_key
DTMF digit that breaks the campon loop and skips directly to fallback extension
Default: none
Usage:
<action application="set" data="campon=true"/>
<action application="set" data="campon_stop_key=1"/>
<action application="set" data="campon_announce_sound=press_one_to_stop.wav"/>
<action application="set" data="campon_fallback_exten=1000"/>
<action application="bridge" data="sofia/gateway/myprovider/5551231234"/>
campon_timeout
This variable controls how long to attempt each bridge before timing out. It works exactly like call_timeout but only applies to camping.
Default: 10
Usage:
<action application="set" data="campon=true"/>
<action application="set" data="campon_timeout=20"/>
<action application="bridge" data="sofia/gateway/myprovider/5551231234"/>
## cdr_csv_base
Description needed! Please contribute one.
Usage:
Example needed! Please contribute one.
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_cdr_csv | mod_cdr_csv.c | 6542 |
CHANNEL
Description needed! Please contribute one.
Usage:
Example needed! Please contribute one.
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_dialplan_asterisk | mod_dialplan_asterisk.c | 6205 |
channel_name
Description needed! Please contribute one.
Usage:
Example needed! Please contribute one.
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
core | switch_channel.c | 6542 |
codec_string
Sets the base codec string to use.
Usage:
<action application="set" data="codec_string=PCMU,GSM"/>
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_opal | mod_opal.cpp | 10567 |
mod_sofia | sofia_glue.c | 5114 |
conference_auto_outcall_announce
File name of audio message to play to conference member joining conference via the conference_set_auto_outcall application. Because the conference would be originating an outbound call to a member this typically would be a greeting with an explanation that the recipient will be joining a conference call.
Usage:
<action application="set" data="conference_auto_outcall_announce=sounds/soundfile.wav"/>
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_conference | mod_conference.c | 11104 |
conference_auto_outcall_caller_id_name
Caller ID name to use when dialing endpoints to join the conference via the conference_set_auto_outcall application.
Usage:
<action application="set" data="conference_auto_outcall_caller_id_name=$${effective_caller_id_name}"/>
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_conference | mod_conference.c | 11104 |
conference_auto_outcall_caller_id_number
Caller ID number to use when dialing endpoints to join the conference via the conference_set_auto_outcall application.
Usage:
<action application="set" data="conference_auto_outcall_caller_id_number=${effective_caller_id_number}"/>
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_conference | mod_conference.c | 11104 |
conference_auto_outcall_flags
Conference flags to set for members joining conference via the conference_set_auto_outcall application
Usage:
<action application="set" data="conference_auto_outcall_flags=mute"/>
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_conference | mod_conference.c | 11104 |
conference_auto_outcall_maxwait
Maximum time in seconds that the channel that initiated the conference_set_auto_outcall will wait for members to join the conference.
Usage:
<action application="set" data="conference_auto_outcall_maxwait=10"/>
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_conference | mod_conference.c |
conference_auto_outcall_prefix
The value of conference_auto_outcall_prefix is prepended to each of conference_set_auto_outcall values, of which there can be more than one.
Usage:
<extension name="mad_boss_intercom">
<condition field="destination_number" expression="^0911$">
<action application="set" data="conference_auto_outcall_caller_id_name=Mad Boss1"/>
<action application="set" data="conference_auto_outcall_caller_id_number=0911"/>
<action application="set" data="conference_auto_outcall_timeout=60"/>
<action application="set" data="conference_auto_outcall_flags=mute"/>
<action application="set" data="conference_auto_outcall_prefix={sip_auto_answer=true,execute_on_answer='bind_meta_app 2 a s1 transfer::intercept:${uuid} inline'}"/>
<action application="set" data="sip_exclude_contact=${network_addr}"/>
<action application="conference_set_auto_outcall" data="${group_call(sales)}"/>
<action application="conference" data="madboss_intercom1@default+flags{endconf|deaf}"/>
</condition>
</extension>
See also:
See Conferencing and Intercom for an example of using all this to page via multiple extensions.
See conference for details on initiating conferences.
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_conference | mod_conference.c | 11104 |
conference_auto_outcall_profile
Conference profile to use for members joining the conference via the conference_set_auto_outcall application.
Usage:
<action application="set" data="conference_auto_outcall_profile=default"/>
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_conference | mod_conference.c |
conference_auto_outcall_timeout
Originate timeout to use when joining a member to a conference via conference_set_auto_outcall.
Usage:
<action application="set" data="conference_auto_outcall_timeout=60"/>
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_conference | mod_conference.c | 11104 |
conference_controls
Set this variable to specify which conference control set to use when transferring a caller into a conference. This allows you, for example, to have a control set for the conference moderator and another control set for regular conference members. The control set for the moderator could include the ability to mute or kick people, for example.
NOTE: You must create the desired conference control set. Also, if this is not set then the default conference control set is used for the conference member.
Usage:
<action application="set" data="conference_controls=moderator"/>
Implemented By:
Module Name | Source File | Last Updated |
---|---|---|
mod_conference | mod_conference.c | ac19f73c |
conference_enforce_security
Allows the conference security to be overridden. This applies in two different scenarios, one for inbound and one for outbound. By default, conference security is always applied to inbound calls and is always skipped for outbound calls. This channel variable allows the behavior to be modified.
Usage:
Inbound
<action application="set" data="conference_enforce_security=false"/>
<action application="conference" data="3000"/>
Outbound
originate {conference_enforce_security=true}sofia/internal/1001 &conference(3000)
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_conference | mod_conference.c | 12863 |
conference_enter_sound
When set, this channel variable will override the enter-sound param on conference profile for any conferences into which the call leg is transferred.
Usage:
<action application="set" data="conference_enter_sound=silence_stream://10"/>
Implemented By:
Module Name | Source File | Last Updated |
---|---|---|
Mod_conference | mod_conference.c |
conference_exit_sound
Usage:
<action application="set" data="conference_exit_sound=silence_stream://10"/>
Implemented By:
Module Name | Source File | Last Updated |
---|---|---|
Mod_conference | mod_conference.c |
conference_last_matching_digits
Contains the last matching digits that the user on this channel sent into the conference.
Usage:
<action application="log" data="INFO Last digits sent by this user: ${conference_last_matching_digits}"/>
See also:
Implemented By:
Module Name | Source File | Last Updated |
---|---|---|
mod_conference | mod_conference.c | f6bcf830 |
conference_member_id
Contains the conference_member_id value for any conference to which the channel may be connected.
Implemented By:
mod_conference.c
conference_moderator
Is true if the channel is connected to a conference as a moderator.
Implemented By:
mod_conference.c
conference_name
The name of the last conference joined by this channel.
Usage:
<action application="log" data="INFO Last conference joined by this user: ${conference_name}"/>
Implemented By:
Module Name | Source File | Last Updated |
---|---|---|
mod_conference | mod_conference.c |
conference_recording
Contains the file name of the conference recording for the conference to which the channel is connected.
Implemented By:
mod_conference.c
conference_uuid
Every instance of a conference has its own UUID. This channel variable stores the conference UUID for the most recent conference in which the channel was a member. It is set as soon as the channel enters the conference, and will show up in XML CDRs and uuid_dump calls, as well as any events that show channel variables.
Implemented By:
Module Name | Source File | Last Updated |
---|---|---|
mod_conference | mod_conference.c | 2011-03-18 |
continue_on_fail
Controls what happens when the called party can not be reached (busy/offline). If "true" the dialplan continues to be processed. If "false" the dialplan will stop processing. Can contain the return messages that will continue on fail also.
Usage:
<action application="set" data="continue_on_fail=true"/>
or,
<action application="set" data="continue_on_fail=NORMAL_TEMPORARY_FAILURE,USER_BUSY,NO_ANSWER,NO_ROUTE_DESTINATION"/>
or Q.850 cause codes,
<action application="set" data="continue_on_fail=3,17,18,27"/>
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_dptools | mod_dptools.c | 5875 |
mod_lcr | mod_lcr.c | 10510 |
copy_xml_cdr
Copy's the other leg's XML CDR into this leg's CDR. For example, the A leg's CDR will contain a variable named b_leg_cdr whose contents are the URL-encoded XML CDR data from the B leg. This variable must be set on the B leg, so use {copy_xml_cdr=true} in the dialstring or use exportinstead of set.
Usage:
<action application="bridge" data="{copy_xml_cdr=true} user/${dialed_extension}@${domain_name}"/>
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
core | switch_ivr_bridge.c | 8955 |
created_time
Contains the start time (in microseconds) of when the call was created.
Usage:
In the event that a call is transferred, this will contain the time of when the entire call was created, not that specific transfer. If you need the transfer created time, see Variable_profile_created_time and Variable_progress_time
current_application
Description needed! Please contribute one.
Usage:
Example needed! Please contribute one.
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_fifo | mod_fifo.c | 10277 |
mod_sofia | sofia.c | 11562 |
core | switch_core_session.c | 10653 |
current_application_data
Description needed! Please contribute one.
Usage:
Example needed! Please contribute one.
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_fifo | mod_fifo.c | 10277 |
mod_sofia | sofia.c | 11562 |
current_application_response
Description needed! Please contribute one.
Usage:
Example needed! Please contribute one.
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_dptools | mod_dptools.c | 10653 |
core | switch_core_session.c | 10653 |
default_language
Controls the default language the Say Phrase engine will use when no language is explicitly specified in the API call. This permits you to easily support multiple languages by only changing a single variable at call time.
Usage:
<action application="set" data="default_language=fr"/>
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
core | switch_ivr_play_say.c | 4796 |
destination_number
Description needed! Please contribute one.
Usage:
Example needed! Please contribute one.
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_sofia | sofia.c | 6054 |
core | switch_ivr_bridge.c | 7083 |
detect_speech_result
The result of play_and_detect_speech.
Usage:
This value is read-only.
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_dptools | mod_dptools.c |
dialed_domain
Description needed! Please contribute one.
Usage:
Example needed! Please contribute one.
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_commands | mod_commands.c | 10917 |
mod_dptools | mod_dptools.c | 7225 |
dialed_group
Description needed! Please contribute one.
Usage:
Example needed! Please contribute one.
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_commands | mod_commands.c | 10917 |
dialed_user
Description needed! Please contribute one.
Usage:
Example needed! Please contribute one.
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_commands | mod_commands.c | 10917 |
mod_dptools | mod_dptools.c | 7225 |
digits_dialed
Description needed! Please contribute one.
Usage:
Example needed! Please contribute one.
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
core | switch_channel.c | 12244 |
direction
It is used to determine if the particular leg of the call is inbound or outbound.
Usage:
<condition field="${direction}" expression="^inbound$">
See also:
disable_app_log
Description needed! Please contribute one.
Usage:
Example needed! Please contribute one.
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
core | switch_core_session.c | 8517 |
rtp_disable_hold
When set to true the user may not put the call on hold.
Usage:
<action application="set" data="rtp_disable_hold=true"/>
Implemented By:
mod_sofia
rtp_pass_codecs_on_stream_change
Pass codecs thru from a to b on stream change, example adding or removing video from a bridged call.
Usage
<action application="set" data="rtp_pass_codecs_on_stream_change=true"/>
disable_q850_reason
When set to true, this disables sending of the Reason header, which includes the Q.850 reason code, in responses and BYEs. For a list of hangup causes and their Q.850 codes, see Hangup Causes. This is available as of revision 15850 committed 12/8/2009.
Usage:
<action application="set" data="disable_q850_reason=true"/>
disable_radius_start
Description needed! Please contribute one.
Usage:
Example needed! Please contribute one.
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_radius_cdr | mod_radius_cdr.c | 10793 |
disable_radius_stop
Description needed! Please contribute one.
Usage:
Example needed! Please contribute one.
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_radius_cdr | mod_radius_cdr.c | 10793 |
disable_rtp_auto_adjust
Disable rtp auto adjust if it not behaves as you expected.
It stops the switch from rewriting the RTP destination based on the source
When RTP Auto-Adjust is ON FreeSWITCH will change the destination RTP address (port?) to match the source of the incoming packets, this doesn't work if the other end is really wanting to send and receive on a different addr (port?).
Usage:
<action application="set" data="disable_rtp_auto_adjust=true"/>
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_dingaling | mod_dingaling.c | 12641 |
mod_sofia | sofia_glue.c | 8908 |
DISPLACE_HANGUP_ON_ERROR
When set to true this channel variable will cause the call to hangup if there is an error when trying to uuid_displace the call. The default is "false". The default behavior before 2013/03/21 was to hangup.
Usage:
<action application="set" data="DISPLACE_HANGUP_ON_ERROR=true"/>
See also:
Implemented By:
Module Name | Source File | Last Updated |
---|---|---|
Core | switch_ivr_async.c |
dl_cid_msg
Description needed! Please contribute one.
Usage:
Example needed! Please contribute one.
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_dingaling | mod_dingaling.c | 4270 |
dl_from_host
Description needed! Please contribute one.
Usage:
Example needed! Please contribute one.
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_dingaling | mod_dingaling.c | 4943 |
dl_from_user
Description needed! Please contribute one.
Usage:
Example needed! Please contribute one.
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_dingaling | mod_dingaling.c | 4943 |
dl_host
Description needed! Please contribute one.
Usage:
Example needed! Please contribute one.
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_dingaling | mod_dingaling.c | 4943 |
dl_user
Description needed! Please contribute one.
Usage:
Example needed! Please contribute one.
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_dingaling | mod_dingaling.c | 4943 |
domain_name
Description needed! Please contribute one.
Usage:
Example needed! Please contribute one.
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
core | switch_ivr.c | 9056 |
drop_dtmf
Set this on an inbound channel before answer or on an outbound channel before the bridge/originate in order to prevent DTMF events from being sent to the channel.
Only tested with RFC2833, may also work for INFO / inband. See Jira issue FS-4769. Commit 60f7849cbe72.
Usage:
<action application="set" data="drop_dtmf=true"/>
<action application="answer"/>
or,
<action application="export" data="nolocal:drop_dtmf=true"/>
<action application="bridge" data="sofia/internal/100@1.2.3.4"/>
or,
<action application="bridge" data="{drop_dtmf=true}sofia/internal/100@1.2.3.4"/>
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_sofia | sofia_glue.c |
dtmf_type
For inband DTMF, Misc. Dialplan Tools start_dtmf must be used in the dialplan.
Usage:
<action application="set" data="dtmf_type=info"/>
or,
<action application="set" data="dtmf_type=rfc2833"/>
or,
<action application="set" data="dtmf_type=none"/>
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_sofia | sofia_glue.c | 6952 |
dtmf_verbose
Enables verbose logging of Spandsp DTMF detector. Default is false. Set this variable prior to executing spandsp_start_dtmf.
Usage:
See also:
Implemented By:
Module Name | Source File | Last Updated |
---|---|---|
mod_spandsp | mod_spandsp_dsp.c |
duration
Description needed! Please contribute one.
Usage:
Example needed! Please contribute one.
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
core | switch_channel.c | 6917 |
easy_acctcode
Description needed! Please contribute one.
Usage:
Example needed! Please contribute one.
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_easyroute | mod_easyroute.c | 10929 |
easy_destnum
Description needed! Please contribute one.
Usage:
Example needed! Please contribute one.
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_easyroute | mod_easyroute.c | 10929 |
easy_dialstring
Description needed! Please contribute one.
Usage:
Example needed! Please contribute one.
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_easyroute | mod_easyroute.c | 10929 |
easy_group
Description needed! Please contribute one.
Usage:
Example needed! Please contribute one.
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_easyroute | mod_easyroute.c | 10929 |
easy_limit
Description needed! Please contribute one.
Usage:
Example needed! Please contribute one.
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_easyroute | mod_easyroute.c | 10929 |
eavesdrop_annnounce_macro
Description needed! Please contribute one.
Usage:
Example needed! Please contribute one.
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
core | switch_ivr_async.c | 8098 |
eavesdrop_announce_id
Description needed! Please contribute one.
Usage:
Example needed! Please contribute one.
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
core | switch_ivr_async.c | 8098 |
eavesdrop_group
Description needed! Please contribute one.
Usage:
Example needed! Please contribute one.
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
core | switch_ivr_async.c | 8686 |
eavesdrop_indicate_failed
Description needed! Please contribute one.
Usage:
Example needed! Please contribute one.
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_dptools | mod_dptools.c | 8098 |
eavesdrop_indicate_idle
Description needed! Please contribute one.
Usage:
Example needed! Please contribute one.
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_dptools | mod_dptools.c | 8091 |
eavesdrop_indicate_new
Description needed! Please contribute one.
Usage:
Example needed! Please contribute one.
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_dptools | mod_dptools.c | 8091 |
eavesdrop_require_group
Description needed! Please contribute one.
Usage:
Example needed! Please contribute one.
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_dptools | mod_dptools.c | 8098 |
effective_caller_id_name
Sets the effective callerid name. This is automatically exported to the B-leg; however, it is not valid in an origination string. In other words, set this before calling bridge, otherwise use origination_caller_id_name
For Snom 370/820 users:
If you want to display LEG A's name (if available) as soon as LEG B (here the local Snom) rings, you have to set origination_caller_id_name or effective_caller_id_name as described. Otherwise, in LEG B's display, you will see LEG A's number during ringing and switching to LEG A's name after picking up the call by LEG B. To remove it set it to "_undef_".
Usage:
<action application="set" data="effective_caller_id_name=Bob Smith"/>
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_voicemail | mod_voicemail.c | 8854 |
core | switch_core_session.c | 9248 |
effective_caller_id_number
Sets the effective callerid number. This is automatically exported to the B-leg; however, it is not valid in an origination string. In other words, set this before calling bridge, otherwise use origination_caller_id_number
Usage:
<action application="set" data="effective_caller_id_number=9185551212"/>
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_voicemail | mod_voicemail.c | 8854 |
core | switch_core_session.c | 9248 |
enable_file_write_buffering
Enable file buffering when recording a file, defaults to true if not set.
Buffer size defaults to `SWITCH_DEFAULT_FILE_BUFFER_LEN` but can be overridden by putting bytes size instead of true (see below example).
Related discussion; http://lists.freeswitch.org/pipermail/freeswitch-users/2012-April/082835.html
Usage:
<action application="set" data="enable_file_write_buffering=false"/>
<action application="set" data="enable_file_write_buffering=true"/>
<action application="set" data="enable_file_write_buffering=65535"/>
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
core | switch_ivr_play_say.c | 11677 |
enable_heartbeat_events
Description needed! Please contribute one.
Usage:
Example needed! Please contribute one.
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
core | switch_channel.c | 9882 |
end_epoch
Description needed! Please contribute one.
Usage:
Example needed! Please contribute one.
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
core | switch_channel.c | 6917 |
end_stamp
Description needed! Please contribute one.
Usage:
Example needed! Please contribute one.
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
core | switch_channel.c | 6917 |
end_uepoch
Description needed! Please contribute one.
Usage:
Example needed! Please contribute one.
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
core | switch_channel.c | 6917 |
ep_codec_prefer_sdp
ep_codec_prefer_sdp contains the "endpoint" codec string on the A leg. The order of preference is chosen by the A leg
Usage:
Need some example.
Implemented By:
Module Name | Source File | Last Updated |
---|---|---|
Sofia | mod_sofia_glue.c |
ep_codec_string
ep_codec_string contains the "endpoint" codec string on the A leg. This codec list includes only the codecs that both the A leg and FreeSWITCH agree upon. This variable is set only when inbound-late-negotiation is enable on the SIP profile.
NOTE: When there is more than one codec in the list, the order of preference is chosen by FreeSWITCH, not by the A leg
Usage:
<action application="bridge" data="{absolute_codec_string=${ep_codec_string}}sofia/foo/bar@baz"/>
Implemented By:
Module Name | Source File | Last Updated |
---|---|---|
Sofia | mod_sofia.c |
exec_after_bridge_app
Execute an application command after the bridge has been terminated. To be used with exec_after_bridge_arg. By contrast, to execute when the bridge has been established use execute_on_answer
Usage:
<action application="set" data="exec_after_bridge_app=transfer"/>
<action application="set" data="exec_after_bridge_arg=2102"/>
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
core | switch_ivr_bridge.c | 5546 |
exec_after_bridge_arg
Argument passed to exec_after_bridge_app.
Usage:
<action application="set" data="exec_after_bridge_app=transfer"/>
<action application="set" data="exec_after_bridge_arg=2102"/>
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
core | switch_ivr_bridge.c | 5546 |
execute_on_answer
Execute an application (not an api) when the called party answers. To set an api, use api_on_answer. execute_on_answer will also allow for more control when dealing with no answer conditions in cases where you cannot ignore early media.
The command is executed only on channels that are not already answered. Just use export or export with nolocal: prefix to make sure it is executed when b-leg answers.
In the second usage example below, we have originated an outbound call to a local extension, where we will wait 30 seconds while manually ignoring media. In this case we use 'set' and not 'export'.
Usage:
<action application="export" data="nolocal:execute_on_answer=lua incrInUse.lua ${uuid}"/>
or, to wait 30 seconds for an answer while 'manually' ignoring early media
originate {ignore_early_media=true}sofia/gateway/my_gateway/5551212 885551212
<extension name="exe_on_ans">
<condition field="destination_number" expression="^88(\d+)$">
<action application="set" data="execute_on_answer=transfer ANSWEREDCALL XML default"/>
<action application="log" data="INFO Waiting 30 seconds for $1 to answer..."/>
<action application="sleep" data="30000"/>
<action application="log" data="INFO Call to $1 was not answered, taking alternative action..."/>
<action application="transfer" data="UNANSWEREDCALL XML default"/>
</condition>
</extension>
If you need to set multiple execute_on_answers, see: The execute on family
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
core | switch_channel.c | 9636 |
execute_on_fax_detect
This is an example of how to create a channel variable page. This section is the description of the variable. Put the description information here and then the usage example below.
Usage:
<action application="set" data="skeleton=foo"/>
execute_on_fax_failure
This is an example of how to create a channel variable page. This section is the description of the variable. Put the description information here and then the usage example below.
Usage:
<action application="set" data="skeleton=foo"/>
execute_on_fax_result
This is an example of how to create a channel variable page. This section is the description of the variable. Put the description information here and then the usage example below.
Usage:
<action application="set" data="skeleton=foo"/>
execute_on_fax_success
This is an example of how to create a channel variable page. This section is the description of the variable. Put the description information here and then the usage example below.
Usage:
<action application="set" data="skeleton=foo"/>
execute_on_media
Execute an application when the far end sends media, i.e. ringing or 183/SDP.
The command is executed only on channels that are not already answered. Just use export or export with nolocal: prefix to make sure it is executed when b-leg answers.
In the second usage example below, we have originated an outbound call to a local extension, where we will wait 30 seconds without ignoring media. In this case we use 'set' and not 'export'.
Usage:
<action application="export" data="nolocal:execute_on_media=lua incrInUse.lua ${uuid}"/>
execute_on_media_timeout
Execute an application if the far end stops sending media and times out.
Usage:
<action application="export" data="nolocal:execute_on_media_timeout=lua oops_timeout.lua ${uuid}"/>
<action application="set" data="execute_on_media_timeout=transfer HANDLE_MEDIA_TIMEOUT XML default"/>
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
Sofia | mod_sofia.c | 7eafa85d |
execute_on_originate
Executes code on successful origination. Use the '<app> <arg>' format to execute in the origination thread or use '<app>::<arg>' to execute asynchronously.
Successful origination means the remote server responds, NOT when the call is actually answered.
Usage:
originate {ignore_early_media=true,execute_on_originate='cng_plc'}sofia/gateway/foo/123456789 9664
originate {ignore_early_media=true,execute_on_originate='my_app::my_arg'}sofia/gateway/foo/123456789 9664
execute_on_pre_answer
Execute an application (not an api) when the called party "preanswers" - that is, when some form of early media is coming or the far end sends a 180 Ringing.
The command is executed only on channels that are not already answered. Just use export or export with nolocal: prefix to make sure it is executed when b-leg answers.
In the second usage example below, we have originated an outbound call to a local extension, where we will wait 30 seconds without ignoring media. In this case we use 'set' and not 'export'.
Usage:
```xml
## execute_on_ring
Execute a command when the called party rings.
Usage:
<action application="set" data="nolocal:execute_on_ring=lua markring ${uuid}"/>
execute_on_sip_extra_headers
Usage:
Avi Marcus - could this be used to send an incoming LRN header (e.g. alcazar) to a non-blocking curl or lua script (for caching)? Or would api_on_sip_extra_headers be good for that?
execute_on_sip_reinvite
Execute a command when SIP Reinvite.
Usage:
Add description here.
export_vars
export_vars lists variables to be exported to the other leg upon bridge. Unlike export, it only lists the variables to export without actually setting them. Note: This is useful to transfer information from a-leg (INVITE) to the future b-leg of REFER.
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_dptools | mod_dptools.c | 5118 |
core | switch_core_session.c | 4796 |
EXTEN
Description needed! Please contribute one.
Usage:
Example needed! Please contribute one.
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_dialplan_asterisk | mod_dialplan_asterisk.c | 6205 |
fail_on_single_reject
This is useful when using the "," AND operator in the DATA field of a bridge. The AND operator notifies a list of destinations, bridging to the first destination that accepts the call. Typically if a destination in the list rejects the call, the bridge will continue to be attempted until either another destination accepts the call, or a timeout occurs.
This variable allows one to terminate the bridging attempt on a single rejection of the call. This means the bridge attempt would fail, and if continue_on_fail has not been set, the call is terminated. This variable would be set within a condition before a bridge application. When used in conjunction with the continue_on_fail variable, one can perform operations such as rolling over a rejected caller to an answering machine application.
The default setting is FALSE, meaning a single rejection will not terminate the bridging attempt.
It can also be set to a list of failure causes to stop on, and can be negated to a list of causes not to stop on (i.e. stop on all other failure causes).
Usage:
<action application="set" data="fail_on_single_reject=true"/> <action application="bridge" data="sofia/$${profile}/$${kitchen}%$${domain},sofia/$${profile}/$${dining}%$${domain}"/> <action application="javascript" data="answermachine.js"/>
or,
<action application="set" data="fail_on_single_reject=USER_BUSY"/>
or,
<action application="set" data="fail_on_single_reject=!NORMAL_CIRCUIT_CONGESTION"/>
or to use a list,
<action application="set" data="fail_on_single_reject=^^:CALL_REJECTED:NORMAL_CLEARING:USER_BUSY"/>
or for negated list,
<action application="set" data="fail_on_single_reject=!^^:ALLOTTED_TIMEOUT:NETWORK_OUT_OF_ORDER"/>
failed_xml_cdr_prefix
If you set that on the A leg and any and all failed B originates generate a full XML CDR report and set it as a variable, this includes during a forked dial.
So say you try to call sofia/profile/a@xxxxxxx,sofia/profile/b@xxxxxxx
And it fails completely, before you make the call you set failed_xml_cdr_prefix to "bad_call"
Then you end up with ${bad_call_1} and ${bad_call_2} which are each a full XML report including all the vars etc.
Usage:
<action application="set" data="failed_xml_cdr_prefix=failinggw" />
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
core | switch_ivr_originate.c | 8730 |
failure_causes
Controls which failure causes will be considered as a failure to the bridge(s). This will change the values for which continue_on_fail will fail by default unless continue_on_fail is set to true.
Usage:
<action application="set" data="failure_causes=USER_BUSY,NO_ANSWER"/>
or Q.850 cause codes,
<action application="set" data="failure_causes=487"/>
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_dptools | mod_dptools.c | 12058 |
fax_bad_rows
Description needed! Please contribute one.
Usage:
Example needed! Please contribute one.
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_spandsp | mod_spandsp_fax.c | 3026e639fe0b8cfec9a37f2ce99aee7779b9e736 |
fax_disable_v17
fax_disable_v17 prevents the use of the V.17 (9,600bps to 14,400bps) FAX modem. This means FAXes will be limited to the use of V.29 (9,600bps and 7,200bps) and V.27ter (4,800bps). Some VoIP systems handle V.17 so poorly there are often good reasons to want to disable the use of this modem.
Usage:
Example needed! Please contribute one.
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_spandsp | mod_spandsp_fax.c | 3026e639fe0b8cfec9a37f2ce99aee7779b9e736 |
fax_document_total_pages
Description needed! Please contribute one.
Usage:
Example needed! Please contribute one.
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_spandsp | mod_spandsp_fax.c | 3026e639fe0b8cfec9a37f2ce99aee7779b9e736 |
fax_ecm_requested
Description needed! Please contribute one.
Usage:
Example needed! Please contribute one.
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_spandsp | mod_spandsp_fax.c | 3026e639fe0b8cfec9a37f2ce99aee7779b9e736 |
fax_ecm_used
Description needed! Please contribute one.
Usage:
Example needed! Please contribute one.
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_spandsp | mod_spandsp_fax.c | 3026e639fe0b8cfec9a37f2ce99aee7779b9e736 |
fax_end_page
Description needed! Please contribute one.
Usage:
Example needed! Please contribute one.
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_spandsp | mod_spandsp_fax.c | 3026e639fe0b8cfec9a37f2ce99aee7779b9e736 |
fax_filename
Description needed! Please contribute one.
Usage:
Example needed! Please contribute one.
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_spandsp | mod_spandsp_fax.c | 3026e639fe0b8cfec9a37f2ce99aee7779b9e736 |
fax_force_caller
Description needed! Please contribute one.
Usage:
Example needed! Please contribute one.
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_spandsp | mod_spandsp_fax.c | 3026e639fe0b8cfec9a37f2ce99aee7779b9e736 |
fax_ident
Description needed! Please contribute one.
Usage:
Example needed! Please contribute one.
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_spandsp | mod_spandsp_fax.c | 3026e639fe0b8cfec9a37f2ce99aee7779b9e736 |
fax_image_resolution
Description needed! Please contribute one.
Usage:
Example needed! Please contribute one.
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_spandsp | mod_spandsp_fax.c | 3026e639fe0b8cfec9a37f2ce99aee7779b9e736 |
fax_image_size
Description needed! Please contribute one.
Usage:
Example needed! Please contribute one.
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_spandsp | mod_spandsp_fax.c | 3026e639fe0b8cfec9a37f2ce99aee7779b9e736 |
fax_local_station_id
Description needed! Please contribute one.
Usage:
Example needed! Please contribute one.
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_spandsp | mod_spandsp_fax.c | 3026e639fe0b8cfec9a37f2ce99aee7779b9e736 |
fax_prefix
Description needed! Please contribute one.
Usage:
Example needed! Please contribute one.
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_spandsp | mod_spandsp_fax.c | 3026e639fe0b8cfec9a37f2ce99aee7779b9e736 |
fax_remote_station_id
Description needed! Please contribute one.
Usage:
Example needed! Please contribute one.
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_spandsp | mod_spandsp_fax.c | 3026e639fe0b8cfec9a37f2ce99aee7779b9e736 |
fax_result_code
Result Code | Message |
---|---|
0 | OK |
2 | Timed out waiting for initial communication |
3 | Timed out waiting for the first message |
5 | The HDLC carrier did not stop in a timely manner |
6 | Failed to train with any of the compatible modems |
13 | Unexpected message received |
14 | Received bad response to DCS or training |
15 | Received a DCN from remote after sending a page |
17 | Received a DCN while waiting for a DIS |
20 | Received no response to DCS or TCF |
23 | Invalid ECM response received from transmitter |
31 | Timer T2 expired while waiting for fax page |
32 | Timer T2 expired while waiting for next fax page |
48 | Disconnected after permitted retries |
49 | The call dropped prematurely |
Usage:
Example needed! Please contribute one.
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_spandsp | mod_spandsp_fax.c | 3026e639fe0b8cfec9a37f2ce99aee7779b9e736 |
fax_result_text
Description needed! Please contribute one.
Usage:
Example needed! Please contribute one.
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_spandsp | mod_spandsp_fax.c | 3026e639fe0b8cfec9a37f2ce99aee7779b9e736 |
fax_success
Description needed! Please contribute one.
Usage:
Example needed! Please contribute one.
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_spandsp | mod_spandsp_fax.c | 3026e639fe0b8cfec9a37f2ce99aee7779b9e736 |
fax_transfer_rate
Description needed! Please contribute one.
Usage:
Example needed! Please contribute one.
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_spandsp | mod_spandsp_fax.c | 3026e639fe0b8cfec9a37f2ce99aee7779b9e736 |
fax_use_ecm
Description needed! Please contribute one.
Usage:
Example needed! Please contribute one.
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_fax | mod_fax.c | 9468 |
fax_v17_disabled
Description needed! Please contribute one.
Usage:
Example needed! Please contribute one.
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_spandsp | mod_spandsp_fax.c | 3026e639fe0b8cfec9a37f2ce99aee7779b9e736 |
fax_verbose
Description needed! Please contribute one.
Usage:
Example needed! Please contribute one.
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_spandsp | mod_spandsp_fax.c | 3026e639fe0b8cfec9a37f2ce99aee7779b9e736 |
fifo_announce
Description needed! Please contribute one.
Usage:
Example needed! Please contribute one.
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_fifo | mod_fifo.c | 8689 |
fifo_bridged
Description goes here
Usage:
Example needed
fifo_caller_consumer_import
Import list of variables from the caller to the consumer.
Usage:
<action application="set" data="fifo_caller_consumer_import=var1,var2"/>
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_fifo | mod_fifo.c | 12722 |
fifo_caller_exit_key
Description needed! Please contribute one.
Usage:
Example needed! Please contribute one.
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_fifo | mod_fifo.c | 8026 |
fifo_chime_freq
Description needed! Please contribute one.
Usage:
Example needed! Please contribute one.
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_fifo | mod_fifo.c | 8026 |
fifo_chime_list
Description needed! Please contribute one.
Usage:
Example needed! Please contribute one.
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_fifo | mod_fifo.c | 8026 |
fifo_consumer_caller_import
Import list of variables from the consumer to the caller
Usage:
<action application="set" data="fifo_consumer_caller_import=var1,var2"/>
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_fifo | mod_fifo.c | 12722 |
fifo_consumer_exit_key
Description needed! Please contribute one.
Usage:
Example needed! Please contribute one.
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_fifo | mod_fifo.c | 9043 |
fifo_consumer_id
Description needed! Please contribute one.
Usage:
Example needed! Please contribute one.
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_fifo | mod_fifo.c | 8026 |
fifo_consumer_wrapup_key
Description needed! Please contribute one.
Usage:
Example needed! Please contribute one.
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_fifo | mod_fifo.c | 7961 |
fifo_consumer_wrapup_sound
Description needed! Please contribute one.
Usage:
Example needed! Please contribute one.
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_fifo | mod_fifo.c | 7961 |
fifo_consumer_wrapup_time
Description needed! Please contribute one.
Usage:
Example needed! Please contribute one.
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_fifo | mod_fifo.c | 9860 |
fifo_hold_music
Description needed! Please contribute one.
Usage:
Example needed! Please contribute one.
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_fifo | mod_fifo.c | 8065 |
fifo_manual_bridged
Description goes here
fifo_member_wait
If it's set to 'wait', then the consumer's leg of the call will not hangup when the caller hangs up [default].
If it's set to 'nowait' then the consumer's leg of the call will hangup when the caller hangs up.
Usage:
<action application="set" data="result=${fifo_member(add MyQueName {fifo_member_wait=nowait}user/1001@VoiceNetwork.ca )"/>
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_fifo | mod_fifo.c | 12782 |
fifo_orbit_announce
Description needed! Please contribute one.
Usage:
Example needed! Please contribute one.
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_fifo | mod_fifo.c | 8026 |
fifo_orbit_context
Sets the context for the fifo_orbit_exten when the que times out. This must be set before you place the caller in the que.
Usage:
<action application="set" data="fifo_orbit_context=MyContext"/>"
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_fifo | mod_fifo.c | 12782 |
fifo_orbit_exten
Set a destination extension and timeout, and it will cause FreeSWITCH to run the caller through the FIFO until the specified timeout, then the caller will get transferred to the destination extension. Can be set before launching the caller into the FIFO.
Usage: Use before a FIFO statement:
<action application="set" data="fifo_orbit_exten=MyFIFOVoicemail:60"/>
This will cause 60 seconds of time to pass with the caller in the FIFO before sending the call to the Voicemail extension specified.
<action application="fifo" data="MyFIFO in"/>
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_fifo | mod_fifo.c | 8026 |
fifo_outbound_uuid
Description needed! Please contribute one.
Usage:
Example needed! Please contribute one.
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_fifo | mod_fifo.c | 10277 |
fifo_override_announce
Description needed! Please contribute one.
Usage:
Example needed! Please contribute one.
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_fifo | mod_fifo.c | 8014 |
fifo_pop_order
Description needed! Please contribute one.
Usage:
Example needed! Please contribute one.
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_fifo | mod_fifo.c | 8689 |
fifo_position
Description goes here
Usage:
Example needed
See also:
Implemented By:
mod_fifo.c
fifo_record_template
Description needed! Please contribute one.
Usage:
Example needed! Please contribute one.
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_fifo | mod_fifo.c | 8039 |
fifo_role
For reporting purposes, i.e. in the CDRs, the variable will contain "consumer" or "caller" depending upon the call leg.
Usage:
None
See also:
Mod_fifo
fifo_timestamp
Contains the timestamp of when the caller was bridged to the consumer.
Usage:
None
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_fifo | mod_fifo.c | 8689 |
force_transfer_context
When handling transfer/REFER FreeSWITCH normally inherits the context from the original channel. This variable forces the context in which to handle the transfer/REFER
Usage:
<action application="bridge" data="{force_transfer_context=some_context}sofia/gateway/gw_name/12345"/>
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
core | switch_ivr.c | 10466 |
force_transfer_dialplan
When handling transfer/REFER FreeSWITCH normally inherits the diaplan from the original channel. This variable forces the dialplan in which to handle the transfer/REFER
Usage:
Example needed! Please contribute one.
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
core | switch_ivr.c | 10466 |
group_confirm_cancel_timeout
If set, cancels a leg timeout after the call is answered.
When using group confirm, a call passes through three phases:
- Call is ringing.
- Call is answered, waiting to be confirmed.
- Call is confirmed and bridged.
Normally, a timeout on the leg will apply to phases 1 and 2, but the example below would apply the leg timeout only during phase 1
Usage:
<action application="set" data="group_confirm_cancel_timeout=1"/>
group_confirm_file
This variable is used together with group_confirm_key. In group_confirm_file, you specify the wav file you want to play when the called party picks up the call. In the group_confirm_key variable, you define the DTMF that the called party should send to FS to bridge the call. If a wrong DTMF or no DTMF is sent, the called won't be bridged and the wav file will be repeated. Although this is the standard usage of these variables (group_confirm_key and group_confirm_file), they can be used in a more flexible manner. Please see Freeswitch_IVR_Originate#Answer_confirmation.
Usage:
<action application="set" data="group_confirm_file=/usr/local/freeswitch/sounds/take_call_question.wav" /> <action application="set" data="group_confirm_key=1" />
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
core | switch_ivr_originate.c | 4796 |
group_confirm_key
When using group_confirm_file to play a file asking the user if they want to actually accept the call, use the group_confirm_key for which key to bind to to actually connect the call.
Usage:
<action application="set" data="group_confirm_key=1"/>
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
core | switch_ivr_originate.c | 4796 |
hangup_after_bridge
Controls what happens to a calling (A) party when in a bridge state and the called (B) party hangs up. If "true" the dialplan will stop processing and the A leg will be terminated when the B leg terminates. If "false" (default) the dialplan continues to be processed after the B leg terminates. This is checked after park_after_bridge and transfer_after_bridge.
Usage:
<action application="set" data="hangup_after_bridge=true"/>
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_dptools | mod_dptools.c | 8412 |
mod_enum | mod_enum.c | 3495 |
mod_lcr | mod_lcr.c | 10510 |
core | switch_ivr.c | 5738 |
core | switch_ivr_bridge.c | 10504 |
hangup_after_conference
Controls what happens to a calling (A) party when in a conference and the conference ends (e.g. endconf flag set and moderator leaves). If "true" (default) the dialplan will stop processing and the A leg will be terminated. If "false" the dialplan continues to be processed after the end of conference.
Usage:
<action application="set" data="hangup_after_conference=false"/>
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_conference | mod_conference.c | 7a7f386886a069 |
hangup_cause
This is set to the hangup cause of the A leg of the call (note that as such it doesn't make much sense before the end of the call). Often this will take the hangup cause from the B leg of the call, if there is one. For a list of hangup causes, see Hangup Causes.
Usage:
<action application="log" data="1 A-leg hangup cause: ${hangup_cause}"/>
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
core | switch_core_state_machine.c | 12784 |
hangup_cause_q850
This is set to the Q850 numeric code of the hangup cause of the A leg of the call (note that as such it doesn't make much sense before the end of the call). Often this will take the hangup cause from the B leg of the call, if there is one. For a list of hangup causes, see Hangup Causes.
Usage:
<action application="log" data="1 A-leg hangup Q850 cause: ${hangup_cause_q850}"/>
hold_events
It's a variable that display start and stop times for each hold.
Example:
This CDR shows that the phone was put on hold twice with hold start and stop time.
variable_hold_events: [{{1347487292379229,1347487293856872},{1347487288539686,1347487290757780}}]
hold_hangup_xfer_exten
Controls what happens to a calling (A) party when in a bridge state and the bridge ends while the called (B) party is on hold. If not set on leg B (ie. the default), then A leg is hung up. If it is set on leg B, then leg A is transferred to the given extension, as per transfer_after_bridge.
Usage:
<action application="set" data="hold_hangup_xfer_exten=1000:XML:default"/>
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
core | switch_ivr_bridge.c | 10 Dec 09 |
hold_music
Per-channel hold music. Supports all audio formats and audio streams. The hold_music variable can also be set globally at vars.xml.
Usage:
<action application="set" data="hold_music=/sounds/holdmusic.wav" />
You can also set your hold_music to the special value "indicate_hold" instead of a music source and it will pass the hold req through but not the SDP.
or,
<action application="set" data="hold_music=silence" />
For multi-tenant environment, if you want to have a separate MOH for the phone with hold button (like Polycom) that utilizes RE-INVITE with no media ip addr (0.0.0.0) for hold, you can override the hold-music values in the sip profile parameter similar to the following example:
<action application="bridge_export" data="hold_music=$${sounds_dir}/music/company-a.mp3"/>
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_fifo | mod_fifo.c | 8065 |
mod_sofia | sofia_glue.c | 5449 |
core | switch_ivr.c | 8232 |
core | switch_ivr_play_say.c | 8065 |
ignore_display_updates
Tells freeswitch not to send display UPDATEs to the leg of the call. (update_display)
Usage: From dialplan/default.xml:
To set on A-Leg
<action application="set" data="ignore_display_updates=true"/>
To set on B-Leg
<action application="bridge" data="{ignore_display_updates=true}sofia/gateway/provider/18005551212"/>
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_sofia | mod_sofia.c | ?? |
mod_sofia | sofia.c | ?? |
ignore_early_media
Default value is false. Controls if the call returns on early media or not. You may specify a value for ignore_early_media in the argument to the bridge application, using the { } syntax. (ignore_early_media may not be specified on a per-leg basis, using the [ ] syntax, as it specifically is a global variable to the originate session.)
Usage:
<action application="set" data="ignore_early_media=true"/>
or,
<action application="bridge" data="{ignore_early_media=true}sofia/test-int/1001@somebox,sofia/test-int/1000@somehost"/>
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
core | switch_ivr_originate.c | 4796 |
import
The import variable, when used before a bridge, imports the variables of the other channel on the actual channel.
Usage:
<action application\="set" data\="import=this_is_a_variable_name"/> |
---|
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_lcr | mod_lcr.c | 11976 |
inherit_codec
If late negotiation is on, and you set inherit_codec=true on the A leg, the negotiated codec of the B leg will be forced onto the A leg.
Usage:
<action application="set" data="inherit_codec=true"/>
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
core | switch_ivr_originate.c | 11256 |
intercept_unanswered_only
If set to true, the leg will only be intercepted if the channel is not answered.
Default: false
Usage:
<action application="set" data="intercept_unanswered_only=true"/> <action application="intercept" data="myUUID"/>
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
core | switch_ivr_bridge.c | git-36ba0f24 |
intercept_unbridged_only
If set to true, the leg will only be intercepted if the channel is not bridged to anyone.
Default: false
Usage:
<action application="set" data="intercept_unbridged_only=true"/> <action application="intercept" data="myUUID"/>
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
core | switch_ivr_bridge.c | git-58fe45a |
ivr_menu_status
ivr_menu_status, is a channel variable with options of success, failure or timeout
variable_ivr_menu_status: [success] variable_ivr_menu_status: [failure] variable_ivr_menu_status: [timeout]
ivr_menu_terminator
You can set to none or the dtmf chars you want to terminate input.
Usage:
<action application="set" data="ivr_menu_terminator=#"/>
jitterbuffer_msec
Activates the jitter buffer. The jitter buffer has three params: length, max length, and max drift.
Usage:
<action application="set" data="jitterbuffer_msec=60:200:20"/> <action application="answer"/>
Or to set it on the subsequent outbound call: export sets a variable on both the current channel and on any channels it creates, the 'nolocal:' disables setting it on the current channel and only sets it on the subsequent outbound channels.
<action application="export" data="nolocal:jitterbuffer_msec=60"/> <action application="bridge" data="sofia/default/888@conference.freeswitch.org"/>
You can also activate the Jitter Buffer in the bridge as follows:
<action application="bridge" data="{jitterbuffer_msec=60}sofia/gateway/$1@gateway.com"/>
This will add a jitter buffer to packets flowing from a remote gateway towards a local freeswitch user. The network would look like this:
(local sip user) -----> FreeSWITCH -----> (remote gateway)
Where the link between the freeswitch and the remote gateway has a bad, jitter causing connection, and say the local sip user has no jitter buffering on their IP-phone. This will help the voice quality for the incoming audio.
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_fax | mod_fax.c | 9468 |
mod_sofia | sofia_glue.c | 5114 |
last_bridge_hangup_cause
This is set to the hangup cause of the last bridged B leg of the call. For a list of hangup causes, see Hangup Causes.
Usage:
<action application="log" data="B-leg hangup cause: ${last_bridge_hangup_cause}"/>
last_bridge_proto_specific_hangup_cause
This shows the last bride hangup cause by SIP response code, e.g. "sip:404"
last_matching_digits
Contains the last set of digits that the user dialed when using the "dmachine" digit-handling. This is most commonly used with the bind_digit_action dialplan application.
Usage:
<action application="log" data="INFO User just dialed ${last_matching_digits}"/>
See also:
Implemented By:
Module Name | Source File | Last Updated |
---|---|---|
mod_dptools | mod_dptools.c |
last_transferred_conference
Contains the name of the last conference that this channel was connected to.
Usage:
<action application="log" data="INFO Last conference this person visited was [${last_transferred_conference}]"/>
See also:
Implemented By:
Module Name | Source File | Last Updated |
---|---|---|
mod_conference | mod_conference.c |
leg_delay_start
You can specify a wait time for before each leg is called in a forked dial scenario. Useful for follow me dialplans. Leg delay start will ring an extension early if all other less delayed (or no delayed) legs have declined, failed, or are not available (as of FS 1.0.5). Leg_delay_start is best thought of as the minimum time to ring the other extensions prior to this one if they are able to ring.
Note that this doesn't work with Enterprise Originate. For Enterprise originate see variable originate_delay_start
Usage:
<action application="bridge" data="sofia/profile/dest1,[leg_delay_start=10]sofia/profile/dest2,[leg_delay_start=15]sofia/profile/dest3"/>
A more complex example with breakdown and timeline (seconds in brackets):
<action application="bridge","users/1000,[leg_delay_start=8]user/2302,[leg_delay_start=20]sofia/gateway/flowroute/1231231234"/>
Assuming all users just let it ring:
[00] - user 1000 rings [08] - user 2302 rings [20] - user 1231231234 rings
Assuming user 1000 decline after 2 seconds, other users ring:
[00] - user 1000 rings [02] - user 2302 rings [14] - user 1231231234 rings
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
core | switch_ivr_originate.c | 12403 |
leg_progress_timeout
Description needed! Please contribute one.
Usage:
Make all bridged calls fail over to the next in 6 seconds. <action application="export" data="leg_progress_timeout=6"/>
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
core | switch_ivr_originate.c | 11286 |
leg_timeout
Timeout for each leg in an originate dialstring. Can be used in per-leg [], but not in global {} for the dialstring. For global, use originate_timeout.
You can also use leg_progress_timeout to specify the maximum time we will wait before we get media (whether its early media, ringing or answer), allowing you to avoid going to voicemail for a particular line.
If you are using group confirm then you can cancel the timeout by using the group_confirm_cancel_timeout channel variable. If leg_delay_start is also used, leg_timeout will not start the timeout counter until after the extension starts to be bridged to.
Usage:
<action application="bridge" data="[leg_timeout=15]user/hastoanswerquickly/some.domain.com,[leg_timeout=60]user/hasaminutetoanswer@some.domain.com"/>
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
core | switch_ivr_originate.c | 11286 |
loopback_bowout_on_execute
Set to true to have one-legged loopback channels "bow out" of the call.
Usage:
<action application="set" data="loopback_bowout_on_execute=true"/>
See also:
Freeswitch mailing list
bowout_on_execute is only useful to 1 leg calls you never have to set it. loopback_bowout is true by default it tries to cut loopback out by doing uuid_bridge
Comment: When setting "loopback_bowout=false", a bridged loopback call result in 4 legs (a-leg, loopback-a, loopback-b, b-leg).
When setting "loopback_bowout=true", a bridged loopback call results in 2 legs (a-leg, b-leg).
During call setup with "loopback_bowout=true", there will always be 4 legs however at the beginning. loopback-a and loopback-b will be destroyed when a-leg and b-leg are successfully bridged
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_loopback | mod_loopback.c |
loopback_export
A list of channel variables to pass from loopback-a to loopback-b.
Usage:
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_loopback | mod_loopback.c |
media_bug_answer_req
Start recording only when the channel has been answered.
NOTE: RECORD_ANSWER_REQ should be used on releases prior to 1.0.5, or builds prior to the revision 15235
Usage:
<action application="set" data="media_bug_answer_req=true"/>
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
core | switch_ivr_async.c | 15235 |
core | switch_core_media_bug.c | 15235 |
min_dup_digit_spacing_ms
Duplicate inband DTMF that starts sooner than this time will be ignored. That is, this is the minimum gap from the end of the first digit and the start of the repeated digit required for two digits to be detected. This value is 0 by default. Set this variable prior to executing spandsp_start_dtmf.
Usage:
<action application="set" data="min_dup_digit_spacing_ms=40"/> <action application="spandsp_start_dtmf" />
See also:
- Channel Variables
- Mod_spandsp
Implemented By:
Module Name | Source File | Last Updated |
---|---|---|
mod_spandsp | mod_spandsp_dsp.c |
monitor_early_media_fail
Monitors early media for failure conditions, such as a busy signal. This allows faster processing of failed calls when ignoring early media.
The syntax is a series of ! delimited early media conditions in the following format:
condition_name:number_of_hits:tone_detect_frequencies
condition_name | user defined name for the error condition |
---|---|
number_of_hits | the number of times the tone must be heard before considering it a fail |
tone_detect_frequencies | the frequencies to listen for (delimited by + instead of ,). See tone_detect |
NOTE: this variable only works when ignore_early_media is set to true.
Usage:
<action application="bridge" data="{ignore_early_media=true,monitor_early_media_fail=user_busy:2:480+620!destination_out_of_order:2:1776.7}sofia/dial/string"/>
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
core | switch_ivr_originate.c | 11295 |
monitor_early_media_ring
Monitors early media for a user-specific ring tone. Each time the tone is heard, the switch will increment an internal counter for that leg. Once the counter reaches monitor_early_media_ring_total (or this variable has not been set) then the early media will be sent.
The syntax is a series of ! delimited early media conditions in the following format:
condition_name:number_of_hits:tone_detect_frequencies
condition_name | Optional? user-defined name for the error condition |
---|---|
number_of_hits | the number frequencies for the tone detector to find before considering it a hit. 1:400.0+480.0 means the ring count is incremented if 400hz OR 480hz is detected. 2:400.0+480.0 means the ring count is incremented if 400 hz AND 480 hz are detected. |
tone_detect_frequencies | the frequencies to listen for (delimited by + instead of ,). Examples are 400.0+480.0 [for a US Ring] See tone_detect |
NOTE: this variable only works when ignore_early_media is not present.
Usage:
<action application="bridge" data="{monitor_early_media_ring_total=3,monitor_early_media_ring=usring:1:440.0+480.0!ukring:2:400+450}sofia/gateway/yourgateway/1239@conference.freeswitch.org"/>
This will bridge to a special conference that rings (US_RING) 10 times, and then plays the Star Wars Imperial March. You may optionally call 1239a@conference.freeswitch.org to test a UK_RING. A successful log looks like:
2010-10-22 10:26:53.975865 [DEBUG] switch_ivr_originate.c:614 sofia/internal/1239@conference.freeswitch.org setting ring total to 3 2010-10-22 10:26:55.007970 [DEBUG] switch_rtp.c:2544 Correct ip/port confirmed. 2010-10-22 10:26:56.427869 [DEBUG] switch_ivr_async.c:2424 TONE monitor_early_media_ring_2 HIT 1/2 2010-10-22 10:27:01.707869 [DEBUG] switch_ivr_async.c:2424 TONE monitor_early_media_ring_2 HIT 1/2 2010-10-22 10:27:02.667868 [DEBUG] switch_ivr_async.c:2424 TONE monitor_early_media_ring_2 HIT 2/2 2010-10-22 10:27:02.667868 [DEBUG] switch_ivr_async.c:2430 TONE monitor_early_media_ring_2 DETECTED 2010-10-22 10:27:02.667868 [DEBUG] switch_ivr_originate.c:352 Ring 1/3 2010-10-22 10:27:02.667868 [DEBUG] switch_ivr_async.c:2436 Re-enabling monitor_early_media_ring_2 2010-10-22 10:27:07.767869 [DEBUG] switch_ivr_async.c:2424 TONE monitor_early_media_ring_2 HIT 1/2 2010-10-22 10:27:13.487868 [DEBUG] switch_ivr_async.c:2424 TONE monitor_early_media_ring_2 HIT 1/2 2010-10-22 10:27:14.027868 [DEBUG] switch_ivr_async.c:2424 TONE monitor_early_media_ring_2 HIT 2/2 2010-10-22 10:27:14.027868 [DEBUG] switch_ivr_async.c:2430 TONE monitor_early_media_ring_2 DETECTED 2010-10-22 10:27:14.027868 [DEBUG] switch_ivr_originate.c:352 Ring 2/3 2010-10-22 10:27:14.027868 [DEBUG] switch_ivr_async.c:2436 Re-enabling monitor_early_media_ring_2 2010-10-22 10:27:20.147869 [DEBUG] switch_ivr_async.c:2424 TONE monitor_early_media_ring_2 HIT 1/2 2010-10-22 10:27:20.907868 [DEBUG] switch_ivr_async.c:2424 TONE monitor_early_media_ring_2 HIT 2/2 2010-10-22 10:27:20.907868 [DEBUG] switch_ivr_async.c:2430 TONE monitor_early_media_ring_2 DETECTED 2010-10-22 10:27:20.907868 [DEBUG] switch_core_media_bug.c:441 Removing BUG from sofia/internal/1239@conference.freeswitch.org 2010-10-22 10:27:20.911864 [INFO] switch_ivr_originate.c:3290 Sending early media
The last line is most important. The call continues, but early media is delayed from sending until monitor_early_media_ring_total is matched. If monitor_early_media_ring_total is not met, then the call will timeout.
anthm had the following to say in IRC
<anthm> The point of the feature is because of problems with providers who send a 183 then play a busy tone and do not indicate fail over sip <anthm> so it looks like a good call. <anthm> so the monitor is to confirm that it is a ring tone as expected.
<anthm> monitor_early_media_ring stops the default behavior to have originate return when it hears a ring <anthm> well when it gets 183 <anthm> cos you don't know if its a ring <anthm> so it listens for one <anthm> when it really hears one, it lets the channel accept the 183
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
core | switch_ivr_originate.c | 11295 |
monitor_early_media_ring_total
Specifies the number of user defined ring tones that can be heard before failing. To be used with monitor_early_media_ring
Usage:
<action application="bridge" data="{ignore_early_media=true,monitor_early_media_ring_total=3,monitor_early_media_ring=us_ring:1:440.0+480.0}sofia/dial/string"/>
See also:
- monitor_early_media_ring
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
core | switch_ivr_originate.c | 11738 |
monitor_fail_dispo
This variable can be used to provide a custom originate_disposition as the result of an early media failure using monitor_early_media_fail or monitor_early_media_ring. If this variable is not set, a default value of 'monitor_early_media_fail' or 'monitor_early_media_ring' will be placed inoriginate_disposition
Usage:
<action application="bridge" data="{ignore_early_media=true,monitor_fail_dispo=strange_bleep_attack,monitor_early_media_fail=user_busy:2:1234}sofia/dial/string"/>
See also:
- Early Media
- monitor_early_media_fail
- monitor_early_media_ring
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
core | switch_ivr_originate.c | 11437 |
no_throttle_limits
You set this variable to true on your outbound calls to have them not count in sps
Usage:
{no_throttle_limits=true}
originate_delay_start
You can specify a wait time in milliseconds before origination. This variable can be used in Enterprise originate where the variable leg_delay_start cannot be used.
Usage:
<action application="bridge" data="sofia/profile/dest1::{originate_delay_start=10000}sofia/profile/dest2::{originate_delay_start=15000}sofia/profile/dest3"/>
A more complex example with breakdown and timeline (seconds in brackets):
<action application="bridge","users/1000::{originate_delay_start=8000}user/2302::{originate_delay_start=14000}sofia/gateway/flowroute/1231231234"/>
Assuming all users just let it ring:
[00] - user 1000 rings [10] - user 2302 rings [15] - user 1231231234 rings
Assuming user 1000 decline after 2 seconds, other users ring:
[00] - user 1000 rings [08] - user 2302 rings [14] - user 1231231234 rings
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
core | switch_ivr_originate.c | 032761c |
originate_disposition
Read Only. This is the originate disposition aka hangup cause returned. (LEG B)
It is also redefined after every bridge attempts if the bridge is not successful.
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
core | switch_ivr_originate.c | 11437 |
originate_timeout
Determines how long a bridge or originate action action will stay in the "originate" state. In effect, it is a way to control the timeout for a bridge/originate consisting of multiple endpoints. Default value is 60.
Usage:
<action application="bridge" data="{originate_timeout=10}[leg_timeout=5]sofia/default/foo1@bar1|[leg_timeout=5]sofia/default/foo2@bar2"/>
WARNING: Beware that if you are not using {ignore_early_media=true} call_timeout is no longer applicable as soon as early media signal is received.
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
core | switch_ivr_originate.c |
originating_leg_uuid
Shows the uuid of the originating leg on an outbound channel
Usage:
In A-leg CDR:
<uuid>cb5f5b90-75a0-11e0-873b-d1cba9e0f1b8</uuid> <call_uuid>cb5f5b90-75a0-11e0-873b-d1cba9e0f1b8</call_uuid>
In B-leg CDR:
<uuid>cb8633aa-75a0-11e0-873d-d1cba9e0f1b8</uuid> <call_uuid>cb5f5b90-75a0-11e0-873b-d1cba9e0f1b8</call_uuid> <originating_leg_uuid>cb5f5b90-75a0-11e0-873b-d1cba9e0f1b8</originating_leg_uuid>
Note that the leg uuid's are different. The call_uuid matches the two legs together, but the originating_leg_uuid can do so as well.
See also:
Implemented By:
Module Name | Source File | Last Updated |
---|---|---|
Core | switch_ivr_originate.c |
origination_callee_id_name
Set on the inbound leg to control what caller ID name appears in the caller phone's display. Also see ignore_display_updates which affect the processing of these variables.
Usage:
<action application="bridge" data="{origination_callee_id_name=Reginald}sofia/gateway/provider/<Reginald's cellphone number>" />
If you find using set doesn't work, try using export instead.
origination_callee_id_number
Set on the inbound leg to control what caller ID number appears in the caller phone's display. Also see ignore_display_updates which affect the processing of these variables.
Usage:
<action application="bridge" data="{origination_callee_id_name=Reginald,origination_callee_id_number=2332}sofia/gateway/provider/<Reginald's cellphone number>" />
origination_caller_id_name
Sets the origination callerid name (LEG A).
If you want to set the Caller ID on an origination call you should add this inside the {} brackets before the dialstring. Set it to "_undef_" if you want to remove it. Setting it to a null string \"\" may break your system.
For Snom 370/820 users:
If you want to display LEG A's name (if available) as soon as LEG B (here the local Snom) rings, you have to set origination_caller_id_name or effective_caller_id_name as described. Otherwise, in LEG B's display, you gonna see LEG A's number during ringing and switching to LEG A's name after picking up the call by LEG B.
Usage:
originate {origination_caller_id_name='Caller Name',origination_caller_id_number=5551231234}sofia/gateway/test/1231231234 &park()
See also:
- effective_caller_id_name
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_dptools | mod_dptools.c | 10917 |
core | switch_ivr_originate.c | 12604 |
origination_caller_id_number
Sets the origination callerid number. (LEG A)
If you want to set the Caller ID on an origination call you should add this inside the {} brackets before the dialstring.
Usage:
originate {origination_caller_id_name='Caller Name',origination_caller_id_number=5551231234}sofia/gateway/test/1231231234 &park()
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_dptools | mod_dptools.c | 10917 |
core | switch_ivr_originate.c | 12604 |
origination_channel_name
Set this in the {} when doing an originate to create a custom channel name
Usage:
originate {origination_channel_name='this_is_my_channel_name'}loopback/9664 9195
See also:
- Mod_commands#originate
Implemented By:
Module | Source File | Last Updated |
---|---|---|
core | switch_ivr_originate.c |
origination_privacy
Sets privacy profile for caller. Options are any combination of "screen", "hide_name", "hide_number". Screen is on by default.
Note: screen is the keyword that makes the caller ID as P-Asserted-Identity vs P-Preferred-Identity.
Usage:
<action application="set" data="origination_privacy=hide_name"/>
To do 3 at once:
Usage:
<action application="set" data="origination_privacy=hide_name+hide_number+screen"/> <action application="set" data="origination_privacy=hide_name:hide_number:screen"/>
Note: There is no real separator; therefore, you can use separator to make it readable or nothing.
See also:
Channel_Variables#sip_cid_type
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
core | switch_ivr_originate.c | 12604 |
originator_codec
Sets the codec for calls originated from LEG A (setting the codec for LEG B). This will automatically be appended to the codec_string unless an absolute_codec_string has been set.
Usage:
<action application="set" data="originator_codec=PCMU"/>
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_opal | mod_opal.cpp | 10567 |
mod_sofia | sofia_glue.c | 5114 |
core | switch_core_session.c | 4796 |
outbound_redirect_fatal
When doing a simultaneous call to multiple endpoints, a 302 redirect can cause all the endpoints to stop ringing and the call will follow the redirect. When this channel variable is set it causes an endpoint that sends back a 302 redirect to be removed from the call list and the other endpoints continue to ring.
Usage:
<action application="set" data="outbound_redirect_fatal=true"/>
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
Sofia | sofia.c |
park_after_bridge
If set to true, it will park the call after mod_dptools: bridge returns. This is checked before transfer_after_bridge and hangup_after_bridge.
Default: false
Usage:
<action application="set" data="park_after_bridge=true"/> <action application="bridge" data="sofia/gateway/myprovider/5551231234"/>
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
core | switch_ivr_bridge.c | 12691 |
park_timeout
When set, a parked call will disconnect after the timeout has occurred. Timeout is specified in seconds. If no park_timeout value is set then the parked call will be held indefinitely or until it is removed with a uuid_transfer.
Usage:
<action application="set" data="park_timeout=30"/> <action application="park"/>
You can also specify which hangup_cause you need when the channel is disconnected by park_timeout.
Usage: <action application="set" data="park_timeout=30:MEDIA_TIMEOUT"/>
See also:
- park
- uuid_transfer
- Hangup_causes
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_sofia | sofia.c | 10391 |
core | switch_ivr.c | 10391 |
pass_rfc2833
If set, it passes RFC 2833 DTMF's from one side of a bridge to the other, untouched. If unset, it decodes and re-encodes them before passing them on.
Note: this has no effect when bypass_media or proxy_media is set.
Usage:
<action application="set" data="pass_rfc2833=true"/>
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_sofia | sofia_glue.c | 5114 |
passthru_ptime_mismatch
If ptime from leg A and leg B don't match and if mod_com_g729 is used, the call would normally use the codec to re-packetize the RTP stream.
With this parameter, mod_com_g729 will re-packetize without decoding/encoding, as mod_g729 would do.
Usage:
This has to be set in {} before bridging. That will probably not work if set using export before bridging.
<action application="bridge" data="{passthru_ptime_mismatch=true}sofia/gateway/trunk/$1"/>
Note: It may also be set globally in vars.xml.
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
core | switch_channel.c | e7cafae29d0720f06280255a55cc94548e7e12a4 |
core | switch_core_io.c | e7cafae29d0720f06280255a55cc94548e7e12a4 |
process_cdr
Indicates how to process CDR records.
Can be undefined or set to "false", "true", "a_only", "b_only"
- false - indicates to not process the record.
- true - or undefined indicates the default behavior which is to process all CDR records.
- a_only - indicates to only process CDR records on the inbound leg of a call.
- b_only - indicates to only process CDR records on the outbound leg of a call.
This variable is unconditionally exported
Usage:
<action application="set" data="process_cdr=a_only"/>
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
core | switch_core_session.c | 7214 |
core | switch_core_state_machine.c | 7212 |
profile_created_time
Contains the start time (in microseconds) of when the call profile was created.
Usage:
In the event that a call is transferred, this is the effective 'created_time' for that transfer.
For example, if you did the following call flow:
2000 > 2001 (created_time=1357138714 / profile_created_time=1357138714) transfer to 2002 (created_time=1357138714 / profile_created_time=1357138752) transfer to 2003 (created_time=1357138714 / profile_created_time=1357138766) transfer to 2004 (created_time=1357138714 / profile_created_time=1357138784)
You would still use 'progress_time' to retrieve the progress start time, no matter if it's a transfer leg or not.
See also:
- created_time
- progress_time
Implemented By:
Logic taken from here:
src/switch_channel.c:switch_channel_set_caller_profile() caller_profile->times->profile_created = switch_micro_time_now();
progress_time
TODO
Usage:
TODO
See also:
TODO
Implemented By:
TODO
proto_specific_hangup_cause
This variable will cause FreeSWITCH to force the SIP response code to a specific setting when hanging up a call. The example below is one where all possible extensions have been tested and failed and you want FreeSWITCH to generate and respond with a specific code. (This is not a passthrough example).
By the way, you'll be unable to rewrite the hangup cause for a bridge that gets a 180 or 183 packet from the gateway before getting a 4xx, 5xx or 6xx packet (because those bridges don't 'fail'). This happens with SIP providers that give a 183 Session Progress before a 404 Not Found if the PSTN number dialled doesn't exist.
Usage:
<extension name="nothing_left" continue="true"> <condition break="always"> <action application="set" data="proto_specific_hangup_cause=sip:503"/> <action application="hangup"/> </condition> </extension>
Example:
SIP Response Map
<extension name="from_gw_to_internal"> <condition field="destination_number" expression="^(.*)$"> <action application="set" data="hangup_after_bridge=true"/> <action application="set" data="continue_on_fail=19"/> <action application="bridge" data="{sip_cid_type=none}sofia/gateway/gw/$1"/> <action application="transfer" data="480to503"/> </condition> </extension>
<extension name="480to503"> <condition field="${proto_specific_hangup_cause}" expression="<a href="sip:480">sip:480"> <action application="set" data="sip_ignore_remote_cause=true"/> <action application="respond" data="503"/> <action application="hangup" data="NORMAL_CIRCUIT_CONGESTION"/> </condition> </extension>
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_sofia | mod_sofia.c | 10090 |
mod_sofia | sofia.c | 12676 |
read_codec
Read only. The negotiated codec of the inbound call leg.
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
core | switch_core_codec.c | 10045 |
RECORD_APPEND
Recording is appended to file. set RECORD_APPEND=true on the channel and all recordings will behave this way to formats which support it (curently mod_sndfile for WAV, etc.)
Usage:
<action application="set" data="RECORD_APPEND=true"/>
RECORD_ARTIST
Set prior to performing a record to store in the file header meta data (provided the file format supports meta headers).
Usage:
<action application="set" data="RECORD_ARTIST=Unknown"/>
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_voicemail | mod_voicemail.c | 6984 |
core | switch_ivr_async.c | 4796 |
core | switch_ivr_play_say.c | 4796 |
RECORD_BRIDGE_REQ
Record session only when the channel is bridged.
Usage:
<action application="set" data="RECORD_BRIDGE_REQ=true"/>
See also:
http://jira.freeswitch.org/browse/FS-5127
RECORD_COMMENT
Set prior to performing a record to store in the file header meta data (provided the file format supports meta headers).
Usage:
<action application="set" data="RECORD_COMMENT=This is a blog spot"/>
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_voicemail | mod_voicemail.c | 6984 |
core | switch_ivr_async.c | 4796 |
core | switch_ivr_play_say.c | 4796 |
RECORD_COPYRIGHT
Set prior to performing a record to store in the file header meta data (provided the file format supports meta headers).
Usage:
<action application="set" data="RECORD_COPYRIGHT=(c)2007-me"/>
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_voicemail | mod_voicemail.c | 6984 |
core | switch_ivr_async.c | 4796 |
core | switch_ivr_play_say.c | 4796 |
RECORD_DATE
Set prior to performing a record to store in the file header meta data (provided the file format supports meta headers).
Usage:
<action application="set" data="RECORD_DATE=2007-01-16"/>
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_voicemail | mod_voicemail.c | 6984 |
core | switch_ivr_async.c | 4796 |
core | switch_ivr_play_say.c | 4796 |
RECORD_DISCARDED
If a recording gets dropped or discarded then this channel variable is set to true. Useful for diagnostics.
Usage:
N/A
See also:
Implemented By:
Module Name | Source File | Last Updated |
---|---|---|
core | switch_ivr_async.c | c465c435 |
record_fill_cng
Description need please.
Usage:
<action application="set" data="record_fill_cng=1200"/>
See also:
- Variable record_waste_resources
Implemented By:
Module Name | Source File | Last Updated |
---|---|---|
Core | switch_ivr_play_say.c |
RECORD_HANGUP_ON_ERROR
When set to true this channel variable will cause the call to hangup if there is an error when trying to record the call. This is not a common feature, however in cases where a call MUST be recorded it makes it impossible to have calls that are not recorded. (Useful in some business scenarios.)
Usage:
<action application="set" data="RECORD_HANGUP_ON_ERROR=true"/>
Implemented By:
Module Name | Source File | Last Updated |
---|---|---|
Core | switch_ivr_async.c | a3e6bead |
RECORD_MIN_SEC
Sets the minimum recording length. Normally a recording must be at least 3 seconds long. If a recording does not meet the minimum length, it is deleted after being recorded.
Usage:
<action application="set" data="RECORD_MIN_SEC=0"/>
See also:
- record_session
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
core | switch_ivr_async.c | 15272 |
record_ms
Read Only. Contains the length in milliseconds of the most recent recording.
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
core | switch_ivr_play_say.c | 15155 |
record_sample_rate
Set the sample rate of the recording.
Usage:
<action application="set" data="record_sample_rate=8000"/>
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_dptools | mod_dptools.c | 6706 |
RECORD_READ_ONLY
Record read stream only.
Usage:
<action application="set" data="RECORD_READ_ONLY=true"/>
See also:
- RECORD_STEREO
- RECORD_WRITE_ONLY
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
core | switch_ivr_async.c | 15257 |
record_restart_limit_on_dtmf
When set to true it allows the person recording to press a DTMF key and extend the amount of time they have before the recording times out.
Usage:
<action application="set" data="record_restart_limit_on_dtmf=true"/>
See also:
- Misc._Dialplan_Tools_record
Implemented By:
Module Name | Source File | Last Updated |
---|---|---|
Core | switch_ivr_play_say.c |
record_sample_rate
Specify the sampling rate of the recorded file.
Usage:
<action application="set" data="record_sample_rate=8000"/>
See also:
- Misc._Dialplan_Tools_record
Implemented By:
Module Name | Source File | Last Updated |
---|---|---|
Core | switch_ivr_play_say.c |
record_samples
Read Only. Contains the number of audio samples in the most recent recording.
RECORD_SOFTWARE
Set prior to performing a record to store in the file header meta data (provided the file format supports meta headers).
Usage:
<action application="set" data="RECORD_SOFTWARE=FreeSWITCH"/>
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_voicemail | mod_voicemail.c | 6984 |
core | switch_ivr_async.c | 4796 |
core | switch_ivr_play_say.c | 4796 |
RECORD_STEREO
Record leg A and leg B streams (i.e. the caller is recorded to the left channel and the reciever is recorded on right channel) into different channel in a stereo file.
Usage:
<action application="set" data="RECORD_STEREO=true"/>
See also:
- RECORD_WRITE_ONLY
- RECORD_READ_ONLY
- RECORD_STEREO_SWAP
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
core | switch_ivr_async.c | 5772 |
RECORD_STEREO_SWAP
Allow to invert the recording channel when RECORD_STEREO variable is set to true. So the caller is recorded to the right channel and the receiver is recorded on left channel.
Usage:
<action application="set" data="RECORD_STEREO_SWAP=true"/>
See also:
- RECORD_STEREO
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
core | src/switch_core_media_bug.c | 2011-12-14 |
RECORD_TITLE
Set prior to performing a record to store in the file header meta data (provided the file format supports meta headers).
Usage:
<action application="set" data="RECORD_TITLE=MegaMusic"/>
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_voicemail | mod_voicemail.c | 6984 |
core | switch_ivr_async.c | 4796 |
core | switch_ivr_play_say.c | 4796 |
record_waste_resources
By default record doesn't send RTP packets. This is generally acceptable, but for longer recordings or depending on the RTP timer of your gateway, your channel may hangup with cause MEDIA_TIMEOUT. Setting this variable will 'waste' bandwidth by sending RTP even during recording. The value can be true/false/<desired silence factor>. By default the silence factor is 1400 if you set record_waste_resources=true.
Usage:
<action application="set" data="record_waste_resources=true"/>
See also:
- RECORD_WRITE_ONLY
- RECORD_READ_ONLY
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
core | switch_ivr_play_say.c | 13152 |
RECORD_WRITE_ONLY
Record write stream only.
Usage:
<action application="set" data="RECORD_WRITE_ONLY=true"/>
See also:
- RECORD_STEREO
- RECORD_READ_ONLY
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
core | switch_ivr_async.c | 15257 |
recording_follow_transfer
Set to true if you want recording to continue after a transfer.
Example
<action application="set" data="recording_follow_transfer=true"/>
rtcp_octet_count
Contains number of RTCP octets collected during the call.
Usage:
N/A
See also:
- Variable_rtcp_packet_count
- CDR
Implemented By:
Module Name | Source File | Last Updated |
---|---|---|
Mod_sofia | sofia_glue.c | d5ff3e04 |
rtcp_packet_count
Contains number of RTCP packets collected during the call.
Usage:
N/A
See also:
- Variable_rtcp_octet_count
- CDR
Implemented By:
Module Name | Source File | Last Updated |
---|---|---|
Mod_sofia | sofia_glue.c | d5ff3e04 |
rtp_disable_vad_in
Description needed! Please contribute one.
Usage:
Example needed! Please contribute one.
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_sofia | sofia_glue.c | 5364 |
rtp_disable_vad_out
Description needed! Please contribute one.
Usage:
Example needed! Please contribute one.
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_sofia | sofia_glue.c | 5364 |
rtp_enable_vad_in
Description needed! Please contribute one.
Usage:
Example needed! Please contribute one.
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_sofia | sofia_glue.c | 5364 |
rtp_enable_vad_out
Description needed! Please contribute one.
Usage:
Example needed! Please contribute one.
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_sofia | sofia_glue.c | 5364 |
sdp_m_per_ptime
Adds a new m= line for each distinct ptime in codec list.
When this variable is not set:
- When mixing codecs with various ptime in a codec list, they will now be allowed to co-exist in the sdp but it will send no ptime attr. This means the ptime preferences on the offer will be ignored when mixing codecs with various ptimes. When receiving a codec list with no ptime attr, the ptime will be chosen from local preference instead of assuming 20ms. This means if offer contains PCMU with no ptime and FS has PCMU@40i
- Dynamic payloads will now start at 98 and increment per additional dynamic codec per call. So now you can add CELT@32000h,CELT@48000h and each one will be auto-assigned a dynamic payload type.
Is now implied to be true, if you don't like this set it to false but its going to be undefined behaviour. This basically means if you call in with ptime 30 then you have a bunch of ptime 20 codecs in your outbound list that there will be one m= line with 30 and the original inbound codec and more m= lines for each discinct ptime in your list. This is, of course, will depend on disable_trancoding or absolute_codec_string as well
Usage:
<action application="set" data="sdp_m_per_ptime=true"/>
See also:
Implemented By:
Module Name | Source File | Last Updated |
---|---|---|
mod_sofia | sofia_glue.c | dfa78985 |
sdp_secure_savp_only
When sip_secure_media=true FreeSWITCH will offer both AVP and SAVP in the SDP. Setting sdp_secure_savp_only=true (in addition to sip_secure_media=true) will cause FreeSWITCH to offer only SAVP in the SDP.
Usage:
<action application="export" data="sdp_secure_savp_only=true"/>
See also:
Implemented By:
Module Name | Source File | Last Updated |
---|---|---|
mod_sofia | sofia_glue.c | 097d9a1 |
send_silence_when_idle
When greater than 0, this variable tells FreeSWITCH to transmit comfort noise when idle. The value of this variable is set to the divisor of the silence generating function. 400 or 1400 are common values set, but you may experiment with other choices to pick one that sounds best.
When true, FreeSWITCH will pick a default comfort noise value.
When -1, FreeSWITCH will transmit silence without comfort noise.
Usage:
<action application="set" data="send_silence_when_idle=400"/>
See also:
- bridge_generate_comfort_noise
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
core | switch_ivr.c | 11274 |
core | switch_ivr_originate.c | 10664 |
session_in_hangup_hook
Allows channel variables to be accessible in the api_hangup_hook that gets executed for the channel.
Usage:
<action application="set" data="session_in_hangup_hook=true"/>
See Lua#Special_Case:_env_object for an example of how to use this.
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
core | switch_core_state_machine.c | 10468 |
signal_bond
UUID of the channel this channel is bridged/bonded to. Not present on a one-legged call.
Usage:
Example needed! Please contribute one.
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_dptools | mod_dptools.c | 7885 |
mod_loopback | mod_loopback.c | 9646 |
mod_sofia | mod_sofia.c | 7103 |
mod_sofia | sofia.c | 11562 |
mod_sofia | sofia_glue.c | 8232 |
core | switch_channel.c | 7546 |
core | switch_core_session.c | 4944 |
core | switch_ivr.c | 7870 |
core | switch_ivr_async.c | 7075 |
core | switch_ivr_bridge.c | 9603 |
core | switch_ivr_play_say.c | 8065 |
sip_acl_authed_by
Contains the name of the ACL node that authorized this call.
Usage:
None.
See also:
Implemented By:
sip_acl_token
Contains the ACL auth token for the current call.
Usage:
N/A
See also:
- [Mod_sofia]
Implemented By:
- [Mod_sofia]
sip_auth_password
For mod_sofia use with sip_auth_username to answer auth challenges without defining a full gateway.
Usage:
originate {sip_auth_username=<your_user_name>,sip_auth_password=<your_password>}sofia/external/1xxxxxxx@12.34.56.78 &echo
See also:
- sip_auth_username
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_sofia | sofia_reg.c | 12819 |
sip_auth_username
For mod_sofia answer auth challenges without defining a full gateway. Used in tandem with sip_auth_password. Also indicates the SIP username a device successfully registered to FreeSWITCH with.
Usage:
originate {sip_auth_username=<your_user_name>,sip_auth_password=<your_password>}sofia/external/1xxxxxxx@12.34.56.78 &echo
Notes:
This should contain the username of the authenticated user that has triggered this request, if applicable.
Depending on how your FreeSWITCH instance is configured, you may experience problems with this variable being incorrect or blank after calling the 'bridge' application. If this happens, you may want to force the channel variable, for example;
<action application="bridge" data="{sip_auth_username=${sip_auth_username}}sofia/gateway/external/2000"/>
See also:
- sip_auth_password
- sip_authorized
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_sofia | sofia_reg.c | 12819 |
sip_authorized
sip_authorized indicates whether the SIP device accessing the dialplan is authorized to FreeSWITCH or not. The SIP device can be authorized either via an ACL or via digest authentication.
Usage:
Example needed! Please contribute one.
See also:
- sip_auth_username
- sip_acl_authed_by
- sip_acl_token_vars
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_sofia | sofia.c | 5299 |
sip_auto_simplify
When set, this directs FreeSWITCH to remove itself from the SIP signaling path if it can safely do so.
Usage:
<action application="set" data="sip_auto_simplify=true"/>
See also:
- uuid_simplify
sip_callee_id_name
DEPRECATED. Use variable origination_callee_id_name instead
sip_callee_id_number
DEPRECATED. Use variable effective_callee_id_number instead
<anthm>: sip_callee_id_X was deprecated in favor of effective_callee_id_X
<anthm>: effective and sip are the same it was just a mistake to call it sip in case we use it for other protocols so both work for now
sip_callee_id_number
Set on the inbound leg to control what caller ID number appears in the caller phone's display.
Usage:
<action application="set" data="sip_callee_id_name=Reginald" /> <action application="set" data="sip_callee_id_number=2332" /> <action application="bridge" data="sofia/gateway/provider/<Reginald's cellphone number>" />
If you find using set doesn't work, try using export instead.
See also:
- sip_callee_id_name
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_sofia | sofia.c | 15110 |
mod_sofia | mod_sofia.c | 15211 |
sip_cid_in_1xx
Prevents FreeSWITCH when it receives 183 from leg-B to automatically insert RPID before sending 183 to leg-A.
Usage:
<action application="set" data="sip_cid_in_1xx=false"/>
See also:
This can be defined profile-wide with: pass-callee-id=false
sip_cid_type
Modify caller ID in SDP header. Can be set to none, default, pid, or rpid (default).
rpid=Remote-Party-ID header pid=P-Asserted-Identity header none=Caller ID will be in the SIP "From" Note: for gateways, this will not work. You will require caller-id-in-from=true in the gateway settings
Usage:
{sip_cid_type=none}sofia/default/user@example.com
Or,
{sip_cid_type=rpid}sofia/default/user@example.com i.e. Remote-Party-ID: "9094445555" <sip:9094445555@1.2.3.4>;party=calling;screen=yes;privacy=off
Or,
{sip_cid_type=pid}sofia/default/user@example.com i.e. P-Asserted-Identity: "9094445555" <sip:9094445555@1.2.3.4>
Or,
{sip_cid_type=rpid,origination_caller_id_name=test,origination_caller_id_number=1234}sofia/default/user@example.com
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_sofia | sofia.c | 12427 |
mod_sofia | sofia_glue.c | 12427 |
sip_copy_multipart
FreeSWITCH supports only INVITEs with multipart bodies. FreeSWITCH does not support response messages such as 183 Session Progress or 200 Ok with multipart bodies. Typically SIP bodies only have one MIME part with an SDP using MIME type application/sdp. The SIP spec allows for multiple bodies defined with MIME type multipart/mixed. In this case FreeSWITCH will do it's best to find the MIME part with the SDP and parse that as it normally does. However, you can change FreeSWITCH behavior with multipart bodies and bridge using this variable.
Usage:
To have FreeSWITCH keep the multiple MIME parts intact when using bridge (default):
<action application="set" data="sip_copy_multipart=true"/>
NOTE: FreeSWITCH will "do the right thing" and attach an application/sdp type generated by FreeSWITCH (per your settings) for the B leg as it normally would. The other non-SDP MIME parts just pass through.
To have FreeSWITCH strip the multiple MIME parts when using bridge:
<action application="set" data="sip_copy_multipart=false"/>
sip_enable_soa
For per call basis which can be set to false to disable SIP SOA from sofia and most likely result in untouched exchange of SDP.
Usage:
<action application="set" data="bypass_media=true"/> <action application="export" data="sip_enable_soa=false"/>
sip_execute_on_image
Execute an application as soon as you get a T.38 invite. This variable is similar to execute_on_answer.
You can run t38_gateway or rxfax etc when you get a T.38 re-invite but no CNG tone, or you want to ignore the tone and only react when getting a T.38 re-invite.
Usage:
Need an example.
See also:
- execute_on_answer
- mod_dptools
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_sofia | sofia_glue.c |
sip_force_audio_fmtp
Set the audio fmtp.
Usage:
Please add example if you have one.
See also:
Implemented By:
Module Name | Source File | Last Updated |
---|---|---|
mod_sofia | mod_sofia.c | 6360264f |
sip_from_display
Description needed! Please contribute one.
Usage:
Example needed! Please contribute one.
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_sofia | sofia_glue.c | 12566 |
sip_hangup_disposition
This variable contains the value of who sent the SIP BYE message. Some examples from XML CDRs:
<sip_hangup_disposition>send_bye</sip_hangup_disposition> <sip_hangup_disposition>recv_bye</sip_hangup_disposition> <sip_hangup_disposition>send_refuse</sip_hangup_disposition> <sip_hangup_disposition>send_cancel</sip_hangup_disposition>
Interpretation of these values differs on incoming and outgoing calls since FreeSWITCH is at different ends of the session.
Value | Incoming | Outgoing |
---|---|---|
send_bye | FS sent BYE to the caller (we hung up) | FS sent BYE to the endpoint (we hung up) |
recv_bye | FS received BYE from the caller (they hung up) | FS received BYE from the endpoint (they hung up) |
send_refuse | FS rejected the call (e.g. 4xx or 5xx) | Endpoint rejected the call (e.g. 4xx or 5xx) |
send_cancel | n/a | FS aborted the call (we sent CANCEL) |
Usage:
Look in CDR for the value; only valid after call has ended.
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_sofia | mod_sofia.c | |
mod_sofia | sofia.c |
sip_ignore_183nosdp
Ignoring 183 w/o SDP. This option is not for normal basic call flow.
Usage:
<action application="set" data="sip_ignore_183nosdp=true"/>
See also:
- Channel_Variables#sip_ignore_183nosdp
sip_ignore_reinvites
Tells FreeSWITCH to accept/ignore re-INVITEs from remote end.
Usage:
Don't allow any re-INVITEs once bridged.
<action application="set" data="sip_ignore_reinvites=true"/>
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_sofia | sofia.c | 9634 |
sip_invite_contact_params
Description needed! Please contribute one.
Usage:
Example needed! Please contribute one.
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_sofia | sofia_glue.c | 12290 |
sip_invite_domain
Set the from domain in leg (B).
Usage:
<action application="bridge" data="{sip_invite_domain=${sip_from_host}}sofia/gateway/gw1/$1@domain.org"/>
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_sofia | sofia_glue.c | 9127 |
sip_invite_from_params
Sets the from parameters on the B-leg of the call. The from parameters come after user@host:port and before '>'. The initial semi-colon is added after the port.
Usage:
{sip_invite_from_params=otg=mytrunk}sofia/gateway/sonus/$1
Result:
From: <sip:5552345678@sonus:5060;otg=mytrunk>;tag=blah
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_sofia | sofia_glue.c | 12287 |
ip_invite_params
Description needed! Please contribute one.
Usage:
Example needed! Please contribute one.
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_sofia | mod_sofia.c | 11069 |
mod_sofia | sofia_glue.c | 7861 |
sip_invite_req_uri
Sets the uri in the header Request-Line INVITE when calling bridge or originate.
NOTE: RFC 3261 specifies that compliant endpoints SHOULD route based on the Request URI, not the URI in To:
Usage:
<action application="bridge" data="{sip_invite_req_uri=sip:11112222@test1.com}sofia/external/33334444%192.168.4.6"/>
Result:
INVITE sip:11112222@test1.com SIP/2.0 From: "" <sip:0000000000@192.168.2.7>;tag=N6K579y4g6j0D To: <sip:33334444@192.168.4.6>
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_sofia |
sip_invite_route_uri
Sets the uri in the header Route when calling bridge or originate.
Usage:
originate {sip_invite_route_uri=<sip:+48399999999@10.0.0.51:5080;lr;orig>,origination_caller_id_number=399999000}sofia/internal/+48399999999@domain.com &echo
Result:
INVITE sip:+48399999999@domain.com SIP/2.0 Via: SIP/2.0/UDP 10.0.0.51;rport;branch=z9hG4bKpmFv4aXv4tKcK Route: <sip:+48399999999@10.0.0.51:5080;lr;orig>
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_sofia | sofia_glue.c |
sip_invite_tel_params
Tel URI parameters (npdi, rn) appearing in SIP URI on outbound calls.
Usage:
<action application="bridge" data="{sip_invite_tel_params=npdi=yes;rn=555000001,sip_invite_params=user=phone}sofia/gateway/test_gw/555000002"/>
Result:
INVITE sip:555000002;npdi=yes;rn=5555550001@1.2.3.4;user=phone SIP/2.0
See also:
http://jira.freeswitch.org/browse/FS-5118
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_sofia | sofia_glue.c | a3786d5 |
sip_invite_to_params
Description needed! Please contribute one.
Usage:
Example needed! Please contribute one.
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_sofia | sofia_glue.c | 12287 |
sip_invite_to_uri
Sets the uri in the header To when calling bridge or originate.
Usage:
originate {sip_invite_to_uri=<sip:11112222@test1.com>}sofia/internal/33334444@192.168.4.6 &park
Result:
INVITE sip:33334444@192.168.4.6 SIP/2.0 From: "" <sip:0000000000@192.168.2.7>;tag=N6K579y4g6j0D To: <sip:11112222@test1.com>
Alternatively, if you need to set just the username in the header To, you can pass it at the end of the dial string:
originate sofia/internal/33334444@192.168.4.6^11112222 &park
Result:
INVITE sip:33334444@192.168.4.6 SIP/2.0 From: "" <sip:0000000000@192.168.2.7>;tag=Qr6pB00BBrZ5m To: <sip:11112222@@192.168.4.6>
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_sofia |
rtp_jitter_buffer_during_bridge
Enables/disables the jitter buffer during bridge.
Usage:
<action application="set" data="rtp_jitter_buffer_during_bridge=true"/>
or,
<action application="set" data="rtp_jitter_buffer_during_bridge=false"/>
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_sofia | mod_sofia.c |
sip_jitter_buffer_plc
Enables/disables packet loss concealment (PLC) when using the jitter buffer. PLC is enabled by default when the jitter buffer is enabled. Set this variable before enabling the jitter buffer for it to have an effect.
Usage:
<action application="set" data="sip_jitter_buffer_plc=true"/>
or,
<action application="set" data="sip_jitter_buffer_plc=false"/>
See also:
sip_local_sdp_str
Description goes here
Usage:
Example needed.
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_sofia | sofia_glue.c | ? |
mod_sofia | sofia_glue.c | ? |
sip_mirror_remote_audio_codec_payload
To tell sip to break the rfc and expect the codec payload the other side replies with rather than the one it offered which is the correct behavior.
Usage:
Examples needed!!!
Note: This variable can be set globally or per channel.
See also:
sip_network_destination
It is intended for use with devices registering behind a NAT where the Request-URI should contain the contact that was bound to the AOR during the registration request while the request itself should be sent to the public IP and port number of the NAT "pinhole". It does not add a Route header field to the request like the ;fs_path= or the sip_route_uri options do.
Usage:
<action application="bridge" data="{sip_network_destination=sip:5551234567@66.243.109.243:10005}sofia/external/5551234567@172.16.110.45:5060"/>
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_sofia | sofia_glue.c | 12409 |
sip_recovery_break_rfc
To NOT reverse the from and to on UAS Re-INVITEs. This breaks RFC.
Usage:
Examples needed!!!
Note: This variable can be set globally or per channel.
sip_renegotiate_codec_on_reinvite
Allow SDP codec change with re-INVITE
Usage:
<action application="bridge" data="{sip_renegotiate_codec_on_reinvite=true}sofia/gateway/trunk/$1"/>
Note: It may also be set globally in vars.xml or set it in sip profiles (<param name="renegotiate-codec-on-reinvite" value="true"/>)
sip_wait_for_aleg_ack
When you set the variable sip_wait_for_aleg_ack on the b leg in the {} for the outbound call, this should make the B leg delay sending the ACK until it sees that the A leg has recv'd an ack.
Usage:
<action application="bridge" data="{sip_wait_for_aleg_ack=true}sofia/internal/foo@bar.com"/>
skeleton
This is an example of how to create a channel variable page. This section is the description of the variable. Put the description information here and then the usage example below.
Usage:
<action application="set" data="skeleton=foo"/>
See also:
- __Channel Variables
- Add_Your_See_Other_Items_Here
Implemented By:
Module Name | Source File | Last Updated |
---|---|---|
Leave blank if you don't know itLeave blank if you don't know it | Leave blank if you don't know it |
skip_cdr_causes
This is a list of call hangup causes that should not trigger cdr processing.
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_sofia | sofia.c | 2012-09-19, git commit 3cf238fc9a1370a45d362c2193b8d3634ccd1d11 |
spandsp_dtmf_rx_filter_dialtone
Sets the filter dialtone parameter in the spandsp DTMF detector. Dialtone filtering is disabled by default. Set this variable prior to executing spandsp_start_dtmf.
Usage:
<action application="set" data="spandsp_dtmf_rx_filter_dialtone=true"/>
<action application="spandsp_start_dtmf" />
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_spandsp | mod_spandsp_dsp.c |
spandsp_dtmf_rx_reverse_twist
Sets the reverse twist setting in the spandsp DTMF detector. Reverse twist is set to 4 dB by default. This value can be safely increased up to 6 or 7 without a significant increase in talk-off to allow DTMFs that exceed this threshold to be detected. Set this variable prior to executing spandsp_start_dtmf.
Usage:
<action application="set" data="spandsp_dtmf_rx_reverse_twist=6"/>
<action application="spandsp_start_dtmf" />
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_spandsp | mod_spandsp_dsp.c |
spandsp_dtmf_rx_threshold
Sets the threshold parameter in the spandsp DTMF detector. Threshold is set to -42 dBm0 by default. Set this variable prior to executing spandsp_start_dtmf.
Usage:
<action application="set" data="spandsp_dtmf_rx_threshold=-42"/>
<action application="spandsp_start_dtmf" />
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_spandsp | mod_spandsp_dsp.c |
spandsp_dtmf_rx_twist
Sets the twist parameter in the spandsp DTMF detector. Twist is set to 8 dB by default. Set this variable prior to executing spandsp_start_dtmf.
Usage:
<action application="set" data="spandsp_dtmf_rx_twist=8"/>
<action application="spandsp_start_dtmf" />
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_spandsp | mod_spandsp_dsp.c |
suppress_cng
Enable/Disable comfort noise. (RTP/AVP 13)
You may wish to suppress comfort noise (which actually supresses VAD) if the call audio is being compromised by aggressive or poorly implemented VAD+silence suppression (eg. short words are missed in conversation)
Usage:
<action application="export" data="suppress_cng=true"/>
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_spandsp | mod_spandsp_dsp.c |
switch_m_sdp
The B-leg remote SDP.
Usage:
- Used to store the remote SDP used by the other leg/channel of a call. (In the A-leg that will be the remote SDP of the B-leg.)
- This variable is set, but never used, by FreeSWITCH. ("read-only")
- Known as SWITCH_B_SDP_VARIABLE in mod_sofia and the core switch code
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_sofia | mod_sofia.c | 8372 |
mod_sofia | sofia.c | 10463 |
mod_sofia | sofia_glue.c | 5114 |
core | switch_core_session.c | 4796 |
switch_r_sdp
Read only. This variable holds the remote SDP for the current leg/channel.
Usage:
Rewriting SDP:
<action application="set"><![CDATA[switch_r_sdp=(sdp here) ]]>
</action>
note: Don't add a carriage return after "set"> or you'll end up writing a variable with a different name, leaving switch_r_sdp with the same value.
Known as SWITCH_R_SDP_VARIABLE in source code
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_sofia | mod_sofia.c | 8542 |
mod_sofia | sofia.c | 9591 |
core | switch_core_session.c | 4796 |
temp_hold_music
This variable specifies a hold music value that gets played to a caller only until they get transferred. After the transfer, the hold_music variable will apply.
Usage:
<action application="set" data="temp_hold_music=local_stream://alternate_moh"/>
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
Core | switch_channel.c |
timer_name
If set will make playback and speak use a timer to clock the audio instead of the read.
Usage:
<action application="set" data="timer_name=soft"/>
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
Core | switch_ivr_play_say.c | 4796 |
timezone
Sets the timezone for this particular call. Can be used, e.g., to set the timezone for a caller who is checking his/her voicemail. The value is expressed in Linux timezone format (ex. America/New_York -- see /usr/share/zoneinfo/zone.tab for the standard list of Linux timezones).
Note that this channel variable is only respected by the phrase layer -- ie, if you're using the 'say' application to announce times, this will work fine.
You can set the time zone Globally for Freeswitch in /conf/vars.xml by adding this line: <X-PRE-PROCESS cmd="set" data="timezone=America/Toronto"/> (of course replace the 'America/Toronto' with your own time zone.
If you would like to specify the time zone in the dialplan add <action application="set" data="timezone=America/Toronto"/> to your dialplan. (again replace with the proper timezone)
Finally you can specify a time zone for a particular extension in /conf/directory/default/'extension'.xml
Usage:
<action application="set" data="timezone=GMT0"/>
or,
<action application="set" data="timezone=America/New_York"/>
Directory Usage:
<param name="timezone" value="America/New_York"/>
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_say_en | mod_say_en.c | 10214 |
tod_tz_offset
Sets the GMT offset to be used on this call for tod based conditions.
NOTE: the variable must actually be set before the comparison, so either set it inline, transfer, or set it in the user directory.
You can set the offset Globally for Freeswitch in /conf/vars.xml by adding this line: <X-PRE-PROCESS cmd="set" data="tod_tz_offset=5"/> (of course replace the '5' with your GMT offset.
If you would like to specify the time offset in the dialplan add <action application="set" data="tod_tz_offset=5"/> to your dialplan. (again replace with the proper offset)
Finally you can set the variable for a particular extension in /conf/directory/default/'extension'.xml
Usage:
<action application="set" data="tod_tz_offset=5"/>
Directory Usage:
<variable name="tod_tz_offset" value="+2"/>
See also:
transfer_after_bridge
This variable can control what happens when a call is hang up. This can be used in conjunction with mod_fifo to control the "agent", possibly sending them back to an agent queue. This is checked after park_after_bridge and before hangup_after_bridge.
Usage:
<action application="set" data="transfer_after_bridge=1000"/>
or,
<action application="set" data="transfer_after_bridge=1000:XML:default"/>
(Note the : separator)
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
core | switch_ivr_bridge.c | 12691 |
transfer_on_fail
Allows you to tranfer call flow when a called party can not be reached for specific reasons ( unallocated_number, etc )
Usage:
<action application="set" data="transfer_on_fail= UNALLOCATED_NUMBER"/>
In this example, if you were to attempt a bridge that resulted in "UNALLOCATED_NUMBER" , the call flow would be "transferred" to the "UNALLOCATED_NUMBER" destination in your current dialplan ( probably default xml dialplan )
or,
<action application="set" data="transfer_on_fail=<hangupcauses> <destination> <dialplan> <context>"/>
or,
<action application="set" data="transfer_on_fail=1"/>
<hangupcause> can be the string "auto_cause" which will be replaced by the hangup cause returned by the bridge. this is the default action if no destination, dialplan or context are specified.
<action application="set" data="hangup_on_bridge=true"/>
<action application="set" data="continue_on_fail=true"/>
<action application="set" data="transfer_on_fail=UNALLOCATED_NUMBER auto_cause xml error"/>
<action application="bridge" data="A..."/>
<action application="bridge" data="B..."/>
then have the an "Error" context in your dialplan.
<context name="error">
<extension name="UNALLOCATED_NUMBER" continue="true">
<condition field="${originate_disposition}" expression="UNALLOCATED_NUMBER" continue="false" break="on-true">
<action application="playback" data="/usr/local/freeswitch/sounds/NotInService.wav"/>
<action application="hangup" data="NORMAL_CLEARING"/>
</condition>
</extension>
</context>
See also:
transfer_to
Description needed
Example: Example needed
See also:
uuid_bridge_continue_on_cancel
When set to true causes the system to move on in the dialplan if it hits a bad b-leg. Default is false because this behavior is probably not recommended.
You may find this variable useful when implementing Dialplan_FollowMe
Usage:
<action application="set" data="uuid_bridge_continue_on_cancel=true"/>
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
core | switch_ivr_bridge.c | 17044 |
verbose_sdp
Enable this variable to fix a bug in certain VoIP phones.
The a=rtpmap lines are optional (RFC 3264) for static payload types according to the SDP standard, but not all phones implement this correctly and fail if these lines are missing.
By default FreeSWITCH will omit these lines so that the SDP is smaller (which lowers bandwidth use and avoids packet fragmentation). If this variable is set to true, FS will instead send these a=rtpmap for all codecs.
For example, Polycom phone requires you to list all codecs in the RTP map even though the SDP is valid, but they seem not to look at any codecs that are not in the map.
Usage: in vars.xml
verbose_sdp=true
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_sofia | sofia_glue.c | n/a |
write_codec
Read only. The negotiated codec of the outbound call leg.
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
core | switch_core_codec.c | 10045 |
data
Description needed! Please contribute one.
Usage:
Example needed! Please contribute one.
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_dptools | mod_dptools.c | 7402 |
endpoint_disposition
Description needed! Please contribute one.
Usage:
Example needed! Please contribute one.
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_sofia | mod_sofia.c | 7779 |
mod_sofia | sofia.c | 12652 |
mod_sofia | sofia_glue.c | 5114 |
core | switch_channel.c | 10098 |
fax_document_transferred_pages
Description needed! Please contribute one.
Usage:
Example needed! Please contribute one.
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_fax | mod_fax.c | 10506 |
fax_header
Description needed! Please contribute one.
Usage:
Example needed! Please contribute one.
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_fax | mod_fax.c | 9468 |
fax_start_page
Description needed! Please contribute one.
Usage:
Example needed! Please contribute one.
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_fax | mod_fax.c | 9468 |
fifo_music
Description needed! Please contribute one.
Usage:
Example needed! Please contribute one.
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_fifo | mod_fifo.c | 8065 |
fifo_priority
Description needed! Please contribute one.
Usage:
Example needed! Please contribute one.
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_fifo | mod_fifo.c | 8689 |
fifo_serviced_by
Description needed! Please contribute one.
Usage:
Example needed! Please contribute one.
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_fifo | mod_fifo.c | 8012 |
fifo_serviced_uuid
Description needed! Please contribute one.
Usage:
Example needed! Please contribute one.
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_fifo | mod_fifo.c | 11356 |
fifo_status
The status of the consumer or caller. Usually "WAITING" or "TALKING".
Usage:
None
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_fifo | mod_fifo.c | 9860 |
fifo_strategy
Description needed! Please contribute one.
Usage:
Example needed! Please contribute one.
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_fifo | mod_fifo.c | 8028 |
fifo_target
Description needed! Please contribute one.
Usage:
Example needed! Please contribute one. |
---|
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_fifo | mod_fifo.c | 8689 |
fire_asr_events
If set, fire an event when speech is detected.
Usage:
Example needed! Please contribute one. |
---|
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
core | switch_ivr_async.c | 7402 |
flow_billmsec
Description needed! Please contribute one.
Usage:
Example needed! Please contribute one. |
---|
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
core | switch_channel.c | 6917 |
flow_billsec
Description needed! Please contribute one.
Usage:
Example needed! Please contribute one. |
---|
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
core | switch_channel.c | 6917 |
flow_billusec
Description needed! Please contribute one.
Usage:
Example needed! Please contribute one. |
---|
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
core | switch_channel.c | 6917 |
funny_stun
Description needed! Please contribute one.
Usage:
Example needed! Please contribute one. |
---|
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_sofia | sofia_glue.c | 9088 |
group_context
Description needed! Please contribute one.
Usage:
Example needed! Please contribute one. |
---|
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_dptools | mod_dptools.c | 10917 |
has_t38
Description needed! Please contribute one.
Usage:
Example needed! Please contribute one. |
---|
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_sofia | sofia_glue.c | 11006 |
holding_uuid
Description needed! Please contribute one.
Usage:
Example needed! Please contribute one. |
---|
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_dptools | mod_dptools.c | 7885 |
core | switch_ivr_originate.c | 8475 |
id
Description needed! Please contribute one.
Usage:
Example needed! Please contribute one. |
---|
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_voicemail | mod_voicemail.c | 12887 |
inbound_dialplan
Description needed! Please contribute one.
Usage:
Example needed! Please contribute one. |
---|
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_sofia | sofia.c | 6113 |
instant_ringback
When set, ringback will not wait for indication before sending ringback tone to calling party.
Usage:
<action application="set" data="instant_ringback=true" /> |
---|
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
core | switch_ivr_originate.c | 11740 |
is_outbound
Description needed! Please contribute one.
Usage:
Example needed! Please contribute one. |
---|
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
core | switch_channel.c | 10115 |
language
Description needed! Please contribute one.
Usage:
Example needed! Please contribute one. |
---|
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_dptools | mod_dptools.c | 10516 |
core | switch_ivr_play_say.c | 8054 |
last_app
Description needed! Please contribute one.
Usage:
Example needed! Please contribute one. |
---|
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
core | switch_channel.c | 6917 |
last_arg
Description needed! Please contribute one.
Usage:
Example needed! Please contribute one. |
---|
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
core | switch_channel.c | 6917 |
last_dtmf_duration
Description needed! Please contribute one.
Usage:
Example needed! Please contribute one. |
---|
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
core | switch_core_io.c | 9526 |
last_file_position
Description needed! Please contribute one.
Usage:
Example needed! Please contribute one. |
---|
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_spidermonkey | mod_spidermonkey.c | 5498 |
lcr_auto_route
Description needed! Please contribute one.
Usage:
Example needed! Please contribute one. |
---|
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_lcr | mod_lcr.c | 10510 |
lcr_route_count
Description needed! Please contribute one.
Usage:
Example needed! Please contribute one. |
---|
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_lcr | mod_lcr.c | 10510 |
left_hanging_extension
Description needed! Please contribute one.
Usage:
Example needed! Please contribute one. |
---|
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_sofia | mod_sofia.c | 7425 |
limit_id
Description needed! Please contribute one.
Usage:
Example needed! Please contribute one. |
---|
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_db | mod_db.c |
limit_max
Description needed! Please contribute one.
Usage:
Example needed! Please contribute one. |
---|
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_db | mod_db.c |
limit_rate
Description needed! Please contribute one.
Usage:
Example needed! Please contribute one. |
---|
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_hash | mod_hash.c |
limit_realm
Description needed! Please contribute one.
Usage:
Example needed! Please contribute one. |
---|
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_db | mod_db.c |
limit_usage
Description needed! Please contribute one.
Usage:
Example needed! Please contribute one. |
---|
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_db | mod_db.c | |
mod_hash | mod_hash.c |
local_media_ip
Description needed! Please contribute one.
Usage:
Example needed! Please contribute one. |
---|
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_sofia | sofia_glue.c | 9591 |
local_media_port
Description needed! Please contribute one.
Usage:
Example needed! Please contribute one. |
---|
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_sofia | sofia_glue.c | 9591 |
local_video_ip
Description needed! Please contribute one.
Usage:
Example needed! Please contribute one. |
---|
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_sofia | sofia_glue.c | 5114 |
local_video_port
Description needed! Please contribute one.
Usage:
Example needed! Please contribute one. |
---|
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_sofia | sofia_glue.c | 5114 |
loopback_leg
Description needed! Please contribute one.
Usage:
Example needed! Please contribute one. |
---|
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_loopback | mod_loopback.c | 11536 |
max_forwards
Contains the current Max-Forwards value provided in the originating request. The Max-Forwards value is decremented by one for each hop in a SIP call, when the Max-Forwards value is depleted the receiving agent must not pass the call onwards.
The max_forwards variable may be set on an outbound channel, some providers such as BT IP Exchange insist on a minimum value to faithfully terminate the call, 50 in BT's case.
Beware! If the max-Forwards value is reset it can cause potential cyclic calls between two operators who loop calls back to each other in error. Use wisely.
Usage:
Bridge an incoming call to a provider requiring a minimum Max-Forwards value.
<action application="bridge" data="{max_forwards=65}sofia/gateway/ipexchange/442920180123" /> |
---|
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_sofia | mod_sofia.c | 7347 |
mod_sofia | sofia.c | 4819 |
mod_sofia | sofia_glue.c | 6974 |
core | switch_core_session.c | 6976 |
core | switch_ivr.c | 8103 |
mduration
Description needed! Please contribute one.
Usage:
Example needed! Please contribute one. |
---|
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
core | switch_channel.c | 6917 |
media_webrtc
Used to instruct FS to generate an INVITE for a WebRTC call. For example, in case you need to originate a WebRTC call but you are not calling a SIP UA that is registered with FS (if the UA is registered with FS, FS knows it should originate a WebRTC call).
Usage:
# In the origination vars (commands originate or bridge) add |
---|
memory_debug
Description needed! Please contribute one.
Usage:
Example needed! Please contribute one. |
---|
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
core | switch_core_session.c | 7718 |
monitor_ring_dispo
Description needed! Please contribute one.
Usage:
Example needed! Please contribute one. |
---|
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
core | switch_ivr_originate.c | 11437 |
myid
Description needed! Please contribute one.
Usage:
Example needed! Please contribute one. |
---|
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_voicemail | mod_voicemail.c | 8689 |
NDLB_support_asterisk_missing_srtp_auth
Description needed! Please contribute one.
Usage:
Example needed! Please contribute one. |
---|
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_sofia | sofia_glue.c | 8908 |
new_name
Description needed! Please contribute one.
Usage:
Example needed! Please contribute one. |
---|
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_sofia | sofia.c | 9703 |
nonexistantvar
Description goes here
Usage:
Example needed |
---|
See also:
Implemented By:
original_caller_id_name
Description needed! Please contribute one.
Usage:
Example needed! Please contribute one. |
---|
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
core | switch_ivr_bridge.c | 7083 |
original_caller_id_number
Description needed! Please contribute one.
Usage:
Example needed! Please contribute one. |
---|
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
core | switch_ivr_bridge.c | 7083 |
original_destination_number
Description needed! Please contribute one.
Usage:
Example needed! Please contribute one. |
---|
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
core | switch_ivr_bridge.c | 7083 |
originate_continue_on_timeout
Controls wether or not a bridge should continue after timing out. Default value is false. This variable resets the timeout after each | default is to die on first timeout
Usage:
<action application="set" data="originate_continue_on_timeout=true"/>
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
core | switch_originate.c | 15121 |
originate_retries
Number of retries before giving up on originating a call.
Usage:
Example needed! Please contribute one. |
---|
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
core | switch_ivr_originate.c | 4796 |
originate_retry_sleep_ms
This will set how long FreeSWITCH is going to wait between sending invite messages to the receiving gateway. Using the value of 500 FreeSWITCH will wait 500ms between sending invite messages to the called gateway.
Usage:
<action application="set" data="originate_retry_sleep_ms=500" /> |
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See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
core | switch_ivr_originate.c | 4796 |
origination_timeout
Controls the amount of time that the bridge or originate operation will remain in "originate" state. In effect, this is a control on how long the bridge/originate will spend in total when calling multiple endpoints.
Usage:
<action application="bridge" data="{origination_timeout=10}[leg_timeout=5]sofia/internal/foo1@bar1|[leg_timeout=5]sofia/internal/foo2@bar2" /> |
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See also:
Implemented By:
Module Name | Source File | Last Updated |
---|---|---|
Core | switch_ivr_originate.c |
IMPORTANT NOTE
This variable does not exist in freeswitch source, use originate_timeout instead.
origination_uuid
You can specify the uuid of an originated call using origination_uuid. This way you can hang up the call before it is answered, since you know the uuid. Just remember you need to use the create_uuid command to generate the uuid as 2 calls with the same uuid == bad!
Usage:
originate |
---|
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
core | switch_core_session.c | 10239 |
originator
Holds the UUID of the channel that originated the call. It's used to notify a parent channel that the state of it's child has changed, hence interrupting any blocking reads on the parent. It's automatically set and read by FreeSWITCH internals, usually the user won't want to set it.
Usage:
Example needed! Please contribute one. |
---|
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
core | switch_channel.c | 4795 |
core | switch_core_session.c | 4798 |
originator_video_codec
Description needed! Please contribute one.
Usage:
Example needed! Please contribute one. |
---|
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
core | switch_core_session.c | 11731 |
other_loopback_leg_uuid
Description needed! Please contribute one.
Usage:
Example needed! Please contribute one. |
---|
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_loopback | mod_loopback.c | 12905 |
pa_hold_file
Description needed! Please contribute one.
Usage:
Example needed! Please contribute one. |
---|
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_alsa | mod_alsa.c | 5023 |
mod_portaudio | mod_portaudio.c | 4795 |
pa_ring_file
Description needed! Please contribute one.
Usage:
Example needed! Please contribute one. |
---|
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_alsa | mod_alsa.c | 5023 |
mod_portaudio | mod_portaudio.c | 4795 |
playback_terminators
Allows you to set which DTMF tones, if pressed during the playback of a file, will stop it. The default terminator is *. Keyword 'none' disables on-key termination. keyword 'any' will terminate playback when any key (1234567890*#) is pressed.
Usage:
<action application="set" data="playback_terminators=#*" /> |
---|
Note:
Git Committed 38b3f43d:
add prefix chars to playback_terminators + means include the term in the string and x means include the char and return SWITCH_STATUS_RESTART eg #+* only includes the * if you type it but not the # |
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See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_dptools | mod_dptools.c | 5888 |
pound_replace
Description needed! Please contribute one.
Usage:
Example needed! Please contribute one. |
---|
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
core | switch_ivr_play_say.c | 5908 |
You can customize the fields in FreeSWITCH's internal channels table by using the global variable presence_data_cols
Use : to separate the column names in presence_data_cols
You must define the fields in the channels table first:
ALTER TABLE channels ADD COLUMN accountcode VARCHAR(256) |
---|
<X-PRE-PROCESS cmd="set" data="presence_data_cols=accountcode:domain_name"/>
presence_id
Will instruct mod_sofia to invoke a PRESENCE_IN event that will assert the BLF lamp for the subscribed entity. This only lasts while the channel still exists so it should not be used for persistent status (i.e. a user that is DND, or an agent logged into a queue).
Usage:
<action application="set" data="presence_id=${dialed_extension}@${domain_name}"/>
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_sofia | mod_sofia.c | 6849 |
mod_sofia | sofia.c | 6808 |
core | switch_channel.c | 9363 |
profile_start_epoch
Description needed! Please contribute one.
Usage:
Example needed! Please contribute one. |
---|
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
core | switch_channel.c | 6917 |
profile_start_stamp
Description needed! Please contribute one.
Usage:
Example needed! Please contribute one. |
---|
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
core | switch_channel.c | 6917 |
profile_start_uepoch
Description needed! Please contribute one.
Usage:
Example needed! Please contribute one. |
---|
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
core | switch_channel.c | 6917 |
progress_epoch
Description needed! Please contribute one.
Usage:
Example needed! Please contribute one. |
---|
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
core | switch_channel.c | 11186 |
progress_media_epoch
Description needed! Please contribute one.
Usage:
Example needed! Please contribute one. |
---|
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
core | switch_channel.c | 11186 |
progress_media_stamp
Description needed! Please contribute one.
Usage:
Example needed! Please contribute one. |
---|
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
core | switch_channel.c | 8568 |
progress_media_uepoch
Description needed! Please contribute one.
Usage:
Example needed! Please contribute one. |
---|
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
core | switch_channel.c | 11186 |
progress_mediamsec
Description needed! Please contribute one.
Usage:
Example needed! Please contribute one. |
---|
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
core | switch_channel.c | 8576 |
progress_mediasec
Description needed! Please contribute one.
Usage:
Example needed! Please contribute one. |
---|
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
core | switch_channel.c | 8576 |
progress_mediausec
Description needed! Please contribute one.
Usage:
Example needed! Please contribute one. |
---|
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
core | switch_channel.c | 8576 |
progress_stamp
Description needed! Please contribute one.
Usage:
Example needed! Please contribute one. |
---|
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
core | switch_channel.c | 8568 |
progress_timeout
This is the maximum time we will wait before we get media (wether its early media, ringing or answer)
Usage:
<action application="set" data="progress_timeout=20" /> |
---|
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
core | switch_ivr_originate.c | 11286 |
progress_uepoch
Description needed! Please contribute one.
Usage:
Example needed! Please contribute one. |
---|
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
core | switch_channel.c | 11186 |
progressmsec
Description needed! Please contribute one.
Usage:
Example needed! Please contribute one. |
---|
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
core | switch_channel.c | 8576 |
progresssec
Description needed! Please contribute one.
Usage:
Example needed! Please contribute one. |
---|
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
core | switch_channel.c | 8576 |
progressusec
Description needed! Please contribute one.
Usage:
Example needed! Please contribute one. |
---|
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
core | switch_channel.c | 8576 |
proxy_media
Proxy Media mode puts FreeSWITCH in a "transparent proxy mode" for the RTP streams. The RTP streams still pass through FreeSWITCH (unlike bypass media mode).
NOTE: Late Negotiation (<param name="inbound-late-negotiation" value="true"/>
) must be enabled in SIP profile for this to work in the dialplan.
Usage:
<action application="set" data="proxy_media=true" /> |
---|
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_dptools | mod_dptools.c | 7717 |
rdnis
Description needed! Please contribute one.
Usage:
Example needed! Please contribute one. |
---|
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
core | switch_ivr_bridge.c | 9170 |
read_rate
Description needed! Please contribute one.
Usage:
Example needed! Please contribute one. |
---|
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
core | switch_core_codec.c | 9040 |
read_result
Description needed! Please contribute one.
Usage:
Example needed! Please contribute one. |
---|
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
core | switch_ivr_play_say.c | 9225 |
read_terminator_used
Contains the digit that the caller used to terminate a playback.
Usage:
Example needed! Please contribute one. |
---|
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
core | switch_ivr_play_say.c | 12840 |
recovery_profile_name
Taken from http://lists.freeswitch.org/pipermail/freeswitch-users/2012-December/090291.html
Its used internally so the core recovery engine knows which profile name goes with the call. The profile name to the core is just an arbitrary sub category of the call where to mod_sofia it means the sip profile name.
You personally don't really have much use for it.
Usage:
Example needed! Please contribute one. |
---|
remote_media_ip
Description needed! Please contribute one.
Usage:
Example needed! Please contribute one. |
---|
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_esf | mod_esf.c | 5057 |
mod_sofia | mod_sofia.c | 8689 |
mod_sofia | sofia_glue.c | 6370 |
remote_media_port
Description needed! Please contribute one.
Usage:
Example needed! Please contribute one. |
---|
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_esf | mod_esf.c | 5057 |
mod_sofia | mod_sofia.c | 8689 |
mod_sofia | sofia_glue.c | 6370 |
remote_video_ip
Description needed! Please contribute one.
Usage:
Example needed! Please contribute one. |
---|
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_sofia | sofia_glue.c | 5114 |
remote_video_port
Description needed! Please contribute one.
Usage:
Example needed! Please contribute one. |
---|
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_sofia | sofia_glue.c | 5114 |
ringback
This will set the unanswered aka (early media) calls. The following example uses the US ring tone defined in ~/autoload_configs/switch.conf.xml
Usage:
<action application="set" data="ringback=$${us-ring}" /> |
---|
See also:
transfer_ringback | mod_local_stream | mod_file_string
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
core | switch_ivr_originate.c | 9095 |
rss_alt_config
Description needed! Please contribute one.
Usage:
Example needed! Please contribute one. |
---|
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_rss | mod_rss.c | 6309 |
rtp_adv_audio_ip
This channel variable explicitly sets the ip address in SDP on a one-off basis. The "right" way to do this is ext-rtp-ip in the sofia profile.
Usage: <action application="set" data="rtp_adv_audio_ip=127.0.0.1"></action>
See also: https://wiki.freeswitch.org/wiki/Sofia.conf.xml#ext-rtp-ip
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_sofia | sofia_glue.c | 9088 |
rtp_autoflush
Skip timer sleeps when the socket has data ready.
Usage: <action application="set" data="rtp_autoflush=true"/>
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_sofia | sofia_glue.c | 12854 |
rtp_hold_timeout_sec
Description needed! Please contribute one.
Usage:
Example needed! Please contribute one. |
---|
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_sofia | mod_sofia.c | 10566 |
mod_sofia | sofia_glue.c | 7309 |
rtp_manual_rtp_bugs
Needed!!!
Usage:
<action application="set" data="rtp_manual_rtp_bugs=cisco_skip_mark_bit_2833" /> |
---|
See also:
RTP Issues
Variable_disable_rtp_auto_adjust
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
rtp_rewrite_timestamps
Description needed.
Usage:
See also:
http://wiki.freeswitch.org/wiki/Sofia.conf.xml#rtp-rewrite-timestamps
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_sofia | sofia_glue.c | 5501 |
rtp_stun_ping
Description needed! Please contribute one.
Usage:
Example needed! Please contribute one. |
---|
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_sofia | sofia_glue.c | 9088 |
rtp_timeout_sec
Description needed! Please contribute one.
Usage:
Example needed! Please contribute one. |
---|
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_sofia | mod_sofia.c | 10566 |
mod_sofia | sofia_glue.c | 6093 |
rtp_timer_name
Description needed! Please contribute one.
Usage:
Example needed! Please contribute one. |
---|
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_sofia | sofia_glue.c | 12676 |
signal_bridge_to
Description needed! Please contribute one.
Usage:
Example needed! Please contribute one. |
---|
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
core | switch_ivr.c | 5738 |
core | switch_ivr_bridge.c | 12671 |
sip
Description needed! Please contribute one.
Usage:
Example needed! Please contribute one. |
---|
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_sofia | mod_sofia.c | 8689 |
sip_auth_method
The request method used to authenticate with.
Known values are;
- REGISTER
Usage:
TODO
See also:
TODO
Implemented By:
TODO
sip_auth_realm
This should contain the domain/realm of the authenticated user that has triggered the event, if applicable.
Usage:
originate {sip_auth_username= |
---|
Depending on how your FreeSWITCH instance is configured, you may experience problems with this variable being incorrect or blank after calling the 'bridge' application. If this happens, you may want to force the channel variable, for example;
<action application="bridge" data="{sip_auth_realm=${sip_auth_realm}}sofia/gateway/external/2000" /> |
---|
See also:
TODO
Implemented By:
TODO
sip_auto_answer
Tells the SIP phone to auto-answer the call, if supported. Can be used for intercom support.
Usage: From dialplan/default.xml:
<action application="export" data="sip_auto_answer=true" /> |
---|
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_sofia | mod_sofia.c | 11069 |
mod_sofia | sofia.c | 5817 |
sip_call_id
Description needed! Please contribute one.
Usage:
Example needed! Please contribute one. |
---|
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_sofia | sofia.c | 6481 |
sip_codec_negotiation
Description needed! Please Expand.
sip_codec_negotiation is basically a channel variable equivalent of inbound-codec-negotiation.
sip_codec_negotiation accepts "generous", "scrooge" & "greedy" as values.
on Freeswitch versions before Feb 3 ( 74a0cfd1e101413a3941c41d04ee01d8df2ae418 ) sip_codec_negotiation will always be over-written by the value of sip-codec-negotiation ( in the sofia profile ).
sine this same revision the behaviour was change so that the channel variable will always over-ride the value set in the sip profile. This means you can change codec negotiation on a per call basis.
Usage:
| | | |
See also: Sofia_Configuration_Files#inbound-codec-negotiation
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_sofia | sofia_glue.c | 11881 |
sip_contact_host
Hostname part of the 'Contact' SIP header.
Usage:
For example, if your request header contains:
Contact: <sip:gw+test@server.example.com:5060;transport=udp;gw=test> |
---|
Then the value for this field would be:
server.example.com |
---|
See also:
TODO
Implemented By:
TODO
sip_contact_port
Port part of the 'Contact' SIP header.
Usage:
For example, if your request header contains:
Contact: <sip:gw+test@server.example.com:5060;transport=udp;gw=test> |
---|
Then the value for this field would be:
5060 |
---|
See also:
TODO
Implemented By:
TODO
sip_contact_user
Username part from the Contact SIP header.
Usage:
For example, if your request header contains:
Contact: <sip:gw+test@server.example.com:5060;transport=udp;gw=test> |
---|
Then the value for this field would be:
gw+test |
---|
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_sofia | sofia_glue.c | 9190 |
sip_copy_custom_headers
To pass some custom X-headers from B-leg to A-leg, add {sip_copy_custom_headers=true} to the dial string of the B-leg.
Usage: Set it to false to disable sending custom X- headers to your SIP gateway provider
<action application="set" data="sip_copy_custom_headers=false" /> |
---|
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_sofia | mod_sofia.c | 4350: |
mod_sofia | sofia.c | 4237 |
sip_destination_url
Description needed! Please contribute one.
Usage:
Example needed! Please contribute one. |
---|
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_sofia | mod_sofia.c | 6451 |
sip_enable_soa
For per call basis which can be set to false to disable SIP SOA from sofia and most likely result in untouched exchange of SDP.
Usage:
<action application="set" data="bypass_media=true" /> |
---|
See also:
Implemented By:
Module Name | Source File | Last Updated |
---|---|---|
sip_exclude_contact
Description needed! Please contribute one.
Usage:
Example needed! Please contribute one. |
---|
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_sofia | mod_sofia.c | 7894 |
sip_force_video_fmtp
Description needed! Please contribute one.
Usage:
Example needed! Please contribute one. |
---|
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_fsv | mod_fsv.c | 7564 |
mod_sofia | sofia_glue.c | 7565 |
sip_from_comment
Description needed! Please contribute one.
Usage:
Example needed! Please contribute one. |
---|
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_sofia | sofia.c | 7190 |
sip_from_host
Hostname part of the 'From' SIP header.
Usage:
For example, if your request header contains:
From: <sip:1000@server.example.com>;tag=0e70ba56 |
---|
Then the value for this field would be:
server.example.com |
---|
See also:
TODO
Implemented By:
TODO
sip_from_port
Port part of the 'From' SIP header.
Usage:
For example, if your request header contains:
From: <sip:1000@server.example.com:5661>;tag=0e70ba56 |
---|
Then the value for this field would be:
5661 |
---|
However, if your request header does not contain a port, then the value for this field would be your default SIP port (usually 5060) - for example;
From: <sip:1000@server.example.com>;tag=0e70ba56 |
---|
See also:
TODO
Implemented By:
TODO
sip_from_uri
The SIP URI of the endpoint sending the INVITE.
Usage:
<action application="export" data="sip_from_uri=${sip_from_uri}" /> |
---|
Some SIP providers insist on handling caller ID in a non-normal way. In most cases you can use effective_caller_id_name and effective_caller_id_number to set the caller ID on the outbound (B leg) of a bridged call. In some cases the SIP provider doesn't like that and may just show "anonymous" or some other default caller ID information.
The above example takes the From URI of an incoming SIP call and exports that value to the B leg. This is handy when you have a scenario like this:
Alice ==> FreeSWITCH bridge ==> Bob |
---|
You want Alice's caller ID information to be sent to Bob instead of FreeSWITCH's caller ID information. Usually this "just works" but if it doesn't you can try exporting sip_from_uri as above.
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_sofia | sofia_glue.c | 10397 |
sip_from_user
Username part of the 'From' SIP header.
Usage:
For example, if your request header contains:
From: <sip:1000@server.example.com>;tag=0e70ba56 |
---|
Then the value for this field would be:
1000 |
---|
See also:
TODO
Implemented By:
TODO
sip_from_user_stripped
This is the same as sip_from_user, but has the "+" sign stripped from it.
Code snippet taken from src/mod/endpoints/mod_sofia/sofia.c:
if (!zstr(from_user)) { if (*from_user == '+') { switch_channel_set_variable(channel, "sip_from_user_stripped", (const char *) (from_user + 1)); } else { switch_channel_set_variable(channel, "sip_from_user_stripped", from_user); }} |
---|
Usage:
Example needed! Please contribute one. |
---|
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_sofia | sofia.c | 4819 |
sip_gateway
Description needed! Please contribute one.
Usage:
Example needed! Please contribute one. |
---|
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_sofia | sofia.c | 11468 |
sip_gateway_name
If your call is outbound, and Leg B is routed via a gateway, then this variable will contain the gateway name.
Usage:
Call from user on internal profile, routed out via gateway 'ntl'
sip_profile_name=gatewaysip_gateway_name=ntlsofia_profile_name=internal |
---|
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_sofia | mod_sofia.c | 6489 |
sip_h_Referred-By
This is the line:
Referred-By |
---|
In the SIP message. It may be set on a transfer to a number, which then bridges somewhere else.
Description needed! Please contribute one.
Usage: You can SET it by using:
<action application="set" data="sip_h_referred-by=000@domain.com" /> |
---|
If you wish to unset it, you can use:
<action application="unset" data="sip_h_referred-by" /> |
---|
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_sofia | sofia.c | 7001 |
sip_header_name
Description needed! Please contribute one.
Usage:
Example needed! Please contribute one. |
---|
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_sofia | sofia.c | 7190 |
sip_history_info
Description needed! Please contribute one.
Usage:
Example needed! Please contribute one. |
---|
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_sofia | sofia.c | 10718 |
sip_ignore_remote_cause
Description needed! Please contribute one.
Usage:
Example needed! Please contribute one. |
---|
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_sofia | mod_sofia.c | 10090 |
sip_invite_call_id
SIP Call-ID to use when originating a call
Usage:
<action application="set" data="sip_invite_call_id=mycustomcallid" /> |
---|
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_sofia | sofia.c |
sip_local_url
Description needed! Please contribute one.
Usage:
Example needed! Please contribute one. |
---|
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_sofia | mod_sofia.c | 6451 |
sip_looped_call
True if the call has been authenticated via means other than an ACL and the current request IP/port matches the profile IP/port (see src/mod/endpoints/mod_sofia/sofia.c)
Usage: in dialplan/public.xml:
<condition field="${ |
---|
Example:
| | | |
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_sofia | sofia.c | 7078 |
sip_nat_detected
Description needed! Please contribute one.
Usage:
Example needed! Please contribute one. |
---|
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_sofia | mod_sofia.c | 8395 |
mod_sofia | sofia.c | 8488 |
mod_sofia | sofia_glue.c | 8395 |
sip_outgoing_call_id
Replaced by Variable_sip_invite_call_id
Usage:
Example needed! Please contribute one. |
---|
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_sofia | sofia_glue.c | 10416 |
sip_p_rtp_stat
Description needed! Please contribute one.
Usage:
Example needed! Please contribute one. |
---|
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_sofia | sofia.c | 8821 |
sip_profile
Name of the SIP profile which the request was received on.
Usage:
Example needed! Please contribute one. |
---|
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_dialplan_asterisk | mod_dialplan_asterisk.c | 6214 |
sip_profile_name
If your call is outbound, then this variable will contain the profile name used for the outbound channel (Leg B)
If the outbound channel is a gateway, then this variable will be set to 'gateway', and you'd need to look at 'sip_gateway_name' to get the name of the gateway.
If the outbound channel is another profile, then this variable will be set to the name of that profile.
This should NOT be confused with sofia_profile_name, which is the name of the profile for Leg A.
Usage:
Call from user on internal profile, routed out via gateway 'ntl'
sip_profile_name=gatewaysip_gateway_name=ntlsofia_profile_name=internal |
---|
Call from user on internal profile, routed out via internal profile
sip_profile_name=internalsip_gateway_name=sofia_profile_name=internal |
---|
Call from user on external profile, routed out via internal profile
sip_profile_name=internalsip_gateway_name=sofia_profile_name=external |
---|
See also:
Implemented By:
TODO
sip_received_ip
Description needed! Please contribute one.
Usage:
Example needed! Please contribute one. |
---|
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_sofia | sofia.c | 8395 |
sip_received_port
Description needed! Please contribute one.
Usage:
Example needed! Please contribute one. |
---|
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_sofia | sofia.c | 8395 |
sip_refer_reply
Description needed! Please contribute one.
Usage:
Example needed! Please contribute one. |
---|
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_sofia | mod_sofia.c | 10471 |
mod_sofia | sofia.c | 10471 |
sip_referred_by_cid
Description needed! Please contribute one.
Usage:
Example needed! Please contribute one. |
---|
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_sofia | sofia.c | 7190 |
sip_referred_by_user_stripped
Description needed! Please contribute one.
Usage:
Example needed! Please contribute one. |
---|
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_sofia | sofia.c | 7190 |
sip_reply_host
Description needed! Please contribute one.
Usage:
Example needed! Please contribute one. |
---|
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_sofia | sofia.c | 10573 |
sip_request_host
Hostname part of the SIP header.
Usage:
For example, if your request header starts with:
REGISTER sip:172.16.44.8 SIP/2.0 |
---|
Then the value of this field would be:
172.16.44.8 |
---|
See also:
TODO
Implemented By:
TODO
sip_request_port
Port part of the SIP header.
Usage:
For example, if your request header starts with:
REGISTER sip:172.16.44.8:5061 SIP/2.0 |
---|
Then the value of this field would be:
5061 |
---|
However, if your request header does not contain a port, then the value for this field would be your default SIP port (usually 5060) - for example;
REGISTER sip:172.16.44.8 SIP/2.0 |
---|
See also:
TODO
Implemented By:
TODO
sip_require_timer
FS requires timer by default on session refresh unless its t.38 re-invite. To disable the require timer on session refresh, set it to false.
Usage:
<action application="set" data="sip_require_timer=false" /> |
---|
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
sip_route_uri
This sets where the INVITE packet should actually be sent to.
This should normally be needed in situations where the TO field contains a hostname which points back at itself, and would cause an endless loop if this variable is not set.
Usage:
Very simple usage, set during bridge immediately before the endpoint to bridge to.
bridge {sip_route_uri=sip |
---|
or
A real-life example where this was needed. It changes the Request-URI and sends the INVITE packet to the correct destination by looking up the contact details for the registered endpoint. This is used to send an incoming call to a registered endpoint (PBX in this case) but set a Request-URI so the call can be routed by the receiving party. Without the sip_route_uri variable set, the call would loop back to FreeSWITCH endlessly until the originating party cancels the call.
The incoming number routes to another extension, and exports the original dialled DDI (as dialled_ddi) so this can be used with this bridge command.
bridge {sip_route_uri=sip |
---|
See also:
[ Bug Report FS-5349] (which isn't a bug as such, but gives solution to a scenario where this variable is required)
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_sofia | sofia_glue.c | 12408 |
sip_rtp_rxstat
Description needed! Please contribute one.
Usage:
Example needed! Please contribute one. |
---|
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_sofia | sofia.c | 7427 |
sip_rtp_txstat
Description needed! Please contribute one.
Usage:
Example needed! Please contribute one. |
---|
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_sofia | sofia.c | 7427 |
sip_sticky_contact
Description needed! Please contribute one.
Usage:
Example needed! Please contribute one. |
---|
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_sofia | mod_sofia.c | 8391 |
mod_sofia | sofia_glue.c | 8888 |
sip_subject
Description needed! Please contribute one.
Usage:
Example needed! Please contribute one. |
---|
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_sofia | sofia.c | 5089 |
sip_term_cause
Description needed! Please contribute one.
Usage:
Example needed! Please contribute one. |
---|
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_sofia | sofia.c | 12676 |
sip_term_status
Description needed! Please contribute one.
Usage:
Example needed! Please contribute one. |
---|
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_sofia | sofia.c | 12676 |
sip_to_comment
Description needed! Please contribute one.
Usage:
Example needed! Please contribute one. |
---|
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_sofia | sofia.c | 7190 |
sip_to_host
Hostname part of the 'To' SIP header.
Usage:
For example, if your request header contains:
To: <sip:1000@server.example.com>;tag=0e70ba56 |
---|
Then the value for this field would be:
server.example.com |
---|
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_sofia | mod_sofia.c | 7067 |
mod_sofia | sofia.c | 6874 |
sip_to_port
Port part of the 'To' SIP header.
Usage:
For example, if your request header contains:
To: <sip:1000@server.example.com:5661>;tag=0e70ba56 |
---|
Then the value for this field would be:
5661 |
---|
However, if your request header does not contain a port, then the value for this field would be your default SIP port (usually 5060) - for example;
To: <sip:1000@server.example.com>;tag=0e70ba56 |
---|
See also:
TODO
Implemented By:
TODO
sip_to_uri
Description needed! Please contribute one.
Usage:
Example needed! Please contribute one. |
---|
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_sofia | mod_sofia.c | 7347 |
mod_sofia | sofia.c | 6874 |
sip_to_user
Username part of the 'To' SIP header.
Usage:
To set manually, use:
<action application="set" data="sip_to_user=whatevah" /> |
---|
For example, if your request header contains:
To: <sip:1000@server.example.com>;tag=0e70ba56 |
---|
Then the value for this field would be:
1000 |
---|
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_sofia | sofia.c | 6874 |
sip_transport
Description needed! Please contribute one.
Usage:
Example needed! Please contribute one. |
---|
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_sofia | sofia_glue.c | 7268 |
sip_use_gateway
Description needed! Please contribute one.
Usage:
Example needed! Please contribute one. |
---|
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_sofia | sofia.c | 10222 |
mod_sofia | sofia_reg.c | 9068 |
sip_user_agent
User agent part of the SIP header.
Usage:
For example, if your request header contains:
User-Agent: FreeSWITCH-mod_sofia/1.2.0-rc2+git~20120713T162602Z~0afd7318bd+unclean~20120713T184029Z |
---|
Then the value of this field would be:
FreeSWITCH-mod_sofia/1.2.0-rc2+git~20120713T162602Z~0afd7318bd+unclean~20120713T184029Z |
---|
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_sofia | sofia.c | 9170 |
sip_via_host
Description needed! Please contribute one.
Usage:
Example needed! Please contribute one. |
---|
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_sofia | sofia.c | 4819 |
sip_via_port
Description needed! Please contribute one.
Usage:
Example needed! Please contribute one. |
---|
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_sofia | sofia.c | 4819 |
sip_via_protocol
Description needed! Please contribute one.
Usage:
Example needed! Please contribute one. |
---|
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_sofia | sofia.c | 9923 |
sip_via_rport
Description needed! Please contribute one.
Usage:
Example needed! Please contribute one. |
---|
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_sofia | sofia.c | 4819 |
sip_video_fmtp
Description needed! Please contribute one.
Usage:
Example needed! Please contribute one. |
---|
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_sofia | sofia_glue.c | 7546 |
sip_video_pt
Description needed! Please contribute one.
Usage:
Example needed! Please contribute one. |
---|
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_sofia | sofia_glue.c | 7582 |
socket_host
Description needed! Please contribute one.
Usage:
Example needed! Please contribute one. |
---|
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_event_socket | mod_event_socket.c | 7766 |
socket_path
Description needed! Please contribute one.
Usage:
Example needed! Please contribute one. |
---|
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_event_socket | mod_event_socket.c | 7749 |
SOFIA_CRYPTO_MANDATORY_VARIABLE
Description needed! Please contribute one.
Usage:
Example needed! Please contribute one. |
---|
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_sofia | sofia_glue.c | 7746 |
SOFIA_HAS_CRYPTO_VARIABLE
Description needed! Please contribute one.
Usage:
Example needed! Please contribute one. |
---|
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_sofia | sofia_glue.c | 7244 |
sofia_profile_domain_name
Description needed! Please contribute one.
Usage:
Example needed! Please contribute one. |
---|
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_sofia | sofia.c | 7313 |
sofia_profile_name
The name of the profile of which the call originated (Leg A).
This should NOT be confused with sip_profile_name.
Usage:
Call from user on external profile
sofia_profile_name=external |
---|
Call from user on internal profile
sofia_profile_name=internal |
---|
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_sofia | sofia.c | 5739 |
mod_sofia | sofia_glue.c | 5739 |
sofia_record_file
Description needed! Please contribute one.
Usage:
Example needed! Please contribute one. |
---|
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_sofia | sofia.c | 6952 |
SOFIA_REFER_TO_VARIABLE
Description needed! Please contribute one.
Usage:
Example needed! Please contribute one. |
---|
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_sofia | sofia.c | 7008 |
SOFIA_REPLACES_HEADER
Description needed! Please contribute one.
Usage:
Example needed! Please contribute one. |
---|
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_sofia | sofia.c | 9530 |
mod_sofia | sofia_glue.c | 6120 |
SOFIA_SECURE_MEDIA_CONFIRMED_VARIABLE
Description needed! Please contribute one.
Usage:
Example needed! Please contribute one. |
---|
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_sofia | sofia_glue.c | 7244 |
SOFIA_SECURE_MEDIA_VARIABLE
Description needed! Please contribute one.
Usage:
Example needed! Please contribute one. |
---|
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_sofia | mod_sofia.c | 10228 |
mod_sofia | sofia_glue.c | 7779 |
SOFIA_SESSION_TIMEOUT
Description needed! Please contribute one.
Usage:
Example needed! Please contribute one. |
---|
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_sofia | mod_sofia.c | 6347 |
mod_sofia | sofia_glue.c | 6003 |
sound_prefix
Directory prefix where the sounds lives.
Usage:
Example needed! Please contribute one. |
---|
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
core | switch_ivr_play_say.c | 11108 |
star_replace
Description needed! Please contribute one.
Usage:
Example needed! Please contribute one. |
---|
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
core | switch_ivr_play_say.c | 5908 |
start_epoch
Description needed! Please contribute one.
Usage:
Example needed! Please contribute one. |
---|
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
core | switch_channel.c | 6917 |
start_stamp
Description needed! Please contribute one.
Usage:
Example needed! Please contribute one. |
---|
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
core | switch_channel.c | 6917 |
start_uepoch
Description needed! Please contribute one.
Usage:
Example needed! Please contribute one. |
---|
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
core | switch_channel.c | 6917 |
stream_prebuffer
Description needed! Please contribute one.
Usage:
Example needed! Please contribute one. |
---|
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_spidermonkey | mod_spidermonkey.c | 8689 |
core | switch_cpp.cpp | 5374 |
core | switch_ivr_play_say.c | 5302 |
supress_cng
Description needed! Please contribute one.
Usage:
Example needed! Please contribute one. |
---|
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_sofia | sofia_glue.c | 9648 |
SWITCH_PLAYBACK_TERMINATOR_USED
Description needed! Please contribute one.
Usage:
Example needed! Please contribute one. |
---|
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_dptools | mod_dptools.c | 10489 |
SWITCH_UUID_BRIDGE
Description needed! Please contribute one.
Usage:
Example needed! Please contribute one. |
---|
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
core | switch_ivr_bridge.c | 8686 |
sip_auto_simplify
If FreeSWITCH detects that it can remove itself from the SIP and RTP path (such as the haripin scenario) then it will issue REINVITEs the necessary endpoints.
Usage:
<action application="set" data="timer_name=soft" /> |
---|
See also:
uuid_simplify
tone_detect_expires
Description needed! Please contribute one.
Usage:
Example needed! Please contribute one. |
---|
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
core | switch_ivr_async.c | 11309 |
tone_detect_sleep
Description needed! Please contribute one.
Usage:
Example needed! Please contribute one. |
---|
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
core | switch_ivr_async.c | 11309 |
transfer_fallback_extension
It's an extension the channel falls back to on failed transfer. Set it before the transfer. For example, if you transfer to some invalid or unavailable ext, it will then "fall back" to the ext set in the var.
Usage:
Example needed! Please contribute one. |
---|
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_sofia | sofia.c | 6908 |
transfer_history
Description goes here
Usage:
Example needed.
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
? |
transfer_ringback
This will set the ring tone for answered calls. This is any call that has been setup. One example would be the tone to play during transfer. The following example uses the French ringtone defined in ~/autoload_configs/switch.conf.xml
Usage:
<action application="set" data="transfer_ringback=$${fr-ring}" /> |
---|
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
core | switch_ivr_originate.c | 9095 |
transfer_source
Description goes here
Usage:
Example needed.
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
? |
tts_engine
Description needed! Please contribute one.
Usage:
Example needed! Please contribute one. |
---|
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_dptools | mod_dptools.c | 5876 |
core | switch_ivr_menu.c | 12297 |
core | switch_ivr_play_say.c | 8686 |
tts_voice
Description needed! Please contribute one.
Usage:
Example needed! Please contribute one. |
---|
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_dptools | mod_dptools.c | 5876 |
core | switch_ivr_menu.c | 12297 |
core | switch_ivr_play_say.c | 8686 |
uduration
Description needed! Please contribute one.
Usage:
Example needed! Please contribute one. |
---|
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
core | switch_channel.c | 6917 |
UNIQUEID
Description needed! Please contribute one.
Usage:
Example needed! Please contribute one. |
---|
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_dialplan_asterisk | mod_dialplan_asterisk.c | 6205 |
user_context
Description needed! Please contribute one.
Usage:
Example needed! Please contribute one. |
---|
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_dptools | mod_dptools.c | 6641 |
mod_sofia | sofia.c | 6641 |
user_name
Description needed! Please contribute one.
Usage:
Example needed! Please contribute one. |
---|
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
core | switch_ivr.c | 9056 |
verbose_presence
Description needed! Please contribute one.
Usage:
Example needed! Please contribute one. |
---|
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
core | switch_core_session.c | 9363 |
video_possible
Description needed! Please contribute one.
Usage:
Example needed! Please contribute one. |
---|
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_sofia | sofia_glue.c | 8542 |
video_read_codec
Description needed! Please contribute one.
Usage:
Example needed! Please contribute one. |
---|
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
core | switch_core_codec.c | 7565 |
video_read_rate
Description needed! Please contribute one.
Usage:
Example needed! Please contribute one. |
---|
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
core | switch_core_codec.c | 7565 |
video_write_codec
Description needed! Please contribute one.
Usage:
Example needed! Please contribute one. |
---|
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
core | switch_core_codec.c | 7565 |
video_write_rate
Description needed! Please contribute one.
Usage:
Example needed! Please contribute one. |
---|
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
core | switch_core_codec.c | 7565 |
vm_cc
Description needed! Please contribute one.
Usage:
Example needed! Please contribute one. |
---|
See also: Mod_voicemail#vm_cc
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_voicemail | mod_voicemail.c | 12850 |
vm_message_ext
Determines in which format the voicemail message is saved.
This variable is set in the dial plan.
Valid values are (not exhaustive):
- mp3
- wav
Usage example:
<action application="set" data="vm_message_ext=mp3"/>
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_voicemail | mod_voicemail.c | 9098 |
vmd_detect
Description needed! Please contribute one.
Usage:
Example needed! Please contribute one. |
---|
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_vmd | mod_vmd.c | 10643 |
vname
Description needed! Please contribute one.
Usage:
Example needed! Please contribute one. |
---|
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
core | switch_channel.c | 8776 |
voicemail_account
Description needed! Please contribute one.
Usage:
Example needed! Please contribute one. |
---|
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_voicemail | mod_voicemail.c | 6984 |
voicemail_alternate_greet_id
Overrides the ID the voicemail application reads back. I.e. to say a phone number instead of the user ID.
Usage:
<user id="johnsmith" number-alias="1000"> |
---|
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_voicemail | mod_voicemail.c | 10594 |
voicemail_authorized
Description needed! Please contribute one.
Usage:
Example needed! Please contribute one. |
---|
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_voicemail | mod_voicemail.c | 6804 |
voicemail_caller_id_name
Description needed! Please contribute one.
Usage:
Example needed! Please contribute one. |
---|
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_voicemail | mod_voicemail.c | 6984 |
voicemail_caller_id_number
Description needed! Please contribute one.
Usage:
Example needed! Please contribute one. |
---|
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_voicemail | mod_voicemail.c | 6984 |
voicemail_current_folder
Description needed! Please contribute one.
Usage:
Example needed! Please contribute one. |
---|
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_voicemail | mod_voicemail.c | 8689 |
voicemail_domain
Description needed! Please contribute one.
Usage: This variable will change the domain name in the sender email address, when mod_voicemail emails a voicemail message.
from conf/directory/default/1010.xml:
<include> <user id="1010" mailbox="1010"> <params> <param name="password" value="password" /> <param name="vm-password" value="1010" /> </params> <variables> <variable name="voicemail_domain" value="speakblast.com" /> </variables> </user></include> |
---|
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_voicemail | mod_voicemail.c | 6984 |
voicemail_domain_name
Description needed! Please contribute one.
Usage:
Example needed! Please contribute one. |
---|
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_voicemail | mod_voicemail.c | 6984 |
voicemail_email
Description needed! Please contribute one.
Usage:
Example needed! Please contribute one. |
---|
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_voicemail | mod_voicemail.c | 6984 |
voicemail_file_path
Description needed! Please contribute one.
Usage:
Example needed! Please contribute one. |
---|
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_voicemail | mod_voicemail.c | 6984 |
voicemail_greeting_number
Description needed! Please contribute one.
Usage:
Example needed! Please contribute one. |
---|
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_voicemail | mod_voicemail.c | 6984 |
voicemail_greeting_path
Description needed! Please contribute one.
Usage:
Example needed! Please contribute one. |
---|
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_voicemail | mod_voicemail.c | 10569 |
voicemail_id
Description needed! Please contribute one.
Usage:
Example needed! Please contribute one. |
---|
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_voicemail | mod_voicemail.c | 6984 |
voicemail_message_len
Description needed! Please contribute one.
Usage:
Example needed! Please contribute one. |
---|
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_voicemail | mod_voicemail.c | 6984 |
voicemail_priority
Description needed! Please contribute one.
Usage:
Example needed! Please contribute one. |
---|
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_voicemail | mod_voicemail.c | 6984 |
voicemail_profile_name
Description needed! Please contribute one.
Usage:
Example needed! Please contribute one. |
---|
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_voicemail | mod_voicemail.c | 6984 |
voicemail_read_flags
Description needed! Please contribute one.
Usage:
Example needed! Please contribute one. |
---|
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_voicemail | mod_voicemail.c | 6984 |
voicemail_time
Description needed! Please contribute one.
Usage:
Example needed! Please contribute one. |
---|
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_voicemail | mod_voicemail.c | 6984 |
voicemail_total_new_messages
Description needed! Please contribute one.
Usage:
Example needed! Please contribute one. |
---|
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_voicemail | mod_voicemail.c | 6984 |
voicemail_total_saved_messages
Description needed! Please contribute one.
Usage:
Example needed! Please contribute one. |
---|
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_voicemail | mod_voicemail.c | 6984 |
voicemail_urgent_new_messages
Description needed! Please contribute one.
Usage:
Example needed! Please contribute one. |
---|
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_voicemail | mod_voicemail.c | 6984 |
voicemail_urgent_saved_messages
Description needed! Please contribute one.
Usage:
Example needed! Please contribute one. |
---|
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_voicemail | mod_voicemail.c | 6984 |
waitmsec
Description needed! Please contribute one.
Usage:
Example needed! Please contribute one. |
---|
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
core | switch_channel.c | 3603 |
waitsec
Description needed! Please contribute one.
Usage:
Example needed! Please contribute one. |
---|
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
core | switch_channel.c | 3582 |
waitusec
Description needed! Please contribute one.
Usage:
Example needed! Please contribute one. |
---|
See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
core | switch_channel.c | 3624 |
write_rate
Description needed! Please contribute one.
Usage:
Example needed! Please contribute one. |
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See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
core | switch_core_codec.c | 10045 |
xfer_uuids
Description needed! Please contribute one.
Usage:
Example needed! Please contribute one. |
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See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_dptools | mod_dptools.c | 7885 |
xml_cdr_base
Description needed! Please contribute one.
Usage:
Example needed! Please contribute one. |
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See also:
Implemented By:
Module Name | Source File | Last Revised |
---|---|---|
mod_xml_cdr | mod_xml_cdr.c | 6708 |
https://freeswitch.org/confluence/display/FREESWITCH/Variables