Guys, I made a list by grepping and perling around in the source. grep -Pr "c(.)(\s+)switch_channel_set_variable\(channel, \"" * | grep -v "Binary" > /tmp/chan_vars.tmp This gave me a massive list of all filenames and occurrences of "switch_channel_set_variable" with a channel variable explicitly named in quotes. I then filtered out stuff with svn in the source filename and then create a tab-separated list of chan variables and source filenames. Finally, I distilled that list with this script: perl -n -e 'm/(\w+)\t(.*)/;next if (exists($seen{$1}));print $1 . "\t" . $2 . "\n";$seen{$1}=$2;' /tmp/chan_vars3.tmp This yielded a nice list of variables. Let me know what you think about using this as the base list for channel variables on the wiki: Variable Source File: limit_realm mod/applications/mod_limit/mod_limit.c limit_id mod/applications/mod_limit/mod_limit.c limit_max mod/applications/mod_limit/mod_limit.c amd_result mod/applications/mod_amd/mod_amd.c amd_status mod/applications/mod_amd/mod_amd.c amd_wait_status mod/applications/mod_amd/mod_amd.c amd_wait_result mod/applications/mod_amd/mod_amd.c enum_route_count mod/applications/mod_enum/mod_enum.c enum_auto_route mod/applications/mod_enum/mod_enum.c voicemail_total_new_messages mod/applications/mod_voicemail/mod_voicemail.c voicemail_total_saved_messages mod/applications/mod_voicemail/mod_voicemail.c voicemail_urgent_new_messages mod/applications/mod_voicemail/mod_voicemail.c voicemail_urgent_saved_messages mod/applications/mod_voicemail/mod_voicemail.c voicemail_current_folder mod/applications/mod_voicemail/mod_voicemail.c voicemail_account mod/applications/mod_voicemail/mod_voicemail.c voicemail_domain mod/applications/mod_voicemail/mod_voicemail.c voicemail_caller_id_number mod/applications/mod_voicemail/mod_voicemail.c voicemail_caller_id_name mod/applications/mod_voicemail/mod_voicemail.c voicemail_file_path mod/applications/mod_voicemail/mod_voicemail.c voicemail_read_flags mod/applications/mod_voicemail/mod_voicemail.c voicemail_time mod/applications/mod_voicemail/mod_voicemail.c voicemail_priority mod/applications/mod_voicemail/mod_voicemail.c voicemail_message_len mod/applications/mod_voicemail/mod_voicemail.c voicemail_email mod/applications/mod_voicemail/mod_voicemail.c RECORD_ARTIST mod/applications/mod_voicemail/mod_voicemail.c RECORD_DATE mod/applications/mod_voicemail/mod_voicemail.c RECORD_SOFTWARE mod/applications/mod_voicemail/mod_voicemail.c RECORD_TITLE mod/applications/mod_voicemail/mod_voicemail.c RECORD_COMMENT mod/applications/mod_voicemail/mod_voicemail.c RECORD_COPYRIGHT mod/applications/mod_voicemail/mod_voicemail.c voicemail_notify_email mod/applications/mod_voicemail/mod_voicemail.c mailbox mod/applications/mod_dptools/mod_dptools.c user_name mod/applications/mod_dptools/mod_dptools.c domain_name mod/applications/mod_dptools/mod_dptools.c xfer_uuids mod/applications/mod_dptools/mod_dptools.c dialed_user mod/applications/mod_dptools/mod_dptools.c dialed_domain mod/applications/mod_dptools/mod_dptools.c sip_force_video_fmtp mod/applications/mod_fsv/mod_fsv.c fifo_priority mod/applications/mod_fifo/mod_fifo.c fifo_status mod/applications/mod_fifo/mod_fifo.c fifo_timestamp mod/applications/mod_fifo/mod_fifo.c fifo_target mod/applications/mod_fifo/mod_fifo.c dl_user mod/endpoints/mod_dingaling/mod_dingaling.c dl_host mod/endpoints/mod_dingaling/mod_dingaling.c dl_from_user mod/endpoints/mod_dingaling/mod_dingaling.c dl_from_host mod/endpoints/mod_dingaling/mod_dingaling.c sip_nat_detected mod/endpoints/mod_sofia/sofia_glue.c sofia_profile_name mod/endpoints/mod_sofia/sofia_glue.c auto_answer_destination mod/endpoints/mod_sofia/sofia.c sip_user_agent mod/endpoints/mod_sofia/sofia.c sip_rtp_txstat mod/endpoints/mod_sofia/sofia.c sip_rtp_rxstat mod/endpoints/mod_sofia/sofia.c sip_call_id mod/endpoints/mod_sofia/sofia.c sip_authorized mod/endpoints/mod_sofia/sofia.c sip_term_status mod/endpoints/mod_sofia/sofia.c sip_term_cause mod/endpoints/mod_sofia/sofia.c transfer_fallback_extension mod/endpoints/mod_sofia/sofia.c call_clientcode mod/endpoints/mod_sofia/sofia.c sofia_record_file mod/endpoints/mod_sofia/sofia.c sip_received_ip mod/endpoints/mod_sofia/sofia.c sip_received_port mod/endpoints/mod_sofia/sofia.c sip_looped_call mod/endpoints/mod_sofia/sofia.c sip_from_user_stripped mod/endpoints/mod_sofia/sofia.c sip_from_comment mod/endpoints/mod_sofia/sofia.c sofia_profile_domain_name mod/endpoints/mod_sofia/sofia.c sip_to_comment mod/endpoints/mod_sofia/sofia.c sip_referred_by_user_stripped mod/endpoints/mod_sofia/sofia.c sip_referred_by_cid mod/endpoints/mod_sofia/sofia.c sip_subject mod/endpoints/mod_sofia/sofia.c sip_via_host mod/endpoints/mod_sofia/sofia.c sip_via_port mod/endpoints/mod_sofia/sofia.c sip_via_rport mod/endpoints/mod_sofia/sofia.c sip_gateway mod/endpoints/mod_sofia/sofia.c alert_info mod/endpoints/mod_sofia/sofia.c presence_id mod/endpoints/mod_sofia/sofia.c call_timeout mod/dialplans/mod_dialplan_asterisk/mod_dialplan_asterisk.c EXTEN mod/dialplans/mod_dialplan_asterisk/mod_dialplan_asterisk.c CHANNEL mod/dialplans/mod_dialplan_asterisk/mod_dialplan_asterisk.c UNIQUEID mod/dialplans/mod_dialplan_asterisk/mod_dialplan_asterisk.c socket_path mod/event_handlers/mod_event_socket/mod_event_socket.c socket_host mod/event_handlers/mod_event_socket/mod_event_socket.c last_file_position mod/languages/mod_spidermonkey/mod_spidermonkey.c hangup_cause switch_channel.c endpoint_disposition switch_channel.c start_stamp switch_channel.c profile_start_stamp switch_channel.c answer_stamp switch_channel.c progress_stamp switch_channel.c progress_media_stamp switch_channel.c end_stamp switch_channel.c start_epoch switch_channel.c start_uepoch switch_channel.c profile_start_epoch switch_channel.c profile_start_uepoch switch_channel.c answer_epoch switch_channel.c answer_uepoch switch_channel.c end_epoch switch_channel.c end_uepoch switch_channel.c last_app switch_channel.c last_arg switch_channel.c caller_id switch_channel.c duration switch_channel.c billsec switch_channel.c progresssec switch_channel.c progress_mediasec switch_channel.c flow_billsec switch_channel.c mduration switch_channel.c billmsec switch_channel.c progressmsec switch_channel.c progress_mediamsec switch_channel.c flow_billmsec switch_channel.c uduration switch_channel.c billusec switch_channel.c progressusec switch_channel.c progress_mediausec switch_channel.c flow_billusec switch_channel.c read_codec switch_core_codec.c read_rate switch_core_codec.c write_codec switch_core_codec.c write_rate switch_core_codec.c video_read_codec switch_core_codec.c video_read_rate switch_core_codec.c video_write_codec switch_core_codec.c video_write_rate switch_core_codec.c sound_prefix switch_ivr_play_say.c
Forgot the ones in switch_types.h ;) /b On Jun 6, 2008, at 11:41 AM, Michael Collins wrote:
Guys,
I made a list by grepping and perling around in the source.
grep -Pr "c(.)(\s+)switch_channel_set_variable\(channel, \"" * | grep -v "Binary" > /tmp/chan_vars.tmp
This gave me a massive list of all filenames and occurrences of “switch_channel_set_variable” with a channel variable explicitly named in quotes.
I then filtered out stuff with svn in the source filename and then create a tab-separated list of chan variables and source filenames. Finally, I distilled that list with this script:
perl -n -e 'm/(\w+)\t(.*)/;next if (exists($seen{$1}));print $1 . "\t" . $2 . "\n";$seen{$1}=$2;' /tmp/chan_vars3.tmp
This yielded a nice list of variables. Let me know what you think about using this as the base list for channel variables on the wiki:
Variable Source File: limit_realm mod/applications/mod_limit/ mod_limit.c limit_id mod/applications/mod_limit/ mod_limit.c limit_max mod/applications/mod_limit/ mod_limit.c amd_result mod/applications/mod_amd/mod_amd.c amd_status mod/applications/mod_amd/mod_amd.c amd_wait_status mod/applications/mod_amd/mod_amd.c amd_wait_result mod/applications/mod_amd/mod_amd.c enum_route_count mod/applications/mod_enum/ mod_enum.c enum_auto_route mod/applications/mod_enum/ mod_enum.c voicemail_total_new_messages mod/applications/mod_voicemail/ mod_voicemail.c voicemail_total_saved_messages mod/applications/mod_voicemail/ mod_voicemail.c voicemail_urgent_new_messages mod/applications/mod_voicemail/ mod_voicemail.c voicemail_urgent_saved_messages mod/applications/mod_voicemail/ mod_voicemail.c voicemail_current_folder mod/applications/mod_voicemail/ mod_voicemail.c voicemail_account mod/applications/mod_voicemail/ mod_voicemail.c voicemail_domain mod/applications/mod_voicemail/ mod_voicemail.c voicemail_caller_id_number mod/applications/mod_voicemail/ mod_voicemail.c voicemail_caller_id_name mod/applications/mod_voicemail/ mod_voicemail.c voicemail_file_path mod/applications/mod_voicemail/ mod_voicemail.c voicemail_read_flags mod/applications/mod_voicemail/ mod_voicemail.c voicemail_time mod/applications/mod_voicemail/ mod_voicemail.c voicemail_priority mod/applications/mod_voicemail/ mod_voicemail.c voicemail_message_len mod/applications/mod_voicemail/ mod_voicemail.c voicemail_email mod/applications/mod_voicemail/ mod_voicemail.c RECORD_ARTIST mod/applications/mod_voicemail/ mod_voicemail.c RECORD_DATE mod/applications/mod_voicemail/ mod_voicemail.c RECORD_SOFTWARE mod/applications/mod_voicemail/ mod_voicemail.c RECORD_TITLE mod/applications/mod_voicemail/ mod_voicemail.c RECORD_COMMENT mod/applications/mod_voicemail/ mod_voicemail.c RECORD_COPYRIGHT mod/applications/mod_voicemail/ mod_voicemail.c voicemail_notify_email mod/applications/mod_voicemail/ mod_voicemail.c mailbox mod/applications/mod_dptools/ mod_dptools.c user_name mod/applications/mod_dptools/ mod_dptools.c domain_name mod/applications/mod_dptools/ mod_dptools.c xfer_uuids mod/applications/mod_dptools/ mod_dptools.c dialed_user mod/applications/mod_dptools/ mod_dptools.c dialed_domain mod/applications/mod_dptools/ mod_dptools.c sip_force_video_fmtp mod/applications/mod_fsv/mod_fsv.c fifo_priority mod/applications/mod_fifo/ mod_fifo.c fifo_status mod/applications/mod_fifo/ mod_fifo.c fifo_timestamp mod/applications/mod_fifo/ mod_fifo.c fifo_target mod/applications/mod_fifo/ mod_fifo.c dl_user mod/endpoints/mod_dingaling/ mod_dingaling.c dl_host mod/endpoints/mod_dingaling/ mod_dingaling.c dl_from_user mod/endpoints/mod_dingaling/ mod_dingaling.c dl_from_host mod/endpoints/mod_dingaling/ mod_dingaling.c sip_nat_detected mod/endpoints/mod_sofia/ sofia_glue.c sofia_profile_name mod/endpoints/mod_sofia/ sofia_glue.c auto_answer_destination mod/endpoints/mod_sofia/sofia.c sip_user_agent mod/endpoints/mod_sofia/sofia.c sip_rtp_txstat mod/endpoints/mod_sofia/sofia.c sip_rtp_rxstat mod/endpoints/mod_sofia/sofia.c sip_call_id mod/endpoints/mod_sofia/sofia.c sip_authorized mod/endpoints/mod_sofia/sofia.c sip_term_status mod/endpoints/mod_sofia/sofia.c sip_term_cause mod/endpoints/mod_sofia/sofia.c transfer_fallback_extension mod/endpoints/mod_sofia/sofia.c call_clientcode mod/endpoints/mod_sofia/sofia.c sofia_record_file mod/endpoints/mod_sofia/sofia.c sip_received_ip mod/endpoints/mod_sofia/sofia.c sip_received_port mod/endpoints/mod_sofia/sofia.c sip_looped_call mod/endpoints/mod_sofia/sofia.c sip_from_user_stripped mod/endpoints/mod_sofia/sofia.c sip_from_comment mod/endpoints/mod_sofia/sofia.c sofia_profile_domain_name mod/endpoints/mod_sofia/sofia.c sip_to_comment mod/endpoints/mod_sofia/sofia.c sip_referred_by_user_stripped mod/endpoints/mod_sofia/sofia.c sip_referred_by_cid mod/endpoints/mod_sofia/sofia.c sip_subject mod/endpoints/mod_sofia/sofia.c sip_via_host mod/endpoints/mod_sofia/sofia.c sip_via_port mod/endpoints/mod_sofia/sofia.c sip_via_rport mod/endpoints/mod_sofia/sofia.c sip_gateway mod/endpoints/mod_sofia/sofia.c alert_info mod/endpoints/mod_sofia/sofia.c presence_id mod/endpoints/mod_sofia/sofia.c call_timeout mod/dialplans/ mod_dialplan_asterisk/mod_dialplan_asterisk.c EXTEN mod/dialplans/ mod_dialplan_asterisk/mod_dialplan_asterisk.c CHANNEL mod/dialplans/ mod_dialplan_asterisk/mod_dialplan_asterisk.c UNIQUEID mod/dialplans/ mod_dialplan_asterisk/mod_dialplan_asterisk.c socket_path mod/event_handlers/ mod_event_socket/mod_event_socket.c socket_host mod/event_handlers/ mod_event_socket/mod_event_socket.c last_file_position mod/languages/mod_spidermonkey/ mod_spidermonkey.c hangup_cause switch_channel.c endpoint_disposition switch_channel.c start_stamp switch_channel.c profile_start_stamp switch_channel.c answer_stamp switch_channel.c progress_stamp switch_channel.c progress_media_stamp switch_channel.c end_stamp switch_channel.c start_epoch switch_channel.c start_uepoch switch_channel.c profile_start_epoch switch_channel.c profile_start_uepoch switch_channel.c answer_epoch switch_channel.c answer_uepoch switch_channel.c end_epoch switch_channel.c end_uepoch switch_channel.c last_app switch_channel.c last_arg switch_channel.c caller_id switch_channel.c duration switch_channel.c billsec switch_channel.c progresssec switch_channel.c progress_mediasec switch_channel.c flow_billsec switch_channel.c mduration switch_channel.c billmsec switch_channel.c progressmsec switch_channel.c progress_mediamsec switch_channel.c flow_billmsec switch_channel.c uduration switch_channel.c billusec switch_channel.c progressusec switch_channel.c progress_mediausec switch_channel.c flow_billusec switch_channel.c read_codec switch_core_codec.c read_rate switch_core_codec.c write_codec switch_core_codec.c write_rate switch_core_codec.c video_read_codec switch_core_codec.c video_read_rate switch_core_codec.c video_write_codec switch_core_codec.c video_write_rate switch_core_codec.c sound_prefix switch_ivr_play_say.c
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Oops! How about this: SWITCH_PROTO_SPECIFIC_HANGUP_CAUSE_VARIABLE "proto_specific_hangup_cause" SWITCH_CHANNEL_EXECUTE_ON_ANSWER_VARIABLE "execute_on_answer" SWITCH_CHANNEL_EXECUTE_ON_RING_VARIABLE "execute_on_ring" SWITCH_CALL_TIMEOUT_VARIABLE "call_timeout" SWITCH_HOLDING_UUID_VARIABLE "holding_uuid" SWITCH_API_BRIDGE_END_VARIABLE "api_after_bridge" SWITCH_API_HANGUP_HOOK_VARIABLE "api_hangup_hook" SWITCH_PROCESS_CDR_VARIABLE "process_cdr" SWITCH_BRIDGE_CHANNEL_VARIABLE "bridge_channel" SWITCH_CHANNEL_NAME_VARIABLE "channel_name" SWITCH_BRIDGE_UUID_VARIABLE "bridge_uuid" SWITCH_PLAYBACK_TERMINATORS_VARIABLE "playback_terminators" SWITCH_CACHE_SPEECH_HANDLES_VARIABLE "cache_speech_handles" SWITCH_BYPASS_MEDIA_VARIABLE "bypass_media" SWITCH_PROXY_MEDIA_VARIABLE "proxy_media" SWITCH_ENDPOINT_DISPOSITION_VARIABLE "endpoint_disposition" SWITCH_HOLD_MUSIC_VARIABLE "hold_music" SWITCH_EXPORT_VARS_VARIABLE "export_vars" SWITCH_R_SDP_VARIABLE "switch_r_sdp" SWITCH_L_SDP_VARIABLE "switch_l_sdp" SWITCH_B_SDP_VARIABLE "switch_m_sdp" SWITCH_BRIDGE_VARIABLE "bridge_to" SWITCH_SIGNAL_BRIDGE_VARIABLE "signal_bridge_to" SWITCH_SIGNAL_BOND_VARIABLE "signal_bond" SWITCH_ORIGINATOR_VARIABLE "originator" SWITCH_ORIGINATOR_CODEC_VARIABLE "originator_codec" SWITCH_LOCAL_MEDIA_IP_VARIABLE "local_media_ip" SWITCH_LOCAL_MEDIA_PORT_VARIABLE "local_media_port" SWITCH_REMOTE_MEDIA_IP_VARIABLE "remote_media_ip" SWITCH_REMOTE_MEDIA_PORT_VARIABLE "remote_media_port" SWITCH_REMOTE_VIDEO_IP_VARIABLE "remote_video_ip" SWITCH_REMOTE_VIDEO_PORT_VARIABLE "remote_video_port" SWITCH_LOCAL_VIDEO_IP_VARIABLE "local_video_ip" SWITCH_LOCAL_VIDEO_PORT_VARIABLE "local_video_port" SWITCH_HANGUP_AFTER_BRIDGE_VARIABLE "hangup_after_bridge" SWITCH_EXEC_AFTER_BRIDGE_APP_VARIABLE "exec_after_bridge_app" SWITCH_EXEC_AFTER_BRIDGE_ARG_VARIABLE "exec_after_bridge_arg" SWITCH_MAX_FORWARDS_VARIABLE "max_forwards" SWITCH_DISABLE_APP_LOG_VARIABLE "disable_app_log" ________________________________ From: freeswitch-dev-bounces@lists.freeswitch.org [mailto:freeswitch-dev-bounces@lists.freeswitch.org] On Behalf Of Brian West Sent: Friday, June 06, 2008 9:58 AM To: freeswitch-dev@lists.freeswitch.org Subject: Re: [Freeswitch-dev] INFO: Channel variables Forgot the ones in switch_types.h ;) /b On Jun 6, 2008, at 11:41 AM, Michael Collins wrote: Guys, I made a list by grepping and perling around in the source. grep -Pr "c(.)(\s+)switch_channel_set_variable\(channel, \"" * | grep -v "Binary" > /tmp/chan_vars.tmp This gave me a massive list of all filenames and occurrences of "switch_channel_set_variable" with a channel variable explicitly named in quotes. I then filtered out stuff with svn in the source filename and then create a tab-separated list of chan variables and source filenames. Finally, I distilled that list with this script: perl -n -e 'm/(\w+)\t(.*)/;next if (exists($seen{$1}));print $1 . "\t" . $2 . "\n";$seen{$1}=$2;' /tmp/chan_vars3.tmp This yielded a nice list of variables. Let me know what you think about using this as the base list for channel variables on the wiki: Variable Source File: limit_realm mod/applications/mod_limit/mod_limit.c limit_id mod/applications/mod_limit/mod_limit.c limit_max mod/applications/mod_limit/mod_limit.c enum_route_count mod/applications/mod_enum/mod_enum.c enum_auto_route mod/applications/mod_enum/mod_enum.c voicemail_total_new_messages mod/applications/mod_voicemail/mod_voicemail.c voicemail_total_saved_messages mod/applications/mod_voicemail/mod_voicemail.c voicemail_urgent_new_messages mod/applications/mod_voicemail/mod_voicemail.c voicemail_urgent_saved_messages mod/applications/mod_voicemail/mod_voicemail.c voicemail_current_folder mod/applications/mod_voicemail/mod_voicemail.c voicemail_account mod/applications/mod_voicemail/mod_voicemail.c voicemail_domain mod/applications/mod_voicemail/mod_voicemail.c voicemail_caller_id_number mod/applications/mod_voicemail/mod_voicemail.c voicemail_caller_id_name mod/applications/mod_voicemail/mod_voicemail.c voicemail_file_path mod/applications/mod_voicemail/mod_voicemail.c voicemail_read_flags mod/applications/mod_voicemail/mod_voicemail.c voicemail_time mod/applications/mod_voicemail/mod_voicemail.c voicemail_priority mod/applications/mod_voicemail/mod_voicemail.c voicemail_message_len mod/applications/mod_voicemail/mod_voicemail.c voicemail_email mod/applications/mod_voicemail/mod_voicemail.c RECORD_ARTIST mod/applications/mod_voicemail/mod_voicemail.c RECORD_DATE mod/applications/mod_voicemail/mod_voicemail.c RECORD_SOFTWARE mod/applications/mod_voicemail/mod_voicemail.c RECORD_TITLE mod/applications/mod_voicemail/mod_voicemail.c RECORD_COMMENT mod/applications/mod_voicemail/mod_voicemail.c RECORD_COPYRIGHT mod/applications/mod_voicemail/mod_voicemail.c voicemail_notify_email mod/applications/mod_voicemail/mod_voicemail.c mailbox mod/applications/mod_dptools/mod_dptools.c user_name mod/applications/mod_dptools/mod_dptools.c domain_name mod/applications/mod_dptools/mod_dptools.c xfer_uuids mod/applications/mod_dptools/mod_dptools.c dialed_user mod/applications/mod_dptools/mod_dptools.c dialed_domain mod/applications/mod_dptools/mod_dptools.c sip_force_video_fmtp mod/applications/mod_fsv/mod_fsv.c fifo_priority mod/applications/mod_fifo/mod_fifo.c fifo_status mod/applications/mod_fifo/mod_fifo.c fifo_timestamp mod/applications/mod_fifo/mod_fifo.c fifo_target mod/applications/mod_fifo/mod_fifo.c dl_user mod/endpoints/mod_dingaling/mod_dingaling.c dl_host mod/endpoints/mod_dingaling/mod_dingaling.c dl_from_user mod/endpoints/mod_dingaling/mod_dingaling.c dl_from_host mod/endpoints/mod_dingaling/mod_dingaling.c sip_nat_detected mod/endpoints/mod_sofia/sofia_glue.c sofia_profile_name mod/endpoints/mod_sofia/sofia_glue.c auto_answer_destination mod/endpoints/mod_sofia/sofia.c sip_user_agent mod/endpoints/mod_sofia/sofia.c sip_rtp_txstat mod/endpoints/mod_sofia/sofia.c sip_rtp_rxstat mod/endpoints/mod_sofia/sofia.c sip_call_id mod/endpoints/mod_sofia/sofia.c sip_authorized mod/endpoints/mod_sofia/sofia.c sip_term_status mod/endpoints/mod_sofia/sofia.c sip_term_cause mod/endpoints/mod_sofia/sofia.c transfer_fallback_extension mod/endpoints/mod_sofia/sofia.c call_clientcode mod/endpoints/mod_sofia/sofia.c sofia_record_file mod/endpoints/mod_sofia/sofia.c sip_received_ip mod/endpoints/mod_sofia/sofia.c sip_received_port mod/endpoints/mod_sofia/sofia.c sip_looped_call mod/endpoints/mod_sofia/sofia.c sip_from_user_stripped mod/endpoints/mod_sofia/sofia.c sip_from_comment mod/endpoints/mod_sofia/sofia.c sofia_profile_domain_name mod/endpoints/mod_sofia/sofia.c sip_to_comment mod/endpoints/mod_sofia/sofia.c sip_referred_by_user_stripped mod/endpoints/mod_sofia/sofia.c sip_referred_by_cid mod/endpoints/mod_sofia/sofia.c sip_subject mod/endpoints/mod_sofia/sofia.c sip_via_host mod/endpoints/mod_sofia/sofia.c sip_via_port mod/endpoints/mod_sofia/sofia.c sip_via_rport mod/endpoints/mod_sofia/sofia.c sip_gateway mod/endpoints/mod_sofia/sofia.c alert_info mod/endpoints/mod_sofia/sofia.c presence_id mod/endpoints/mod_sofia/sofia.c call_timeout mod/dialplans/mod_dialplan_asterisk/mod_dialplan_asterisk.c EXTEN mod/dialplans/mod_dialplan_asterisk/mod_dialplan_asterisk.c CHANNEL mod/dialplans/mod_dialplan_asterisk/mod_dialplan_asterisk.c UNIQUEID mod/dialplans/mod_dialplan_asterisk/mod_dialplan_asterisk.c socket_path mod/event_handlers/mod_event_socket/mod_event_socket.c socket_host mod/event_handlers/mod_event_socket/mod_event_socket.c last_file_position mod/languages/mod_spidermonkey/mod_spidermonkey.c hangup_cause switch_channel.c endpoint_disposition switch_channel.c start_stamp switch_channel.c profile_start_stamp switch_channel.c answer_stamp switch_channel.c progress_stamp switch_channel.c progress_media_stamp switch_channel.c end_stamp switch_channel.c start_epoch switch_channel.c start_uepoch switch_channel.c profile_start_epoch switch_channel.c profile_start_uepoch switch_channel.c answer_epoch switch_channel.c answer_uepoch switch_channel.c end_epoch switch_channel.c end_uepoch switch_channel.c last_app switch_channel.c last_arg switch_channel.c caller_id switch_channel.c duration switch_channel.c billsec switch_channel.c progresssec switch_channel.c progress_mediasec switch_channel.c flow_billsec switch_channel.c mduration switch_channel.c billmsec switch_channel.c progressmsec switch_channel.c progress_mediamsec switch_channel.c flow_billmsec switch_channel.c uduration switch_channel.c billusec switch_channel.c progressusec switch_channel.c progress_mediausec switch_channel.c flow_billusec switch_channel.c read_codec switch_core_codec.c read_rate switch_core_codec.c write_codec switch_core_codec.c write_rate switch_core_codec.c video_read_codec switch_core_codec.c video_read_rate switch_core_codec.c video_write_codec switch_core_codec.c video_write_rate switch_core_codec.c sound_prefix switch_ivr_play_say.c 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Follow up: Are most of these read/write? Are there any that are by definition read-only? If so, could someone in the know point those out and I will get with Ray to make sure that we get the channel variables wiki page is updated. Thanks, MC ________________________________ From: freeswitch-dev-bounces@lists.freeswitch.org [mailto:freeswitch-dev-bounces@lists.freeswitch.org] On Behalf Of Brian West Sent: Friday, June 06, 2008 9:58 AM To: freeswitch-dev@lists.freeswitch.org Subject: Re: [Freeswitch-dev] INFO: Channel variables Forgot the ones in switch_types.h ;) /b On Jun 6, 2008, at 11:41 AM, Michael Collins wrote: Guys, I made a list by grepping and perling around in the source. grep -Pr "c(.)(\s+)switch_channel_set_variable\(channel, \"" * | grep -v "Binary" > /tmp/chan_vars.tmp This gave me a massive list of all filenames and occurrences of "switch_channel_set_variable" with a channel variable explicitly named in quotes. I then filtered out stuff with svn in the source filename and then create a tab-separated list of chan variables and source filenames. Finally, I distilled that list with this script: perl -n -e 'm/(\w+)\t(.*)/;next if (exists($seen{$1}));print $1 . "\t" . $2 . "\n";$seen{$1}=$2;' /tmp/chan_vars3.tmp This yielded a nice list of variables. Let me know what you think about using this as the base list for channel variables on the wiki: Variable Source File: limit_realm mod/applications/mod_limit/mod_limit.c limit_id mod/applications/mod_limit/mod_limit.c limit_max mod/applications/mod_limit/mod_limit.c enum_route_count mod/applications/mod_enum/mod_enum.c enum_auto_route mod/applications/mod_enum/mod_enum.c voicemail_total_new_messages mod/applications/mod_voicemail/mod_voicemail.c voicemail_total_saved_messages mod/applications/mod_voicemail/mod_voicemail.c voicemail_urgent_new_messages mod/applications/mod_voicemail/mod_voicemail.c voicemail_urgent_saved_messages mod/applications/mod_voicemail/mod_voicemail.c voicemail_current_folder mod/applications/mod_voicemail/mod_voicemail.c voicemail_account mod/applications/mod_voicemail/mod_voicemail.c voicemail_domain mod/applications/mod_voicemail/mod_voicemail.c voicemail_caller_id_number mod/applications/mod_voicemail/mod_voicemail.c voicemail_caller_id_name mod/applications/mod_voicemail/mod_voicemail.c voicemail_file_path mod/applications/mod_voicemail/mod_voicemail.c voicemail_read_flags mod/applications/mod_voicemail/mod_voicemail.c voicemail_time mod/applications/mod_voicemail/mod_voicemail.c voicemail_priority mod/applications/mod_voicemail/mod_voicemail.c voicemail_message_len mod/applications/mod_voicemail/mod_voicemail.c voicemail_email mod/applications/mod_voicemail/mod_voicemail.c RECORD_ARTIST mod/applications/mod_voicemail/mod_voicemail.c RECORD_DATE mod/applications/mod_voicemail/mod_voicemail.c RECORD_SOFTWARE mod/applications/mod_voicemail/mod_voicemail.c RECORD_TITLE mod/applications/mod_voicemail/mod_voicemail.c RECORD_COMMENT mod/applications/mod_voicemail/mod_voicemail.c RECORD_COPYRIGHT mod/applications/mod_voicemail/mod_voicemail.c voicemail_notify_email mod/applications/mod_voicemail/mod_voicemail.c mailbox mod/applications/mod_dptools/mod_dptools.c user_name mod/applications/mod_dptools/mod_dptools.c domain_name mod/applications/mod_dptools/mod_dptools.c xfer_uuids mod/applications/mod_dptools/mod_dptools.c dialed_user mod/applications/mod_dptools/mod_dptools.c dialed_domain mod/applications/mod_dptools/mod_dptools.c sip_force_video_fmtp mod/applications/mod_fsv/mod_fsv.c fifo_priority mod/applications/mod_fifo/mod_fifo.c fifo_status mod/applications/mod_fifo/mod_fifo.c fifo_timestamp mod/applications/mod_fifo/mod_fifo.c fifo_target mod/applications/mod_fifo/mod_fifo.c dl_user mod/endpoints/mod_dingaling/mod_dingaling.c dl_host mod/endpoints/mod_dingaling/mod_dingaling.c dl_from_user mod/endpoints/mod_dingaling/mod_dingaling.c dl_from_host mod/endpoints/mod_dingaling/mod_dingaling.c sip_nat_detected mod/endpoints/mod_sofia/sofia_glue.c sofia_profile_name mod/endpoints/mod_sofia/sofia_glue.c auto_answer_destination mod/endpoints/mod_sofia/sofia.c sip_user_agent mod/endpoints/mod_sofia/sofia.c sip_rtp_txstat mod/endpoints/mod_sofia/sofia.c sip_rtp_rxstat mod/endpoints/mod_sofia/sofia.c sip_call_id mod/endpoints/mod_sofia/sofia.c sip_authorized mod/endpoints/mod_sofia/sofia.c sip_term_status mod/endpoints/mod_sofia/sofia.c sip_term_cause mod/endpoints/mod_sofia/sofia.c transfer_fallback_extension mod/endpoints/mod_sofia/sofia.c call_clientcode mod/endpoints/mod_sofia/sofia.c sofia_record_file mod/endpoints/mod_sofia/sofia.c sip_received_ip mod/endpoints/mod_sofia/sofia.c sip_received_port mod/endpoints/mod_sofia/sofia.c sip_looped_call mod/endpoints/mod_sofia/sofia.c sip_from_user_stripped mod/endpoints/mod_sofia/sofia.c sip_from_comment mod/endpoints/mod_sofia/sofia.c sofia_profile_domain_name mod/endpoints/mod_sofia/sofia.c sip_to_comment mod/endpoints/mod_sofia/sofia.c sip_referred_by_user_stripped mod/endpoints/mod_sofia/sofia.c sip_referred_by_cid mod/endpoints/mod_sofia/sofia.c sip_subject mod/endpoints/mod_sofia/sofia.c sip_via_host mod/endpoints/mod_sofia/sofia.c sip_via_port mod/endpoints/mod_sofia/sofia.c sip_via_rport mod/endpoints/mod_sofia/sofia.c sip_gateway mod/endpoints/mod_sofia/sofia.c alert_info mod/endpoints/mod_sofia/sofia.c presence_id mod/endpoints/mod_sofia/sofia.c call_timeout mod/dialplans/mod_dialplan_asterisk/mod_dialplan_asterisk.c EXTEN mod/dialplans/mod_dialplan_asterisk/mod_dialplan_asterisk.c CHANNEL mod/dialplans/mod_dialplan_asterisk/mod_dialplan_asterisk.c UNIQUEID mod/dialplans/mod_dialplan_asterisk/mod_dialplan_asterisk.c socket_path mod/event_handlers/mod_event_socket/mod_event_socket.c socket_host mod/event_handlers/mod_event_socket/mod_event_socket.c last_file_position mod/languages/mod_spidermonkey/mod_spidermonkey.c hangup_cause switch_channel.c endpoint_disposition switch_channel.c start_stamp switch_channel.c profile_start_stamp switch_channel.c answer_stamp switch_channel.c progress_stamp switch_channel.c progress_media_stamp switch_channel.c end_stamp switch_channel.c start_epoch switch_channel.c start_uepoch switch_channel.c profile_start_epoch switch_channel.c profile_start_uepoch switch_channel.c answer_epoch switch_channel.c answer_uepoch switch_channel.c end_epoch switch_channel.c end_uepoch switch_channel.c last_app switch_channel.c last_arg switch_channel.c caller_id switch_channel.c duration switch_channel.c billsec switch_channel.c progresssec switch_channel.c progress_mediasec switch_channel.c flow_billsec switch_channel.c mduration switch_channel.c billmsec switch_channel.c progressmsec switch_channel.c progress_mediamsec switch_channel.c flow_billmsec switch_channel.c uduration switch_channel.c billusec switch_channel.c progressusec switch_channel.c progress_mediausec switch_channel.c flow_billusec switch_channel.c read_codec switch_core_codec.c read_rate switch_core_codec.c write_codec switch_core_codec.c write_rate switch_core_codec.c video_read_codec switch_core_codec.c video_read_rate switch_core_codec.c video_write_codec switch_core_codec.c video_write_rate switch_core_codec.c sound_prefix switch_ivr_play_say.c 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Good work, Michael! This is the knowledge we've accumulated regarding the channel variables: Channels Variables Types * PROTO_SPECIFIC_HANGUP_CAUSE * EXECUTE_ON_ANSWER - <Command><Args> - Execute A Command When The Called Party Answer. Example: Execute_On_Answer=Lua Incrinuse.Lua ${Uuid} * CALL_TIMEOUT - <Seconds> - Controls How Long (In Seconds) To Ring The B Leg Of A Call When Using The Bridge Application. * HOLDING_UUID * API_AFTER_BRIDGE * API_HANGUP_HOOK - <Command><Args> - Execute An Api Command On Hangup. Example: <Action Application="Set" Data="Api_Hangup_Hook=Jsrun Cleanup.Js ${Uuid}"/> * PROCESS_CDR - <False|True|A_Only|B_Only> - Indicates How To Process Cdr Records. o "false" indicates to not process the record. o "true" or undefined indicates the default behavior which is to process all CDR records. o "a_only" indicates to only process CDR records on the inbound leg of a call. o "b_only" indicates to only process CDR records on the outbound leg of a call. * BRIDGE_CHANNEL * CHANNEL_NAME * BRIDGE_UUID * PLAYBACK_TERMINATORS - <Keys> - Allows You To Set Which Dtmf Tones, If Pressed During The Playback Of A File, Will Stop It. "None" Is Valid <Keys> And Will Stand For No Termination Keys. Example: Playback_Terminators=#* * CACHE_SPEECH_HANDLES * BYPASS_MEDIA - <True|False> - 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. * PROXY_MEDIA - <True|False> - 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), However It Is Lighter On The Cpu Because Freeswitch Never Even Parses The Packets Or Processes Them In Any Way, It Simply Forwards Them Onwards. * ENDPOINT_DISPOSITION * HOLD_MUSIC - <Audio_Path> - Per-Channel Hold Music. Supports All Audio Formats And Audio Streams. * EXPORT_VARS * SWITCH_R_SDP - <Sdp String> - Changes The Remote Sdp Header * SWITCH_L_SDP - <Sdp String> - Changes The Local Sdp Header * SWITCH_M_SDP - <Sdp String> - Merged? Sdp * BRIDGE_TO - * SIGNAL_BRIDGE_TO - * SIGNAL_BOND * ORIGINATOR - * ORIGINATOR_CODEC - <Codec> - Sets The Codec For Calls Originated
From Leg A (Setting The Codec For Leg B)
* LOCAL_MEDIA_IP - * LOCAL_MEDIA_PORT - * REMOTE_MEDIA_IP - * REMOTE_MEDIA_PORT - * REMOTE_VIDEO_IP * REMOTE_VIDEO_PORT * LOCAL_VIDEO_IP * LOCAL_VIDEO_PORT * HANGUP_AFTER_BRIDGE - <True|False> - 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. * EXEC_AFTER_BRIDGE_APP * EXEC_AFTER_BRIDGE_ARG * MAX_FORWARDS - * DISABLE_APP_LOG * PASS_RFC2833 - <True|False> - Passes Rfc2833 Dtmf'S From One Side Of A Bridge To The Other Untouched. If Unset, It Decodes And Reencodes Them Before Passing Them On. * DEFAULT_LANGUAGE - <Language I.E. Fr> - 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. * CONTINUE_ON_FAIL - <True|False|Couses> - 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. * CALLER_ID_NAME - <Name> - The Caller Id Name Set By The Inbound Call, Not A Real Variable. Practically It Is Read Only. * CALLER_ID_NUMBER - <Number> - The Caller Id Phone Number Set By The Inbound Call, Not A Real Variable. Practically It Is Read Only. * EFFECTIVE_CALLER_ID_NAME - <Name> - Sets The Effective Callerid Name. Automatically Exported To Originated Channels. It Is A Var Set On A Channel In Case It Becomes The A Leg Of A Bridge - It Will Use This Callerid Name On The B Leg Outbound Call. * EFFECTIVE_CALLER_ID_NUMBER - <Number> - Sets The Effective Callerid Number. Automatically Exported To Originated Channels. It Is A Var Set On A Channel In Case It Becomes The A Leg Of A Bridge - It Will Use This Callerid Number On The B Leg Outbound Call. * ABSOLUTE_CODEC_STRING - <Codec> - Sets The Absolute Codec To Use * CODEC_STRING - <Codecs> - Sets The Codec To Add To The List Of Codecs. * RECORD_ANSWER_REQ - <True|False> - Start Recording Only When The Channel Has Been Answered. * RECORD_STEREO - <True|False> - Record Leg A And Leg B Streams Into Different Channel In A Stereo File. * TIMER_NAME - <Soft|Other> - If Set Will Make Playback And Speak Use A Timer To Clock The Audio Instead Of The Read. * FIRE_ASR_EVENTS - * SOUND_PREFIX - * _SOFIA_REPLACES_ - * ALERT_INFO - * ABSOLUTE_CODEC_STRING - <Codec> - Sets The Absolute Codec To Use * READ_CODEC - The Negotiated Codec Of The Inbound Call Leg. * WRITE_CODEC - The Negotiated Codec Of The Outbound Call Leg. * RECORD_RATE - <Rate> - Set The Sample Rate Of The Recording. Usage: Record_Rate=11025 Recording Meta Data - The following are set prior to performing a record to set the file header meta data provided the file format supports meta headers. * RECORD_TITLE - <Text> - Set Prior To Performing A Record To Set The File Header Meta Data Provided The File Format Supports Meta Headers. * RECORD_COPYRIGHT - <Text> - Set Prior To Performing A Record To Set The File Header Meta Data Provided The File Format Supports Meta Headers. * RECORD_SOFTWARE - <Text> - Set Prior To Performing A Record To Set The File Header Meta Data Provided The File Format Supports Meta Headers. * RECORD_ARTIST - <Text> - Set Prior To Performing A Record To Set The File Header Meta Data Provided The File Format Supports Meta Headers. * RECORD_COMMENT - <Text> - Set Prior To Performing A Record To Set The File Header Meta Data Provided The File Format Supports Meta Headers. * RECORD_DATE - <Text> - Set Prior To Performing A Record To Set The File Header Meta Data Provided The File Format Supports Meta Headers. Originate Variables Types * FAIL_ON_SINGLE_REJECT = <true|false> - 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. * IGNORE_EARLY_MEDIA = <true|false> - Controls if the call returns on early media or not. Default is false. * GROUP_CONFIRM_KEY = <key> - the call acceptance DTMF key. If <key> is "EXEC" it will run the application stated in group_confirm_file. i.e: group_confirm_key = 3 * GROUP_CONFIRM_FILE = <audio_file> - sound file for asking an approval DTMF. If group_confirm_key is "EXEC", this argument will have <application><args> as data and will run it before executing call. i.e. group_confirm_file = /wav/announce.wav * FORKED_DIAL * RETURN_RING_READY * ORIGINATE_RETRIES = <number> - number of retries * ORIGINATE_RETRY_SLEEP_MS = <number> - how many milisec between retries * ORIGINATION_CALLER_ID_NAME = <name> - Sets the origination callerid name. (LEG A) * ORIGINATION_CALLER_ID_NUMBER = <number> - Sets the origination callerid number. (LEG A) * RINGBACK = <audio_path> - Ring back is used to specify a sound/tone to be played for the calling user. The value is interpreted as a file if a path is given, else as a teletone. * TRANSFER_RINGBACK = <media_path> - 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 definte in ~/autoload_configs/switch.conf.xml : transfer_ringback="$${fr-ring}" * ORIGINATE_TIMEOUT = <number> - duration of timeout of originate attempt Since there's barely 5% match between our lists, I assume I'm completely off. _____ From: freeswitch-dev-bounces@lists.freeswitch.org [mailto:freeswitch-dev-bounces@lists.freeswitch.org] On Behalf Of Michael Collins Sent: Saturday, June 07, 2008 4:55 AM To: freeswitch-dev@lists.freeswitch.org Subject: Re: [Freeswitch-dev] INFO: Channel variables Follow up: Are most of these read/write? Are there any that are by definition read-only? If so, could someone in the know point those out and I will get with Ray to make sure that we get the channel variables wiki page is updated. Thanks, MC _____ From: freeswitch-dev-bounces@lists.freeswitch.org [mailto:freeswitch-dev-bounces@lists.freeswitch.org] On Behalf Of Brian West Sent: Friday, June 06, 2008 9:58 AM To: freeswitch-dev@lists.freeswitch.org Subject: Re: [Freeswitch-dev] INFO: Channel variables Forgot the ones in switch_types.h ;) /b On Jun 6, 2008, at 11:41 AM, Michael Collins wrote: Guys, I made a list by grepping and perling around in the source. grep -Pr "c(.)(\s+)switch_channel_set_variable\(channel, \"" * | grep -v "Binary" > /tmp/chan_vars.tmp This gave me a massive list of all filenames and occurrences of "switch_channel_set_variable" with a channel variable explicitly named in quotes. I then filtered out stuff with svn in the source filename and then create a tab-separated list of chan variables and source filenames. Finally, I distilled that list with this script: perl -n -e 'm/(\w+)\t(.*)/;next if (exists($seen{$1}));print $1 . "\t" . $2 . "\n";$seen{$1}=$2;' /tmp/chan_vars3.tmp This yielded a nice list of variables. Let me know what you think about using this as the base list for channel variables on the wiki: Variable Source File: limit_realm mod/applications/mod_limit/mod_limit.c limit_id mod/applications/mod_limit/mod_limit.c limit_max mod/applications/mod_limit/mod_limit.c enum_route_count mod/applications/mod_enum/mod_enum.c enum_auto_route mod/applications/mod_enum/mod_enum.c voicemail_total_new_messages mod/applications/mod_voicemail/mod_voicemail.c voicemail_total_saved_messages mod/applications/mod_voicemail/mod_voicemail.c voicemail_urgent_new_messages mod/applications/mod_voicemail/mod_voicemail.c voicemail_urgent_saved_messages mod/applications/mod_voicemail/mod_voicemail.c voicemail_current_folder mod/applications/mod_voicemail/mod_voicemail.c voicemail_account mod/applications/mod_voicemail/mod_voicemail.c voicemail_domain mod/applications/mod_voicemail/mod_voicemail.c voicemail_caller_id_number mod/applications/mod_voicemail/mod_voicemail.c voicemail_caller_id_name mod/applications/mod_voicemail/mod_voicemail.c voicemail_file_path mod/applications/mod_voicemail/mod_voicemail.c voicemail_read_flags mod/applications/mod_voicemail/mod_voicemail.c voicemail_time mod/applications/mod_voicemail/mod_voicemail.c voicemail_priority mod/applications/mod_voicemail/mod_voicemail.c voicemail_message_len mod/applications/mod_voicemail/mod_voicemail.c voicemail_email mod/applications/mod_voicemail/mod_voicemail.c RECORD_ARTIST mod/applications/mod_voicemail/mod_voicemail.c RECORD_DATE mod/applications/mod_voicemail/mod_voicemail.c RECORD_SOFTWARE mod/applications/mod_voicemail/mod_voicemail.c RECORD_TITLE mod/applications/mod_voicemail/mod_voicemail.c RECORD_COMMENT mod/applications/mod_voicemail/mod_voicemail.c RECORD_COPYRIGHT mod/applications/mod_voicemail/mod_voicemail.c voicemail_notify_email mod/applications/mod_voicemail/mod_voicemail.c mailbox mod/applications/mod_dptools/mod_dptools.c user_name mod/applications/mod_dptools/mod_dptools.c domain_name mod/applications/mod_dptools/mod_dptools.c xfer_uuids mod/applications/mod_dptools/mod_dptools.c dialed_user mod/applications/mod_dptools/mod_dptools.c dialed_domain mod/applications/mod_dptools/mod_dptools.c sip_force_video_fmtp mod/applications/mod_fsv/mod_fsv.c fifo_priority mod/applications/mod_fifo/mod_fifo.c fifo_status mod/applications/mod_fifo/mod_fifo.c fifo_timestamp mod/applications/mod_fifo/mod_fifo.c fifo_target mod/applications/mod_fifo/mod_fifo.c dl_user mod/endpoints/mod_dingaling/mod_dingaling.c dl_host mod/endpoints/mod_dingaling/mod_dingaling.c dl_from_user mod/endpoints/mod_dingaling/mod_dingaling.c dl_from_host mod/endpoints/mod_dingaling/mod_dingaling.c sip_nat_detected mod/endpoints/mod_sofia/sofia_glue.c sofia_profile_name mod/endpoints/mod_sofia/sofia_glue.c auto_answer_destination mod/endpoints/mod_sofia/sofia.c sip_user_agent mod/endpoints/mod_sofia/sofia.c sip_rtp_txstat mod/endpoints/mod_sofia/sofia.c sip_rtp_rxstat mod/endpoints/mod_sofia/sofia.c sip_call_id mod/endpoints/mod_sofia/sofia.c sip_authorized mod/endpoints/mod_sofia/sofia.c sip_term_status mod/endpoints/mod_sofia/sofia.c sip_term_cause mod/endpoints/mod_sofia/sofia.c transfer_fallback_extension mod/endpoints/mod_sofia/sofia.c call_clientcode mod/endpoints/mod_sofia/sofia.c sofia_record_file mod/endpoints/mod_sofia/sofia.c sip_received_ip mod/endpoints/mod_sofia/sofia.c sip_received_port mod/endpoints/mod_sofia/sofia.c sip_looped_call mod/endpoints/mod_sofia/sofia.c sip_from_user_stripped mod/endpoints/mod_sofia/sofia.c sip_from_comment mod/endpoints/mod_sofia/sofia.c sofia_profile_domain_name mod/endpoints/mod_sofia/sofia.c sip_to_comment mod/endpoints/mod_sofia/sofia.c sip_referred_by_user_stripped mod/endpoints/mod_sofia/sofia.c sip_referred_by_cid mod/endpoints/mod_sofia/sofia.c sip_subject mod/endpoints/mod_sofia/sofia.c sip_via_host mod/endpoints/mod_sofia/sofia.c sip_via_port mod/endpoints/mod_sofia/sofia.c sip_via_rport mod/endpoints/mod_sofia/sofia.c sip_gateway mod/endpoints/mod_sofia/sofia.c alert_info mod/endpoints/mod_sofia/sofia.c presence_id mod/endpoints/mod_sofia/sofia.c call_timeout mod/dialplans/mod_dialplan_asterisk/mod_dialplan_asterisk.c EXTEN mod/dialplans/mod_dialplan_asterisk/mod_dialplan_asterisk.c CHANNEL mod/dialplans/mod_dialplan_asterisk/mod_dialplan_asterisk.c UNIQUEID mod/dialplans/mod_dialplan_asterisk/mod_dialplan_asterisk.c socket_path mod/event_handlers/mod_event_socket/mod_event_socket.c socket_host mod/event_handlers/mod_event_socket/mod_event_socket.c last_file_position mod/languages/mod_spidermonkey/mod_spidermonkey.c hangup_cause switch_channel.c endpoint_disposition switch_channel.c start_stamp switch_channel.c profile_start_stamp switch_channel.c answer_stamp switch_channel.c progress_stamp switch_channel.c progress_media_stamp switch_channel.c end_stamp switch_channel.c start_epoch switch_channel.c start_uepoch switch_channel.c profile_start_epoch switch_channel.c profile_start_uepoch switch_channel.c answer_epoch switch_channel.c answer_uepoch switch_channel.c end_epoch switch_channel.c end_uepoch switch_channel.c last_app switch_channel.c last_arg switch_channel.c caller_id switch_channel.c duration switch_channel.c billsec switch_channel.c progresssec switch_channel.c progress_mediasec switch_channel.c flow_billsec switch_channel.c mduration switch_channel.c billmsec switch_channel.c progressmsec switch_channel.c progress_mediamsec switch_channel.c flow_billmsec switch_channel.c uduration switch_channel.c billusec switch_channel.c progressusec switch_channel.c progress_mediausec switch_channel.c flow_billusec switch_channel.c read_codec switch_core_codec.c read_rate switch_core_codec.c write_codec switch_core_codec.c write_rate switch_core_codec.c video_read_codec switch_core_codec.c video_read_rate switch_core_codec.c video_write_codec switch_core_codec.c video_write_rate switch_core_codec.c sound_prefix switch_ivr_play_say.c _______________________________________________ Freeswitch-dev mailing list Freeswitch-dev@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev http://www.freeswitch.org
________________________________ From: freeswitch-dev-bounces@lists.freeswitch.org [mailto:freeswitch-dev-bounces@lists.freeswitch.org] On Behalf Of UV Sent: Friday, June 06, 2008 12:20 PM To: freeswitch-dev@lists.freeswitch.org Subject: Re: [Freeswitch-dev] INFO: Channel variables Good work, Michael! This is the knowledge we've accumulated regarding the channel variables: Channels Variables Types * PROTO_SPECIFIC_HANGUP_CAUSE * EXECUTE_ON_ANSWER - <Command><Args> - Execute A Command When The Called Party Answer. Example: Execute_On_Answer=Lua Incrinuse.Lua ${Uuid} * CALL_TIMEOUT - <Seconds> - Controls How Long (In Seconds) To Ring The B Leg Of A Call When Using The Bridge Application. * HOLDING_UUID * API_AFTER_BRIDGE * API_HANGUP_HOOK - <Command><Args> - Execute An Api Command On Hangup. Example: <Action Application="Set" Data="Api_Hangup_Hook=Jsrun Cleanup.Js ${Uuid}"/> * PROCESS_CDR - <False|True|A_Only|B_Only> - Indicates How To Process Cdr Records. o "false" indicates to not process the record. o "true" or undefined indicates the default behavior which is to process all CDR records. o "a_only" indicates to only process CDR records on the inbound leg of a call. o "b_only" indicates to only process CDR records on the outbound leg of a call. * BRIDGE_CHANNEL * CHANNEL_NAME * BRIDGE_UUID * PLAYBACK_TERMINATORS - <Keys> - Allows You To Set Which Dtmf Tones, If Pressed During The Playback Of A File, Will Stop It. "None" Is Valid <Keys> And Will Stand For No Termination Keys. Example: Playback_Terminators=#* * CACHE_SPEECH_HANDLES * BYPASS_MEDIA - <True|False> - 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. * PROXY_MEDIA - <True|False> - 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), However It Is Lighter On The Cpu Because Freeswitch Never Even Parses The Packets Or Processes Them In Any Way, It Simply Forwards Them Onwards. * ENDPOINT_DISPOSITION * HOLD_MUSIC - <Audio_Path> - Per-Channel Hold Music. Supports All Audio Formats And Audio Streams. * EXPORT_VARS * SWITCH_R_SDP - <Sdp String> - Changes The Remote Sdp Header * SWITCH_L_SDP - <Sdp String> - Changes The Local Sdp Header * SWITCH_M_SDP - <Sdp String> - Merged? Sdp * BRIDGE_TO - * SIGNAL_BRIDGE_TO - * SIGNAL_BOND * ORIGINATOR - * ORIGINATOR_CODEC - <Codec> - Sets The Codec For Calls Originated
From Leg A (Setting The Codec For Leg B)
* LOCAL_MEDIA_IP - * LOCAL_MEDIA_PORT - * REMOTE_MEDIA_IP - * REMOTE_MEDIA_PORT - * REMOTE_VIDEO_IP * REMOTE_VIDEO_PORT * LOCAL_VIDEO_IP * LOCAL_VIDEO_PORT * HANGUP_AFTER_BRIDGE - <True|False> - 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. * EXEC_AFTER_BRIDGE_APP * EXEC_AFTER_BRIDGE_ARG * MAX_FORWARDS - * DISABLE_APP_LOG * PASS_RFC2833 - <True|False> - Passes Rfc2833 Dtmf'S From One Side Of A Bridge To The Other Untouched. If Unset, It Decodes And Reencodes Them Before Passing Them On. * DEFAULT_LANGUAGE - <Language I.E. Fr> - 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. * CONTINUE_ON_FAIL - <True|False|Couses> - 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. * CALLER_ID_NAME - <Name> - The Caller Id Name Set By The Inbound Call, Not A Real Variable. Practically It Is Read Only. * CALLER_ID_NUMBER - <Number> - The Caller Id Phone Number Set By The Inbound Call, Not A Real Variable. Practically It Is Read Only. * EFFECTIVE_CALLER_ID_NAME - <Name> - Sets The Effective Callerid Name. Automatically Exported To Originated Channels. It Is A Var Set On A Channel In Case It Becomes The A Leg Of A Bridge - It Will Use This Callerid Name On The B Leg Outbound Call. * EFFECTIVE_CALLER_ID_NUMBER - <Number> - Sets The Effective Callerid Number. Automatically Exported To Originated Channels. It Is A Var Set On A Channel In Case It Becomes The A Leg Of A Bridge - It Will Use This Callerid Number On The B Leg Outbound Call. * ABSOLUTE_CODEC_STRING - <Codec> - Sets The Absolute Codec To Use * CODEC_STRING - <Codecs> - Sets The Codec To Add To The List Of Codecs. * RECORD_ANSWER_REQ - <True|False> - Start Recording Only When The Channel Has Been Answered. * RECORD_STEREO - <True|False> - Record Leg A And Leg B Streams Into Different Channel In A Stereo File. * TIMER_NAME - <Soft|Other> - If Set Will Make Playback And Speak Use A Timer To Clock The Audio Instead Of The Read. * FIRE_ASR_EVENTS - * SOUND_PREFIX - * _SOFIA_REPLACES_ - * ALERT_INFO - * ABSOLUTE_CODEC_STRING - <Codec> - Sets The Absolute Codec To Use * READ_CODEC - The Negotiated Codec Of The Inbound Call Leg. * WRITE_CODEC - The Negotiated Codec Of The Outbound Call Leg. * RECORD_RATE - <Rate> - Set The Sample Rate Of The Recording. Usage: Record_Rate=11025 Recording Meta Data - The following are set prior to performing a record to set the file header meta data provided the file format supports meta headers. * RECORD_TITLE - <Text> - Set Prior To Performing A Record To Set The File Header Meta Data Provided The File Format Supports Meta Headers. * RECORD_COPYRIGHT - <Text> - Set Prior To Performing A Record To Set The File Header Meta Data Provided The File Format Supports Meta Headers. * RECORD_SOFTWARE - <Text> - Set Prior To Performing A Record To Set The File Header Meta Data Provided The File Format Supports Meta Headers. * RECORD_ARTIST - <Text> - Set Prior To Performing A Record To Set The File Header Meta Data Provided The File Format Supports Meta Headers. * RECORD_COMMENT - <Text> - Set Prior To Performing A Record To Set The File Header Meta Data Provided The File Format Supports Meta Headers. * RECORD_DATE - <Text> - Set Prior To Performing A Record To Set The File Header Meta Data Provided The File Format Supports Meta Headers. Originate Variables Types * FAIL_ON_SINGLE_REJECT = <true|false> - 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. * IGNORE_EARLY_MEDIA = <true|false> - Controls if the call returns on early media or not. Default is false. * GROUP_CONFIRM_KEY = <key> - the call acceptance DTMF key. If <key> is "EXEC" it will run the application stated in group_confirm_file. i.e: group_confirm_key = 3 * GROUP_CONFIRM_FILE = <audio_file> - sound file for asking an approval DTMF. If group_confirm_key is "EXEC", this argument will have <application><args> as data and will run it before executing call. i.e. group_confirm_file = /wav/announce.wav * FORKED_DIAL * RETURN_RING_READY * ORIGINATE_RETRIES = <number> - number of retries * ORIGINATE_RETRY_SLEEP_MS = <number> - how many milisec between retries * ORIGINATION_CALLER_ID_NAME = <name> - Sets the origination callerid name. (LEG A) * ORIGINATION_CALLER_ID_NUMBER = <number> - Sets the origination callerid number. (LEG A) * RINGBACK = <audio_path> - Ring back is used to specify a sound/tone to be played for the calling user. The value is interpreted as a file if a path is given, else as a teletone. * TRANSFER_RINGBACK = <media_path> - 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 definte in ~/autoload_configs/switch.conf.xml : transfer_ringback="$${fr-ring}" * ORIGINATE_TIMEOUT = <number> - duration of timeout of originate attempt Since there's barely 5% match between our lists, I assume I'm completely off... Well, I took the variable names right out of the source code which is mostly lower case so that might account for a lot of the differences. However, I will use your list as a reference just to make sure that I've not missing anything... -MC ________________________________ From: freeswitch-dev-bounces@lists.freeswitch.org [mailto:freeswitch-dev-bounces@lists.freeswitch.org] On Behalf Of Michael Collins Sent: Saturday, June 07, 2008 4:55 AM To: freeswitch-dev@lists.freeswitch.org Subject: Re: [Freeswitch-dev] INFO: Channel variables Follow up: Are most of these read/write? Are there any that are by definition read-only? If so, could someone in the know point those out and I will get with Ray to make sure that we get the channel variables wiki page is updated. Thanks, MC ________________________________ From: freeswitch-dev-bounces@lists.freeswitch.org [mailto:freeswitch-dev-bounces@lists.freeswitch.org] On Behalf Of Brian West Sent: Friday, June 06, 2008 9:58 AM To: freeswitch-dev@lists.freeswitch.org Subject: Re: [Freeswitch-dev] INFO: Channel variables Forgot the ones in switch_types.h ;) /b On Jun 6, 2008, at 11:41 AM, Michael Collins wrote: Guys, I made a list by grepping and perling around in the source. grep -Pr "c(.)(\s+)switch_channel_set_variable\(channel, \"" * | grep -v "Binary" > /tmp/chan_vars.tmp This gave me a massive list of all filenames and occurrences of "switch_channel_set_variable" with a channel variable explicitly named in quotes. I then filtered out stuff with svn in the source filename and then create a tab-separated list of chan variables and source filenames. Finally, I distilled that list with this script: perl -n -e 'm/(\w+)\t(.*)/;next if (exists($seen{$1}));print $1 . "\t" . $2 . "\n";$seen{$1}=$2;' /tmp/chan_vars3.tmp This yielded a nice list of variables. Let me know what you think about using this as the base list for channel variables on the wiki: Variable Source File: limit_realm mod/applications/mod_limit/mod_limit.c limit_id mod/applications/mod_limit/mod_limit.c limit_max mod/applications/mod_limit/mod_limit.c enum_route_count mod/applications/mod_enum/mod_enum.c enum_auto_route mod/applications/mod_enum/mod_enum.c voicemail_total_new_messages mod/applications/mod_voicemail/mod_voicemail.c voicemail_total_saved_messages mod/applications/mod_voicemail/mod_voicemail.c voicemail_urgent_new_messages mod/applications/mod_voicemail/mod_voicemail.c voicemail_urgent_saved_messages mod/applications/mod_voicemail/mod_voicemail.c voicemail_current_folder mod/applications/mod_voicemail/mod_voicemail.c voicemail_account mod/applications/mod_voicemail/mod_voicemail.c voicemail_domain mod/applications/mod_voicemail/mod_voicemail.c voicemail_caller_id_number mod/applications/mod_voicemail/mod_voicemail.c voicemail_caller_id_name mod/applications/mod_voicemail/mod_voicemail.c voicemail_file_path mod/applications/mod_voicemail/mod_voicemail.c voicemail_read_flags mod/applications/mod_voicemail/mod_voicemail.c voicemail_time mod/applications/mod_voicemail/mod_voicemail.c voicemail_priority mod/applications/mod_voicemail/mod_voicemail.c voicemail_message_len mod/applications/mod_voicemail/mod_voicemail.c voicemail_email mod/applications/mod_voicemail/mod_voicemail.c RECORD_ARTIST mod/applications/mod_voicemail/mod_voicemail.c RECORD_DATE mod/applications/mod_voicemail/mod_voicemail.c RECORD_SOFTWARE mod/applications/mod_voicemail/mod_voicemail.c RECORD_TITLE mod/applications/mod_voicemail/mod_voicemail.c RECORD_COMMENT mod/applications/mod_voicemail/mod_voicemail.c RECORD_COPYRIGHT mod/applications/mod_voicemail/mod_voicemail.c voicemail_notify_email mod/applications/mod_voicemail/mod_voicemail.c mailbox mod/applications/mod_dptools/mod_dptools.c user_name mod/applications/mod_dptools/mod_dptools.c domain_name mod/applications/mod_dptools/mod_dptools.c xfer_uuids mod/applications/mod_dptools/mod_dptools.c dialed_user mod/applications/mod_dptools/mod_dptools.c dialed_domain mod/applications/mod_dptools/mod_dptools.c sip_force_video_fmtp mod/applications/mod_fsv/mod_fsv.c fifo_priority mod/applications/mod_fifo/mod_fifo.c fifo_status mod/applications/mod_fifo/mod_fifo.c fifo_timestamp mod/applications/mod_fifo/mod_fifo.c fifo_target mod/applications/mod_fifo/mod_fifo.c dl_user mod/endpoints/mod_dingaling/mod_dingaling.c dl_host mod/endpoints/mod_dingaling/mod_dingaling.c dl_from_user mod/endpoints/mod_dingaling/mod_dingaling.c dl_from_host mod/endpoints/mod_dingaling/mod_dingaling.c sip_nat_detected mod/endpoints/mod_sofia/sofia_glue.c sofia_profile_name mod/endpoints/mod_sofia/sofia_glue.c auto_answer_destination mod/endpoints/mod_sofia/sofia.c sip_user_agent mod/endpoints/mod_sofia/sofia.c sip_rtp_txstat mod/endpoints/mod_sofia/sofia.c sip_rtp_rxstat mod/endpoints/mod_sofia/sofia.c sip_call_id mod/endpoints/mod_sofia/sofia.c sip_authorized mod/endpoints/mod_sofia/sofia.c sip_term_status mod/endpoints/mod_sofia/sofia.c sip_term_cause mod/endpoints/mod_sofia/sofia.c transfer_fallback_extension mod/endpoints/mod_sofia/sofia.c call_clientcode mod/endpoints/mod_sofia/sofia.c sofia_record_file mod/endpoints/mod_sofia/sofia.c sip_received_ip mod/endpoints/mod_sofia/sofia.c sip_received_port mod/endpoints/mod_sofia/sofia.c sip_looped_call mod/endpoints/mod_sofia/sofia.c sip_from_user_stripped mod/endpoints/mod_sofia/sofia.c sip_from_comment mod/endpoints/mod_sofia/sofia.c sofia_profile_domain_name mod/endpoints/mod_sofia/sofia.c sip_to_comment mod/endpoints/mod_sofia/sofia.c sip_referred_by_user_stripped mod/endpoints/mod_sofia/sofia.c sip_referred_by_cid mod/endpoints/mod_sofia/sofia.c sip_subject mod/endpoints/mod_sofia/sofia.c sip_via_host mod/endpoints/mod_sofia/sofia.c sip_via_port mod/endpoints/mod_sofia/sofia.c sip_via_rport mod/endpoints/mod_sofia/sofia.c sip_gateway mod/endpoints/mod_sofia/sofia.c alert_info mod/endpoints/mod_sofia/sofia.c presence_id mod/endpoints/mod_sofia/sofia.c call_timeout mod/dialplans/mod_dialplan_asterisk/mod_dialplan_asterisk.c EXTEN mod/dialplans/mod_dialplan_asterisk/mod_dialplan_asterisk.c CHANNEL mod/dialplans/mod_dialplan_asterisk/mod_dialplan_asterisk.c UNIQUEID mod/dialplans/mod_dialplan_asterisk/mod_dialplan_asterisk.c socket_path mod/event_handlers/mod_event_socket/mod_event_socket.c socket_host mod/event_handlers/mod_event_socket/mod_event_socket.c last_file_position mod/languages/mod_spidermonkey/mod_spidermonkey.c hangup_cause switch_channel.c endpoint_disposition switch_channel.c start_stamp switch_channel.c profile_start_stamp switch_channel.c answer_stamp switch_channel.c progress_stamp switch_channel.c progress_media_stamp switch_channel.c end_stamp switch_channel.c start_epoch switch_channel.c start_uepoch switch_channel.c profile_start_epoch switch_channel.c profile_start_uepoch switch_channel.c answer_epoch switch_channel.c answer_uepoch switch_channel.c end_epoch switch_channel.c end_uepoch switch_channel.c last_app switch_channel.c last_arg switch_channel.c caller_id switch_channel.c duration switch_channel.c billsec switch_channel.c progresssec switch_channel.c progress_mediasec switch_channel.c flow_billsec switch_channel.c mduration switch_channel.c billmsec switch_channel.c progressmsec switch_channel.c progress_mediamsec switch_channel.c flow_billmsec switch_channel.c uduration switch_channel.c billusec switch_channel.c progressusec switch_channel.c progress_mediausec switch_channel.c flow_billusec switch_channel.c read_codec switch_core_codec.c read_rate switch_core_codec.c write_codec switch_core_codec.c write_rate switch_core_codec.c video_read_codec switch_core_codec.c video_read_rate switch_core_codec.c video_write_codec switch_core_codec.c video_write_rate switch_core_codec.c sound_prefix switch_ivr_play_say.c _______________________________________________ Freeswitch-dev mailing list Freeswitch-dev@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev http://www.freeswitch.org
Disregard the fact it's all upper-case. It's only for clearer presentation in the documentation. I'll look forward seeing your finished work. Let me know if I can be of any assistance. _____ From: freeswitch-dev-bounces@lists.freeswitch.org [mailto:freeswitch-dev-bounces@lists.freeswitch.org] On Behalf Of Michael Collins Sent: Saturday, June 07, 2008 6:35 AM To: freeswitch-dev@lists.freeswitch.org Subject: Re: [Freeswitch-dev] INFO: Channel variables _____ From: freeswitch-dev-bounces@lists.freeswitch.org [mailto:freeswitch-dev-bounces@lists.freeswitch.org] On Behalf Of UV Sent: Friday, June 06, 2008 12:20 PM To: freeswitch-dev@lists.freeswitch.org Subject: Re: [Freeswitch-dev] INFO: Channel variables Good work, Michael! This is the knowledge we've accumulated regarding the channel variables: Channels Variables Types * PROTO_SPECIFIC_HANGUP_CAUSE * EXECUTE_ON_ANSWER - <Command><Args> - Execute A Command When The Called Party Answer. Example: Execute_On_Answer=Lua Incrinuse.Lua ${Uuid} * CALL_TIMEOUT - <Seconds> - Controls How Long (In Seconds) To Ring The B Leg Of A Call When Using The Bridge Application. * HOLDING_UUID * API_AFTER_BRIDGE * API_HANGUP_HOOK - <Command><Args> - Execute An Api Command On Hangup. Example: <Action Application="Set" Data="Api_Hangup_Hook=Jsrun Cleanup.Js ${Uuid}"/> * PROCESS_CDR - <False|True|A_Only|B_Only> - Indicates How To Process Cdr Records. o "false" indicates to not process the record. o "true" or undefined indicates the default behavior which is to process all CDR records. o "a_only" indicates to only process CDR records on the inbound leg of a call. o "b_only" indicates to only process CDR records on the outbound leg of a call. * BRIDGE_CHANNEL * CHANNEL_NAME * BRIDGE_UUID * PLAYBACK_TERMINATORS - <Keys> - Allows You To Set Which Dtmf Tones, If Pressed During The Playback Of A File, Will Stop It. "None" Is Valid <Keys> And Will Stand For No Termination Keys. Example: Playback_Terminators=#* * CACHE_SPEECH_HANDLES * BYPASS_MEDIA - <True|False> - 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. * PROXY_MEDIA - <True|False> - 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), However It Is Lighter On The Cpu Because Freeswitch Never Even Parses The Packets Or Processes Them In Any Way, It Simply Forwards Them Onwards. * ENDPOINT_DISPOSITION * HOLD_MUSIC - <Audio_Path> - Per-Channel Hold Music. Supports All Audio Formats And Audio Streams. * EXPORT_VARS * SWITCH_R_SDP - <Sdp String> - Changes The Remote Sdp Header * SWITCH_L_SDP - <Sdp String> - Changes The Local Sdp Header * SWITCH_M_SDP - <Sdp String> - Merged? Sdp * BRIDGE_TO - * SIGNAL_BRIDGE_TO - * SIGNAL_BOND * ORIGINATOR - * ORIGINATOR_CODEC - <Codec> - Sets The Codec For Calls Originated From Leg A (Setting The Codec For Leg B) * LOCAL_MEDIA_IP - * LOCAL_MEDIA_PORT - * REMOTE_MEDIA_IP - * REMOTE_MEDIA_PORT - * REMOTE_VIDEO_IP * REMOTE_VIDEO_PORT * LOCAL_VIDEO_IP * LOCAL_VIDEO_PORT * HANGUP_AFTER_BRIDGE - <True|False> - 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. * EXEC_AFTER_BRIDGE_APP * EXEC_AFTER_BRIDGE_ARG * MAX_FORWARDS - * DISABLE_APP_LOG * PASS_RFC2833 - <True|False> - Passes Rfc2833 Dtmf'S From One Side Of A Bridge To The Other Untouched. If Unset, It Decodes And Reencodes Them Before Passing Them On. * DEFAULT_LANGUAGE - <Language I.E. Fr> - 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. * CONTINUE_ON_FAIL - <True|False|Couses> - 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. * CALLER_ID_NAME - <Name> - The Caller Id Name Set By The Inbound Call, Not A Real Variable. Practically It Is Read Only. * CALLER_ID_NUMBER - <Number> - The Caller Id Phone Number Set By The Inbound Call, Not A Real Variable. Practically It Is Read Only. * EFFECTIVE_CALLER_ID_NAME - <Name> - Sets The Effective Callerid Name. Automatically Exported To Originated Channels. It Is A Var Set On A Channel In Case It Becomes The A Leg Of A Bridge - It Will Use This Callerid Name On The B Leg Outbound Call. * EFFECTIVE_CALLER_ID_NUMBER - <Number> - Sets The Effective Callerid Number. Automatically Exported To Originated Channels. It Is A Var Set On A Channel In Case It Becomes The A Leg Of A Bridge - It Will Use This Callerid Number On The B Leg Outbound Call. * ABSOLUTE_CODEC_STRING - <Codec> - Sets The Absolute Codec To Use * CODEC_STRING - <Codecs> - Sets The Codec To Add To The List Of Codecs. * RECORD_ANSWER_REQ - <True|False> - Start Recording Only When The Channel Has Been Answered. * RECORD_STEREO - <True|False> - Record Leg A And Leg B Streams Into Different Channel In A Stereo File. * TIMER_NAME - <Soft|Other> - If Set Will Make Playback And Speak Use A Timer To Clock The Audio Instead Of The Read. * FIRE_ASR_EVENTS - * SOUND_PREFIX - * _SOFIA_REPLACES_ - * ALERT_INFO - * ABSOLUTE_CODEC_STRING - <Codec> - Sets The Absolute Codec To Use * READ_CODEC - The Negotiated Codec Of The Inbound Call Leg. * WRITE_CODEC - The Negotiated Codec Of The Outbound Call Leg. * RECORD_RATE - <Rate> - Set The Sample Rate Of The Recording. Usage: Record_Rate=11025 Recording Meta Data - The following are set prior to performing a record to set the file header meta data provided the file format supports meta headers. * RECORD_TITLE - <Text> - Set Prior To Performing A Record To Set The File Header Meta Data Provided The File Format Supports Meta Headers. * RECORD_COPYRIGHT - <Text> - Set Prior To Performing A Record To Set The File Header Meta Data Provided The File Format Supports Meta Headers. * RECORD_SOFTWARE - <Text> - Set Prior To Performing A Record To Set The File Header Meta Data Provided The File Format Supports Meta Headers. * RECORD_ARTIST - <Text> - Set Prior To Performing A Record To Set The File Header Meta Data Provided The File Format Supports Meta Headers. * RECORD_COMMENT - <Text> - Set Prior To Performing A Record To Set The File Header Meta Data Provided The File Format Supports Meta Headers. * RECORD_DATE - <Text> - Set Prior To Performing A Record To Set The File Header Meta Data Provided The File Format Supports Meta Headers. Originate Variables Types * FAIL_ON_SINGLE_REJECT = <true|false> - 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. * IGNORE_EARLY_MEDIA = <true|false> - Controls if the call returns on early media or not. Default is false. * GROUP_CONFIRM_KEY = <key> - the call acceptance DTMF key. If <key> is "EXEC" it will run the application stated in group_confirm_file. i.e: group_confirm_key = 3 * GROUP_CONFIRM_FILE = <audio_file> - sound file for asking an approval DTMF. If group_confirm_key is "EXEC", this argument will have <application><args> as data and will run it before executing call. i.e. group_confirm_file = /wav/announce.wav * FORKED_DIAL * RETURN_RING_READY * ORIGINATE_RETRIES = <number> - number of retries * ORIGINATE_RETRY_SLEEP_MS = <number> - how many milisec between retries * ORIGINATION_CALLER_ID_NAME = <name> - Sets the origination callerid name. (LEG A) * ORIGINATION_CALLER_ID_NUMBER = <number> - Sets the origination callerid number. (LEG A) * RINGBACK = <audio_path> - Ring back is used to specify a sound/tone to be played for the calling user. The value is interpreted as a file if a path is given, else as a teletone. * TRANSFER_RINGBACK = <media_path> - 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 definte in ~/autoload_configs/switch.conf.xml : transfer_ringback="$${fr-ring}" * ORIGINATE_TIMEOUT = <number> - duration of timeout of originate attempt Since there's barely 5% match between our lists, I assume I'm completely off. Well, I took the variable names right out of the source code which is mostly lower case so that might account for a lot of the differences. However, I will use your list as a reference just to make sure that I've not missing anything. -MC
right now the variable names are not case sensitive anyway. It's probably too late to change that so.... =D On Sat, Jun 7, 2008 at 6:30 AM, UV <uv@talknet.com.au> wrote:
Disregard the fact it's all upper-case. It's only for clearer presentation in the documentation.
I'll look forward seeing your finished work. Let me know if I can be of any assistance.
------------------------------
*From:* freeswitch-dev-bounces@lists.freeswitch.org [mailto: freeswitch-dev-bounces@lists.freeswitch.org] *On Behalf Of *Michael Collins *Sent:* Saturday, June 07, 2008 6:35 AM *To:* freeswitch-dev@lists.freeswitch.org *Subject:* Re: [Freeswitch-dev] INFO: Channel variables
------------------------------
*From:* freeswitch-dev-bounces@lists.freeswitch.org [mailto: freeswitch-dev-bounces@lists.freeswitch.org] *On Behalf Of *UV *Sent:* Friday, June 06, 2008 12:20 PM *To:* freeswitch-dev@lists.freeswitch.org *Subject:* Re: [Freeswitch-dev] INFO: Channel variables
Good work, Michael!
This is the knowledge we've accumulated regarding the channel variables: *Channels Variables Types*
· PROTO_SPECIFIC_HANGUP_CAUSE
· EXECUTE_ON_ANSWER - <Command><Args> - Execute A Command When The Called Party Answer. Example: Execute_On_Answer=Lua Incrinuse.Lua ${Uuid}
· CALL_TIMEOUT - <Seconds> - Controls How Long (In Seconds) To Ring The B Leg Of A Call When Using The Bridge Application.
· HOLDING_UUID
· API_AFTER_BRIDGE
· API_HANGUP_HOOK - <Command><Args> - Execute An Api Command On Hangup. Example: <Action Application="Set" Data="Api_Hangup_Hook=Jsrun Cleanup.Js ${Uuid}"/>
· PROCESS_CDR - <False|True|A_Only|B_Only> - Indicates How To Process Cdr Records.
o "false" indicates to not process the record.
o "true" or undefined indicates the default behavior which is to process all CDR records.
o "a_only" indicates to only process CDR records on the inbound leg of a call.
o "b_only" indicates to only process CDR records on the outbound leg of a call.
· BRIDGE_CHANNEL
· CHANNEL_NAME
· BRIDGE_UUID
· PLAYBACK_TERMINATORS - <Keys> - Allows You To Set Which Dtmf Tones, If Pressed During The Playback Of A File, Will Stop It. "None" Is Valid <Keys> And Will Stand For No Termination Keys. Example: Playback_Terminators=#*
· CACHE_SPEECH_HANDLES
· BYPASS_MEDIA - <True|False> - 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.
· PROXY_MEDIA - <True|False> - 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), However It Is Lighter On The Cpu Because Freeswitch Never Even Parses The Packets Or Processes Them In Any Way, It Simply Forwards Them Onwards.
· ENDPOINT_DISPOSITION
· HOLD_MUSIC - <Audio_Path> - Per-Channel Hold Music. Supports All Audio Formats And Audio Streams.
· EXPORT_VARS
· SWITCH_R_SDP - <Sdp String> - Changes The Remote Sdp Header
· SWITCH_L_SDP - <Sdp String> - Changes The Local Sdp Header
· SWITCH_M_SDP - <Sdp String> - Merged? Sdp
· BRIDGE_TO -
· SIGNAL_BRIDGE_TO -
· SIGNAL_BOND
· ORIGINATOR -
· ORIGINATOR_CODEC - <Codec> - Sets The Codec For Calls Originated From Leg A (Setting The Codec For Leg B)
· LOCAL_MEDIA_IP -
· LOCAL_MEDIA_PORT -
· REMOTE_MEDIA_IP -
· REMOTE_MEDIA_PORT -
· REMOTE_VIDEO_IP
· REMOTE_VIDEO_PORT
· LOCAL_VIDEO_IP
· LOCAL_VIDEO_PORT
· HANGUP_AFTER_BRIDGE - <True|False> - 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.
· EXEC_AFTER_BRIDGE_APP
· EXEC_AFTER_BRIDGE_ARG
· MAX_FORWARDS -
· DISABLE_APP_LOG
· PASS_RFC2833 - <True|False> - Passes Rfc2833 Dtmf'S From One Side Of A Bridge To The Other Untouched. If Unset, It Decodes And Reencodes Them Before Passing Them On.
· DEFAULT_LANGUAGE - <Language I.E. Fr> - 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.
· CONTINUE_ON_FAIL - <True|False|Couses> - 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.
· CALLER_ID_NAME - <Name> - The Caller Id Name Set By The Inbound Call, Not A Real Variable. Practically It Is Read Only.
· CALLER_ID_NUMBER - <Number> - The Caller Id Phone Number Set By The Inbound Call, Not A Real Variable. Practically It Is Read Only.
· EFFECTIVE_CALLER_ID_NAME - <Name> - Sets The Effective Callerid Name. Automatically Exported To Originated Channels. It Is A Var Set On A Channel In Case It Becomes The A Leg Of A Bridge - It Will Use This Callerid Name On The B Leg Outbound Call.
· EFFECTIVE_CALLER_ID_NUMBER - <Number> - Sets The Effective Callerid Number. Automatically Exported To Originated Channels. It Is A Var Set On A Channel In Case It Becomes The A Leg Of A Bridge - It Will Use This Callerid Number On The B Leg Outbound Call.
· ABSOLUTE_CODEC_STRING - <Codec> - Sets The Absolute Codec To Use
· CODEC_STRING - <Codecs> - Sets The Codec To Add To The List Of Codecs.
· RECORD_ANSWER_REQ - <True|False> - Start Recording Only When The Channel Has Been Answered.
· RECORD_STEREO - <True|False> - Record Leg A And Leg B Streams Into Different Channel In A Stereo File.
· TIMER_NAME - <Soft|Other> - If Set Will Make Playback And Speak Use A Timer To Clock The Audio Instead Of The Read.
· FIRE_ASR_EVENTS -
· SOUND_PREFIX -
· _SOFIA_REPLACES_ -
· ALERT_INFO -
· ABSOLUTE_CODEC_STRING - <Codec> - Sets The Absolute Codec To Use
· READ_CODEC - The Negotiated Codec Of The Inbound Call Leg.
· WRITE_CODEC - The Negotiated Codec Of The Outbound Call Leg.
· RECORD_RATE - <Rate> - Set The Sample Rate Of The Recording. Usage: Record_Rate=11025
Recording Meta Data - The following are set prior to performing a record to set the file header meta data provided the file format supports meta headers.
· RECORD_TITLE - <Text> - Set Prior To Performing A Record To Set The File Header Meta Data Provided The File Format Supports Meta Headers.
· RECORD_COPYRIGHT - <Text> - Set Prior To Performing A Record To Set The File Header Meta Data Provided The File Format Supports Meta Headers.
· RECORD_SOFTWARE - <Text> - Set Prior To Performing A Record To Set The File Header Meta Data Provided The File Format Supports Meta Headers.
· RECORD_ARTIST - <Text> - Set Prior To Performing A Record To Set The File Header Meta Data Provided The File Format Supports Meta Headers.
· RECORD_COMMENT - <Text> - Set Prior To Performing A Record To Set The File Header Meta Data Provided The File Format Supports Meta Headers.
· RECORD_DATE - <Text> - Set Prior To Performing A Record To Set The File Header Meta Data Provided The File Format Supports Meta Headers. *Originate Variables Types*
· FAIL_ON_SINGLE_REJECT = <true|false> - 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.
· IGNORE_EARLY_MEDIA = <true|false> - Controls if the call returns on early media or not. Default is false.
· GROUP_CONFIRM_KEY = <key> - the call acceptance DTMF key. If <key> is "EXEC" it will run the application stated in group_confirm_file. i.e: group_confirm_key = 3
· GROUP_CONFIRM_FILE = <audio_file> - sound file for asking an approval DTMF. If group_confirm_key is "EXEC", this argument will have <application><args> as data and will run it before executing call. i.e. group_confirm_file = /wav/announce.wav
· FORKED_DIAL
· RETURN_RING_READY
· ORIGINATE_RETRIES = <number> - number of retries
· ORIGINATE_RETRY_SLEEP_MS = <number> - how many milisec between retries
· ORIGINATION_CALLER_ID_NAME = <name> - Sets the origination callerid name. (LEG A)
· ORIGINATION_CALLER_ID_NUMBER = <number> - Sets the origination callerid number. (LEG A)
· RINGBACK = <audio_path> - Ring back is used to specify a sound/tone to be played for the calling user. The value is interpreted as a file if a path is given, else as a teletone.
· TRANSFER_RINGBACK = <media_path> - 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 definte in ~/autoload_configs/switch.conf.xml : transfer_ringback="$${fr-ring}"
· ORIGINATE_TIMEOUT = <number> - duration of timeout of originate attempt
Since there's barely 5% match between our lists, I assume I'm completely off…
*Well, I took the variable names right out of the source code which is mostly lower case so that might account for a lot of the differences. However, I will use your list as a reference just to make sure that I've not missing anything…*
* *
*-MC*
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-- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale@hotmail.com <MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com<PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <sip%3A888@conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org<googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:213-799-1400
Both of your comments are duly noted we will make sure to doc the case insensitivity we will also document the latest variable you just added regarding SIP call ID -MC ________________________________ From: freeswitch-dev-bounces@lists.freeswitch.org [mailto:freeswitch-dev-bounces@lists.freeswitch.org] On Behalf Of Anthony Minessale Sent: Saturday, June 07, 2008 8:37 AM To: freeswitch-dev@lists.freeswitch.org Subject: Re: [Freeswitch-dev] INFO: Channel variables right now the variable names are not case sensitive anyway. It's probably too late to change that so.... =D On Sat, Jun 7, 2008 at 6:30 AM, UV <uv@talknet.com.au> wrote: Disregard the fact it's all upper-case. It's only for clearer presentation in the documentation. I'll look forward seeing your finished work. Let me know if I can be of any assistance. ________________________________ From: freeswitch-dev-bounces@lists.freeswitch.org [mailto:freeswitch-dev-bounces@lists.freeswitch.org] On Behalf Of Michael Collins Sent: Saturday, June 07, 2008 6:35 AM To: freeswitch-dev@lists.freeswitch.org Subject: Re: [Freeswitch-dev] INFO: Channel variables ________________________________ From: freeswitch-dev-bounces@lists.freeswitch.org [mailto:freeswitch-dev-bounces@lists.freeswitch.org] On Behalf Of UV Sent: Friday, June 06, 2008 12:20 PM To: freeswitch-dev@lists.freeswitch.org Subject: Re: [Freeswitch-dev] INFO: Channel variables Good work, Michael! This is the knowledge we've accumulated regarding the channel variables: Channels Variables Types * PROTO_SPECIFIC_HANGUP_CAUSE * EXECUTE_ON_ANSWER - <Command><Args> - Execute A Command When The Called Party Answer. Example: Execute_On_Answer=Lua Incrinuse.Lua ${Uuid} * CALL_TIMEOUT - <Seconds> - Controls How Long (In Seconds) To Ring The B Leg Of A Call When Using The Bridge Application. * HOLDING_UUID * API_AFTER_BRIDGE * API_HANGUP_HOOK - <Command><Args> - Execute An Api Command On Hangup. Example: <Action Application="Set" Data="Api_Hangup_Hook=Jsrun Cleanup.Js ${Uuid}"/> * PROCESS_CDR - <False|True|A_Only|B_Only> - Indicates How To Process Cdr Records. o "false" indicates to not process the record. o "true" or undefined indicates the default behavior which is to process all CDR records. o "a_only" indicates to only process CDR records on the inbound leg of a call. o "b_only" indicates to only process CDR records on the outbound leg of a call. * BRIDGE_CHANNEL * CHANNEL_NAME * BRIDGE_UUID * PLAYBACK_TERMINATORS - <Keys> - Allows You To Set Which Dtmf Tones, If Pressed During The Playback Of A File, Will Stop It. "None" Is Valid <Keys> And Will Stand For No Termination Keys. Example: Playback_Terminators=#* * CACHE_SPEECH_HANDLES * BYPASS_MEDIA - <True|False> - 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. * PROXY_MEDIA - <True|False> - 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), However It Is Lighter On The Cpu Because Freeswitch Never Even Parses The Packets Or Processes Them In Any Way, It Simply Forwards Them Onwards. * ENDPOINT_DISPOSITION * HOLD_MUSIC - <Audio_Path> - Per-Channel Hold Music. Supports All Audio Formats And Audio Streams. * EXPORT_VARS * SWITCH_R_SDP - <Sdp String> - Changes The Remote Sdp Header * SWITCH_L_SDP - <Sdp String> - Changes The Local Sdp Header * SWITCH_M_SDP - <Sdp String> - Merged? Sdp * BRIDGE_TO - * SIGNAL_BRIDGE_TO - * SIGNAL_BOND * ORIGINATOR - * ORIGINATOR_CODEC - <Codec> - Sets The Codec For Calls Originated
From Leg A (Setting The Codec For Leg B)
* LOCAL_MEDIA_IP - * LOCAL_MEDIA_PORT - * REMOTE_MEDIA_IP - * REMOTE_MEDIA_PORT - * REMOTE_VIDEO_IP * REMOTE_VIDEO_PORT * LOCAL_VIDEO_IP * LOCAL_VIDEO_PORT * HANGUP_AFTER_BRIDGE - <True|False> - 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. * EXEC_AFTER_BRIDGE_APP * EXEC_AFTER_BRIDGE_ARG * MAX_FORWARDS - * DISABLE_APP_LOG * PASS_RFC2833 - <True|False> - Passes Rfc2833 Dtmf'S From One Side Of A Bridge To The Other Untouched. If Unset, It Decodes And Reencodes Them Before Passing Them On. * DEFAULT_LANGUAGE - <Language I.E. Fr> - 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. * CONTINUE_ON_FAIL - <True|False|Couses> - 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. * CALLER_ID_NAME - <Name> - The Caller Id Name Set By The Inbound Call, Not A Real Variable. Practically It Is Read Only. * CALLER_ID_NUMBER - <Number> - The Caller Id Phone Number Set By The Inbound Call, Not A Real Variable. Practically It Is Read Only. * EFFECTIVE_CALLER_ID_NAME - <Name> - Sets The Effective Callerid Name. Automatically Exported To Originated Channels. It Is A Var Set On A Channel In Case It Becomes The A Leg Of A Bridge - It Will Use This Callerid Name On The B Leg Outbound Call. * EFFECTIVE_CALLER_ID_NUMBER - <Number> - Sets The Effective Callerid Number. Automatically Exported To Originated Channels. It Is A Var Set On A Channel In Case It Becomes The A Leg Of A Bridge - It Will Use This Callerid Number On The B Leg Outbound Call. * ABSOLUTE_CODEC_STRING - <Codec> - Sets The Absolute Codec To Use * CODEC_STRING - <Codecs> - Sets The Codec To Add To The List Of Codecs. * RECORD_ANSWER_REQ - <True|False> - Start Recording Only When The Channel Has Been Answered. * RECORD_STEREO - <True|False> - Record Leg A And Leg B Streams Into Different Channel In A Stereo File. * TIMER_NAME - <Soft|Other> - If Set Will Make Playback And Speak Use A Timer To Clock The Audio Instead Of The Read. * FIRE_ASR_EVENTS - * SOUND_PREFIX - * _SOFIA_REPLACES_ - * ALERT_INFO - * ABSOLUTE_CODEC_STRING - <Codec> - Sets The Absolute Codec To Use * READ_CODEC - The Negotiated Codec Of The Inbound Call Leg. * WRITE_CODEC - The Negotiated Codec Of The Outbound Call Leg. * RECORD_RATE - <Rate> - Set The Sample Rate Of The Recording. Usage: Record_Rate=11025 Recording Meta Data - The following are set prior to performing a record to set the file header meta data provided the file format supports meta headers. * RECORD_TITLE - <Text> - Set Prior To Performing A Record To Set The File Header Meta Data Provided The File Format Supports Meta Headers. * RECORD_COPYRIGHT - <Text> - Set Prior To Performing A Record To Set The File Header Meta Data Provided The File Format Supports Meta Headers. * RECORD_SOFTWARE - <Text> - Set Prior To Performing A Record To Set The File Header Meta Data Provided The File Format Supports Meta Headers. * RECORD_ARTIST - <Text> - Set Prior To Performing A Record To Set The File Header Meta Data Provided The File Format Supports Meta Headers. * RECORD_COMMENT - <Text> - Set Prior To Performing A Record To Set The File Header Meta Data Provided The File Format Supports Meta Headers. * RECORD_DATE - <Text> - Set Prior To Performing A Record To Set The File Header Meta Data Provided The File Format Supports Meta Headers. Originate Variables Types * FAIL_ON_SINGLE_REJECT = <true|false> - 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. * IGNORE_EARLY_MEDIA = <true|false> - Controls if the call returns on early media or not. Default is false. * GROUP_CONFIRM_KEY = <key> - the call acceptance DTMF key. If <key> is "EXEC" it will run the application stated in group_confirm_file. i.e: group_confirm_key = 3 * GROUP_CONFIRM_FILE = <audio_file> - sound file for asking an approval DTMF. If group_confirm_key is "EXEC", this argument will have <application><args> as data and will run it before executing call. i.e. group_confirm_file = /wav/announce.wav * FORKED_DIAL * RETURN_RING_READY * ORIGINATE_RETRIES = <number> - number of retries * ORIGINATE_RETRY_SLEEP_MS = <number> - how many milisec between retries * ORIGINATION_CALLER_ID_NAME = <name> - Sets the origination callerid name. (LEG A) * ORIGINATION_CALLER_ID_NUMBER = <number> - Sets the origination callerid number. (LEG A) * RINGBACK = <audio_path> - Ring back is used to specify a sound/tone to be played for the calling user. The value is interpreted as a file if a path is given, else as a teletone. * TRANSFER_RINGBACK = <media_path> - 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 definte in ~/autoload_configs/switch.conf.xml : transfer_ringback="$${fr-ring}" * ORIGINATE_TIMEOUT = <number> - duration of timeout of originate attempt Since there's barely 5% match between our lists, I assume I'm completely off... Well, I took the variable names right out of the source code which is mostly lower case so that might account for a lot of the differences. However, I will use your list as a reference just to make sure that I've not missing anything... -MC _______________________________________________ Freeswitch-dev mailing list Freeswitch-dev@lists.freeswitch.org http://lists.freeswitch.org/mailman/listinfo/freeswitch-dev UNSUBSCRIBE:http://lists.freeswitch.org/mailman/options/freeswitch-dev http://www.freeswitch.org -- Anthony Minessale II FreeSWITCH http://www.freeswitch.org/ ClueCon http://www.cluecon.com/ AIM: anthm MSN:anthony_minessale@hotmail.com <mailto:MSN%3Aanthony_minessale@hotmail.com> GTALK/JABBER/PAYPAL:anthony.minessale@gmail.com <mailto:PAYPAL%3Aanthony.minessale@gmail.com> IRC: irc.freenode.net #freeswitch FreeSWITCH Developer Conference sip:888@conference.freeswitch.org <mailto:sip%3A888@conference.freeswitch.org> iax:guest@conference.freeswitch.org/888 googletalk:conf+888@conference.freeswitch.org <mailto:googletalk%3Aconf%2B888@conference.freeswitch.org> pstn:213-799-1400
Hi, I see you have a mod_amd, I don't see that in the trunk. Is it yours? Is it available? Is it for sale? I'm really interested in a good amd module. Please let me know. Cesar Cepeda. De: freeswitch-dev-bounces@lists.freeswitch.org [mailto:freeswitch-dev-bounces@lists.freeswitch.org] En nombre de Michael Collins Enviado el: Viernes, 06 de Junio de 2008 11:42 Para: freeswitch-dev@lists.freeswitch.org Asunto: [Freeswitch-dev] INFO: Channel variables Guys, I made a list by grepping and perling around in the source. grep -Pr "c(.)(\s+)switch_channel_set_variable\(channel, \"" * | grep -v "Binary" > /tmp/chan_vars.tmp This gave me a massive list of all filenames and occurrences of "switch_channel_set_variable" with a channel variable explicitly named in quotes. I then filtered out stuff with svn in the source filename and then create a tab-separated list of chan variables and source filenames. Finally, I distilled that list with this script: perl -n -e 'm/(\w+)\t(.*)/;next if (exists($seen{$1}));print $1 . "\t" . $2 . "\n";$seen{$1}=$2;' /tmp/chan_vars3.tmp This yielded a nice list of variables. Let me know what you think about using this as the base list for channel variables on the wiki: Variable Source File: limit_realm mod/applications/mod_limit/mod_limit.c limit_id mod/applications/mod_limit/mod_limit.c limit_max mod/applications/mod_limit/mod_limit.c amd_result mod/applications/mod_amd/mod_amd.c amd_status mod/applications/mod_amd/mod_amd.c amd_wait_status mod/applications/mod_amd/mod_amd.c amd_wait_result mod/applications/mod_amd/mod_amd.c enum_route_count mod/applications/mod_enum/mod_enum.c enum_auto_route mod/applications/mod_enum/mod_enum.c voicemail_total_new_messages mod/applications/mod_voicemail/mod_voicemail.c voicemail_total_saved_messages mod/applications/mod_voicemail/mod_voicemail.c voicemail_urgent_new_messages mod/applications/mod_voicemail/mod_voicemail.c voicemail_urgent_saved_messages mod/applications/mod_voicemail/mod_voicemail.c voicemail_current_folder mod/applications/mod_voicemail/mod_voicemail.c voicemail_account mod/applications/mod_voicemail/mod_voicemail.c voicemail_domain mod/applications/mod_voicemail/mod_voicemail.c voicemail_caller_id_number mod/applications/mod_voicemail/mod_voicemail.c voicemail_caller_id_name mod/applications/mod_voicemail/mod_voicemail.c voicemail_file_path mod/applications/mod_voicemail/mod_voicemail.c voicemail_read_flags mod/applications/mod_voicemail/mod_voicemail.c voicemail_time mod/applications/mod_voicemail/mod_voicemail.c voicemail_priority mod/applications/mod_voicemail/mod_voicemail.c voicemail_message_len mod/applications/mod_voicemail/mod_voicemail.c voicemail_email mod/applications/mod_voicemail/mod_voicemail.c RECORD_ARTIST mod/applications/mod_voicemail/mod_voicemail.c RECORD_DATE mod/applications/mod_voicemail/mod_voicemail.c RECORD_SOFTWARE mod/applications/mod_voicemail/mod_voicemail.c RECORD_TITLE mod/applications/mod_voicemail/mod_voicemail.c RECORD_COMMENT mod/applications/mod_voicemail/mod_voicemail.c RECORD_COPYRIGHT mod/applications/mod_voicemail/mod_voicemail.c voicemail_notify_email mod/applications/mod_voicemail/mod_voicemail.c mailbox mod/applications/mod_dptools/mod_dptools.c user_name mod/applications/mod_dptools/mod_dptools.c domain_name mod/applications/mod_dptools/mod_dptools.c xfer_uuids mod/applications/mod_dptools/mod_dptools.c dialed_user mod/applications/mod_dptools/mod_dptools.c dialed_domain mod/applications/mod_dptools/mod_dptools.c sip_force_video_fmtp mod/applications/mod_fsv/mod_fsv.c fifo_priority mod/applications/mod_fifo/mod_fifo.c fifo_status mod/applications/mod_fifo/mod_fifo.c fifo_timestamp mod/applications/mod_fifo/mod_fifo.c fifo_target mod/applications/mod_fifo/mod_fifo.c dl_user mod/endpoints/mod_dingaling/mod_dingaling.c dl_host mod/endpoints/mod_dingaling/mod_dingaling.c dl_from_user mod/endpoints/mod_dingaling/mod_dingaling.c dl_from_host mod/endpoints/mod_dingaling/mod_dingaling.c sip_nat_detected mod/endpoints/mod_sofia/sofia_glue.c sofia_profile_name mod/endpoints/mod_sofia/sofia_glue.c auto_answer_destination mod/endpoints/mod_sofia/sofia.c sip_user_agent mod/endpoints/mod_sofia/sofia.c sip_rtp_txstat mod/endpoints/mod_sofia/sofia.c sip_rtp_rxstat mod/endpoints/mod_sofia/sofia.c sip_call_id mod/endpoints/mod_sofia/sofia.c sip_authorized mod/endpoints/mod_sofia/sofia.c sip_term_status mod/endpoints/mod_sofia/sofia.c sip_term_cause mod/endpoints/mod_sofia/sofia.c transfer_fallback_extension mod/endpoints/mod_sofia/sofia.c call_clientcode mod/endpoints/mod_sofia/sofia.c sofia_record_file mod/endpoints/mod_sofia/sofia.c sip_received_ip mod/endpoints/mod_sofia/sofia.c sip_received_port mod/endpoints/mod_sofia/sofia.c sip_looped_call mod/endpoints/mod_sofia/sofia.c sip_from_user_stripped mod/endpoints/mod_sofia/sofia.c sip_from_comment mod/endpoints/mod_sofia/sofia.c sofia_profile_domain_name mod/endpoints/mod_sofia/sofia.c sip_to_comment mod/endpoints/mod_sofia/sofia.c sip_referred_by_user_stripped mod/endpoints/mod_sofia/sofia.c sip_referred_by_cid mod/endpoints/mod_sofia/sofia.c sip_subject mod/endpoints/mod_sofia/sofia.c sip_via_host mod/endpoints/mod_sofia/sofia.c sip_via_port mod/endpoints/mod_sofia/sofia.c sip_via_rport mod/endpoints/mod_sofia/sofia.c sip_gateway mod/endpoints/mod_sofia/sofia.c alert_info mod/endpoints/mod_sofia/sofia.c presence_id mod/endpoints/mod_sofia/sofia.c call_timeout mod/dialplans/mod_dialplan_asterisk/mod_dialplan_asterisk.c EXTEN mod/dialplans/mod_dialplan_asterisk/mod_dialplan_asterisk.c CHANNEL mod/dialplans/mod_dialplan_asterisk/mod_dialplan_asterisk.c UNIQUEID mod/dialplans/mod_dialplan_asterisk/mod_dialplan_asterisk.c socket_path mod/event_handlers/mod_event_socket/mod_event_socket.c socket_host mod/event_handlers/mod_event_socket/mod_event_socket.c last_file_position mod/languages/mod_spidermonkey/mod_spidermonkey.c hangup_cause switch_channel.c endpoint_disposition switch_channel.c start_stamp switch_channel.c profile_start_stamp switch_channel.c answer_stamp switch_channel.c progress_stamp switch_channel.c progress_media_stamp switch_channel.c end_stamp switch_channel.c start_epoch switch_channel.c start_uepoch switch_channel.c profile_start_epoch switch_channel.c profile_start_uepoch switch_channel.c answer_epoch switch_channel.c answer_uepoch switch_channel.c end_epoch switch_channel.c end_uepoch switch_channel.c last_app switch_channel.c last_arg switch_channel.c caller_id switch_channel.c duration switch_channel.c billsec switch_channel.c progresssec switch_channel.c progress_mediasec switch_channel.c flow_billsec switch_channel.c mduration switch_channel.c billmsec switch_channel.c progressmsec switch_channel.c progress_mediamsec switch_channel.c flow_billmsec switch_channel.c uduration switch_channel.c billusec switch_channel.c progressusec switch_channel.c progress_mediausec switch_channel.c flow_billusec switch_channel.c read_codec switch_core_codec.c read_rate switch_core_codec.c write_codec switch_core_codec.c write_rate switch_core_codec.c video_read_codec switch_core_codec.c video_read_rate switch_core_codec.c video_write_codec switch_core_codec.c video_write_rate switch_core_codec.c sound_prefix switch_ivr_play_say.c
participants (5)
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Anthony Minessale
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Brian West
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Cesar Cepeda
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Michael Collins
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UV