Asterisk - The Open Source Telephony Project
GIT-master-3dee037
- g -
g :
Multitype
g722 :
g722_decoder_pvt
,
g722_encoder_pvt
g726 :
g726_coder_pvt
g726_non_standard :
ast_sip_endpoint_media_configuration
g729onlyA :
ooh323_peer
,
ooh323_pvt
,
ooh323_user
games_left :
skel_current_game
gateway_id :
ast_fax_session_details
gateway_timeout :
ast_fax_session_details
gen_active :
ivr_localuser
gen_add_nai :
sig_ss7_chan
gen_add_num_plan :
sig_ss7_chan
gen_add_number :
sig_ss7_chan
gen_add_pres_ind :
sig_ss7_chan
gen_add_type :
sig_ss7_chan
gen_dig_number :
sig_ss7_chan
gen_dig_scheme :
sig_ss7_chan
gen_dig_type :
sig_ss7_chan
general :
ast_ari_conf
,
ast_xmpp_global_config
,
cel_config
,
conf
,
features_global_config
,
module_config
,
udptl_config
generate :
ast_generator
generate_body_content :
ast_sip_pubsub_body_generator
generate_initial_notify :
sip_subscription_tree
generate_new_sdp :
ast_sip_session_delayed_request
generate_silence :
ast_fax_tech
generator :
ast_channel
generator_data :
subscription_persistence
generatordata :
ast_channel
genergy :
ast_dsp
generic :
ast_bridge_hook_dtmf
,
ast_bridge_hook_timer
,
ast_dns_naptr_record
,
ast_dns_srv_record
,
ast_dns_txt_record
generic_name :
sig_ss7_chan
get :
ast_jb_impl
,
ast_speech_engine
get_and_handle_alarms :
analog_callback
get_attr :
ast_sdp_crypto_api
get_callerid :
analog_callback
get_codec :
ast_rtp_glue
get_connection :
ast_rtp_engine_dtls
get_container :
ast_sip_cli_formatter_entry
get_cpeid :
adsi_funcs
get_cpeinfo :
adsi_funcs
get_event :
analog_callback
get_field :
allowed_field
get_fingerprint :
ast_rtp_engine_dtls
get_fingerprint_hash :
ast_rtp_engine_dtls
get_firstdigit_timeout :
analog_callback
get_format :
set_format_access
get_id :
ast_sip_cli_formatter_entry
get_interdigit_timeout :
analog_callback
get_length :
ast_codec
get_local_candidates :
ast_rtp_engine_ice
get_matchdigit_timeout :
analog_callback
get_merge_priority :
ast_bridge_methods
get_metric_value :
prometheus_metric
get_notify_data :
ast_sip_notifier
get_orig_dialstring :
analog_callback
,
sig_pri_callback
get_password :
ast_rtp_engine_ice
get_pvt_uniqueid :
ast_channel_tech
get_random :
ast_srtp_res
get_rawformat :
set_format_access
get_resource_display_name :
ast_sip_notifier
get_rtp_info :
ast_rtp_glue
get_setting :
ast_speech_engine
get_setup :
ast_rtp_engine_dtls
get_sigpvt_bridged_channel :
analog_callback
get_stat :
ast_rtp_engine
get_sub_fd :
analog_callback
get_trans :
set_format_access
get_trtp_info :
ast_rtp_glue
get_ufrag :
ast_rtp_engine_ice
get_value :
bridge_metric_defs
,
channel_metric_defs
,
endpoint_metric_defs
get_vrtp_info :
ast_rtp_glue
getcomment :
ast_format_def
getpin :
bridge_profile_sounds
gid :
console
,
usergroup_cli_perm
glare_mitigation :
ast_sip_direct_media_configuration
global :
conf
,
features_config
,
parking_config
,
skel_config
,
test_config
,
unbound_config
,
xmpp_config
global_defaults :
test_config
global_disable :
attestation_cfg
,
verification_cfg
global_obj :
aco_info
globbuf :
stackelement
globbuf_pos :
stackelement
glue :
ast_rtp_instance
,
native_rtp_bridge_channel_data
gmuted :
ast_conference
goahead :
state
goback :
state
gone :
jingle_session
google :
ast_xmpp_capabilities
goto_false :
ael_priority
goto_target :
pval
goto_target_in_case :
pval
goto_true :
ael_priority
gotorx :
detect_information
gototx :
detect_information
grammar :
ast_speech_result
granularity_time :
ast_aoc_duration_rate
granularity_time_scale :
ast_aoc_duration_rate
group :
ast_group_info
,
ast_stream
,
dahdi_pvt
,
mbl_pvt
group_list :
ast_group_info
groupmatch :
dahdi_starting_point
gRover :
fft_data
grow_count :
hash_test
gsamp_size :
ast_dsp
gsamps :
ast_dsp
gsm :
gsm_translator_pvt
guardtime :
analog_pvt
,
dahdi_pvt
Generated on Wed Jan 1 2025 20:04:54 for Asterisk - The Open Source Telephony Project by
1.9.4