No application 'AudioSocket' for extension (inbound, voicedemo, 9)

[juansalvatella]

I've been experiencing an intermittent issue but I'm not sure if it's related with the AudioSocket application or with asterisk-config. Every now and then, I get a message from Asterisk that says "No application 'AudioSocket' for extension (inbound, voicedemo, 9)" and after restarting the pod everything works as expected.

Any hints on what could be causing this?

This is my modules.conf (I added the audiosocket modules that are not included in this repo):

[modules]
autoload = yes

; This is a minimal module load. We are loading only the modules required for
; the Asterisk features used in the Super Awesome Company configuration.

; Applications

load = app_bridgewait.so
load = app_dial.so
load = app_playback.so
load = app_stack.so
load = app_verbose.so
load = app_voicemail.so
load = app_directory.so
load = app_confbridge.so
load = app_stasis.so
load = app_system.so
load = app_audiosocket.so

; Bridging

load = bridge_builtin_features.so
load = bridge_builtin_interval_features.so
load = bridge_holding.so
load = bridge_native_rtp.so
load = bridge_simple.so
load = bridge_softmix.so

; Call Detail Records

load = cdr_custom.so

; Channel Drivers

load = chan_bridge_media.so
load = chan_audiosocket.so
load = chan_pjsip.so
noload = chan_sip.so

; Codecs

load = codec_gsm.so
load = codec_resample.so
load = codec_ulaw.so
load = codec_g722.so

; Formats

load = format_gsm.so
load = format_pcm.so
load = format_wav_gsm.so
load = format_wav.so

; Functions

load = func_callerid.so
load = func_cdr.so
load = func_pjsip_endpoint.so
load = func_sorcery.so
load = func_devstate.so
load = func_strings.so

; Core/PBX

load = pbx_config.so

; Resources

load = res_audiosocket.so
load = res_musiconhold.so
load = res_pjproject.so
load = res_pjsip_acl.so
load = res_pjsip_authenticator_digest.so
load = res_pjsip_caller_id.so
load = res_pjsip_dialog_info_body_generator.so
load = res_pjsip_diversion.so
load = res_pjsip_dtmf_info.so
load = res_pjsip_endpoint_identifier_anonymous.so
load = res_pjsip_endpoint_identifier_ip.so
load = res_pjsip_endpoint_identifier_user.so
load = res_pjsip_exten_state.so
load = res_pjsip_header_funcs.so
load = res_pjsip_logger.so
load = res_pjsip_messaging.so
load = res_pjsip_mwi_body_generator.so
load = res_pjsip_mwi.so
load = res_pjsip_nat.so
load = res_pjsip_notify.so
load = res_pjsip_one_touch_record_info.so
load = res_pjsip_outbound_authenticator_digest.so
load = res_pjsip_outbound_publish.so
load = res_pjsip_outbound_registration.so
load = res_pjsip_path.so
load = res_pjsip_pidf_body_generator.so
load = res_pjsip_pidf_digium_body_supplement.so
load = res_pjsip_pidf_eyebeam_body_supplement.so
load = res_pjsip_publish_asterisk.so
load = res_pjsip_pubsub.so
load = res_pjsip_refer.so
;load = res_pjsip_registrar_expire.so
load = res_pjsip_registrar.so
load = res_pjsip_rfc3326.so
load = res_pjsip_sdp_rtp.so
load = res_pjsip_send_to_voicemail.so
load = res_pjsip_session.so
load = res_pjsip.so
load = res_pjsip_t38.so
load = res_pjsip_transport_websocket.so
load = res_pjsip_xpidf_body_generator.so
load = res_rtp_asterisk.so
load = res_sorcery_astdb.so
load = res_sorcery_config.so
load = res_sorcery_memory.so
load = res_sorcery_realtime.so
load = res_stasis_answer.so
load = res_stasis_device_state.so
load = res_stasis_playback.so
load = res_stasis_recording.so
load = res_stasis_snoop.so
load = res_stasis.so
load = res_ari_applications.so
load = res_ari_asterisk.so
load = res_ari_bridges.so
load = res_ari_channels.so
load = res_ari_device_states.so
load = res_ari_endpoints.so
load = res_ari_events.so
load = res_ari_model.so
load = res_ari_playbacks.so
load = res_ari_recordings.so
load = res_ari_sounds.so
load = res_ari.so
load = res_timing_timerfd.so

Thanks and regards,

[Ulexus]

Good point: since AudioSocket was accepted in the official Asterisk distribution, it is marked as "extended," which probably means it needs to be explicitly loaded.

[Ulexus]

Sorry I missed this issue from so long ago.