Hi all, when I make a test call on a new PBX I am building I see the following issue in the console when I disable module autoloading in modules.conf
[May 26 10:48:20] WARNING[2470]: res_pjsip_sdp_rtp.c:1244 create_outgoing_sdp_stream: Unable to get rtp codec payload code for testlaw
The call works fine with various codecs. The error goes away if I reenable autoloading. Does anyone know module am I missing from my configuration please?
[modules]
autoload = no
; 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
; 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_adaptive_odbc.so
; Channel Drivers
load = chan_bridge_media.so
load = chan_pjsip.so
load = chan_iax2.so
load = chan_rtp.so
load = chan_skinny.so
; Codecs
load = codec_a_mu.so
load = codec_adpcm.so
load = codec_alaw.so
load = codec_g722.so
load = codec_g726.so
load = codec_gsm.so
load = codec_ilbc.so
load = codec_lpc10.so
load = codec_resample.so
load = codec_ulaw.so
load = codec_opus.so
load = codec_silk.so
load = codec_g729-ast140-gcc4-glibc-x86_64-core2-sse4.so
; Formats
load = format_g719.so
load = format_g723.so
load = format_g726.so
load = format_g729.so
load = format_gsm.so
load = format_h263.so
load = format_h264.so
load = format_ilbc.so
load = format_jpeg.so
load = format_mp3.so
load = format_pcm.so
load = format_siren14.so
load = format_siren7.so
load = format_sln.so
load = format_vox.so
load = format_wav.so
load = format_wav_gsm.so
; Functions
load = func_db.so
load = func_odbc.so
load = func_callerid.so
load = func_cdr.so
load = func_channel.so
load = func_pjsip_aor.so
load = func_pjsip_contact.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
preload = res_odbc.so
preload = res_odbc_transaction.so
preload = res_config_odbc.so
load = res_musiconhold.so
load = res_pjproject.so
load = res_pjsip.so
load = res_pjsip_acl.so
load = res_pjsip_authenticator_digest.so
load = res_pjsip_caller_id.so
load = res_pjsip_config_wizard.so
load = res_pjsip_dialog_info_body_generator.so
load = res_pjsip_diversion.so
load = res_pjsip_dlg_options.so
load = res_pjsip_dtmf_info.so
load = res_pjsip_empty_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_history.so
load = res_pjsip_logger.so
load = res_pjsip_messaging.so
load = res_pjsip_mwi.so
load = res_pjsip_mwi_body_generator.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_phoneprov_provider.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.so
load = res_pjsip_registrar_expire.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_sips_contact.so
load = res_pjsip_t38.so
load = res_pjsip_transport_management.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_timing_timerfd.so