[Mar 16 16:51:53] WARNING[4452]: loader.c:2381 load_modules: Some non-required modules failed to load.
[Mar 16 16:51:53] ERROR[4452]: loader.c:2396 load_modules: cdr_sqlite3_custom declined to load.
[Mar 16 16:51:53] ERROR[4452]: loader.c:2396 load_modules: cel_sqlite3_custom declined to load.
I exactly work with asterisk-17.3.0
As stated these are non-required modules. If you don’t need them then you can ignore it.
Perhaps , but i don’t know well . the “SIP SHOW PEERS” isn’t recognized by the CLI
Did the chan_sip module load? Was it built? Have you confirmed the module exists in /usr/lib/asterisk/modules?
Yes of course it does…chan_sip.so
And I’ve got that==========>
root@asterisk:/usr/lib/asterisk/modules# ls
app_adsiprog.so func_math.so
app_agent_pool.so func_md5.so
app_alarmreceiver.so func_module.so
app_amd.so func_periodic_hook.so
app_attended_transfer.so func_pitchshift.so
app_authenticate.so func_pjsip_aor.so
app_blind_transfer.so func_pjsip_contact.so
app_bridgeaddchan.so func_pjsip_endpoint.so
app_bridgewait.so func_presencestate.so
app_cdr.so func_rand.so
app_celgenuserevent.so func_realtime.so
app_chanisavail.so func_sha1.so
app_channelredirect.so func_shell.so
app_chanspy.so func_sorcery.so
app_confbridge.so func_sprintf.so
app_controlplayback.so func_srv.so
app_db.so func_strings.so
app_dial.so func_sysinfo.so
app_dictate.so func_talkdetect.so
app_directed_pickup.so func_timeout.so
app_directory.so func_uri.so
app_disa.so func_version.so
app_dumpchan.so func_vmcount.so
app_echo.so func_volume.so
app_exec.so pbx_ael.so
app_externalivr.so pbx_config.so
app_festival.so pbx_dundi.so
app_followme.so pbx_loopback.so
app_forkcdr.so pbx_realtime.so
app_getcpeid.so pbx_spool.so
app_ices.so res_adsi.so
app_image.so res_ael_share.so
app_milliwatt.so res_agi.so
app_minivm.so res_ari_applications.so
app_mixmonitor.so res_ari_asterisk.so
app_morsecode.so res_ari_bridges.so
app_mp3.so res_ari_channels.so
app_nbscat.so res_ari_device_states.so
app_originate.so res_ari_endpoints.so
app_page.so res_ari_events.so
app_playback.so res_ari_model.so
app_playtones.so res_ari_playbacks.so
app_privacy.so res_ari_recordings.so
app_queue.so res_ari.so
app_readexten.so res_ari_sounds.so
app_read.so res_calendar.so
app_record.so res_clialiases.so
app_sayunixtime.so res_clioriginate.so
app_senddtmf.so res_config_sqlite3.so
app_sendtext.so res_convert.so
app_sms.so res_crypto.so
app_softhangup.so res_fax.so
app_speech_utils.so res_format_attr_celt.so
app_stack.so res_format_attr_g729.so
app_stasis.so res_format_attr_h263.so
app_stream_echo.so res_format_attr_h264.so
app_system.so res_format_attr_ilbc.so
app_talkdetect.so res_format_attr_opus.so
app_test.so res_format_attr_silk.so
app_transfer.so res_format_attr_siren14.so
app_url.so res_format_attr_siren7.so
app_userevent.so res_format_attr_vp8.so
app_verbose.so res_hep_pjsip.so
app_voicemail.so res_hep_rtcp.so
app_waitforring.so res_hep.so
app_waitforsilence.so res_http_websocket.so
app_waituntil.so res_limit.so
app_while.so res_manager_devicestate.so
app_zapateller.so res_manager_presencestate.so
bridge_builtin_features.so res_monitor.so
bridge_builtin_interval_features.so res_musiconhold.so
bridge_holding.so res_mutestream.so
bridge_native_rtp.so res_mwi_devstate.so
bridge_simple.so res_parking.so
bridge_softmix.so res_phoneprov.so
cdr_csv.so res_pjproject.so
cdr_custom.so res_pjsip_acl.so
cdr_manager.so res_pjsip_authenticator_digest.so
cdr_sqlite3_custom.so res_pjsip_caller_id.so
cel_custom.so res_pjsip_config_wizard.so
cel_manager.so res_pjsip_dialog_info_body_generator.so
cel_sqlite3_custom.so res_pjsip_diversion.so
chan_bridge_media.so res_pjsip_dlg_options.so
chan_iax2.so res_pjsip_dtmf_info.so
chan_mgcp.so res_pjsip_empty_info.so
chan_oss.so res_pjsip_endpoint_identifier_anonymous.so
chan_phone.so res_pjsip_endpoint_identifier_ip.so
chan_pjsip.so res_pjsip_endpoint_identifier_user.so
chan_rtp.so res_pjsip_exten_state.so
chan_sip.so res_pjsip_header_funcs.so
chan_skinny.so res_pjsip_history.so
chan_unistim.so res_pjsip_logger.so
codec_adpcm.so res_pjsip_messaging.so
codec_alaw.so res_pjsip_mwi_body_generator.so
codec_a_mu.so res_pjsip_mwi.so
codec_g722.so res_pjsip_nat.so
codec_g726.so res_pjsip_notify.so
codec_gsm.so res_pjsip_one_touch_record_info.so
codec_ilbc.so res_pjsip_outbound_authenticator_digest.so
codec_lpc10.so res_pjsip_outbound_publish.so
codec_resample.so res_pjsip_outbound_registration.so
codec_ulaw.so res_pjsip_path.so
format_g719.so res_pjsip_phoneprov_provider.so
format_g723.so res_pjsip_pidf_body_generator.so
format_g726.so res_pjsip_pidf_digium_body_supplement.so
format_g729.so res_pjsip_pidf_eyebeam_body_supplement.so
format_gsm.so res_pjsip_publish_asterisk.so
format_h263.so res_pjsip_pubsub.so
format_h264.so res_pjsip_refer.so
format_ilbc.so res_pjsip_registrar.so
format_pcm.so res_pjsip_rfc3326.so
format_siren14.so res_pjsip_sdp_rtp.so
format_siren7.so res_pjsip_send_to_voicemail.so
format_sln.so res_pjsip_session.so
format_vox.so res_pjsip_sips_contact.so
format_wav_gsm.so res_pjsip.so
format_wav.so res_pjsip_t38.so
func_aes.so res_pjsip_transport_websocket.so
func_base64.so res_pjsip_xpidf_body_generator.so
func_blacklist.so res_prometheus.so
func_callcompletion.so res_realtime.so
func_callerid.so res_rtp_asterisk.so
func_cdr.so res_rtp_multicast.so
func_channel.so res_security_log.so
func_config.so res_smdi.so
func_cut.so res_sorcery_astdb.so
func_db.so res_sorcery_config.so
func_devstate.so res_sorcery_memory_cache.so
func_dialgroup.so res_sorcery_memory.so
func_dialplan.so res_sorcery_realtime.so
func_enum.so res_speech.so
func_env.so res_stasis_answer.so
func_extstate.so res_stasis_device_state.so
func_frame_trace.so res_stasis_playback.so
func_global.so res_stasis_recording.so
func_groupcount.so res_stasis_snoop.so
func_hangupcause.so res_stasis.so
func_holdintercept.so res_statsd.so
func_iconv.so res_stun_monitor.so
func_jitterbuffer.so res_timing_pthread.so
func_lock.so res_timing_timerfd.so
func_logic.so
My first trouble is that I wanna manipulate le file sip.conf instead of iax2.conf
If possible how can i create peers with iax2.conf …thank you @jcolp
The iax2.conf file is used for configuring the chan_iax2 module which uses the IAX2 module. I’d suggest trying to manually load chan_sip using “module load chan_sip.so” and looking at the result. If it works then your modules.conf configuration file may not be configured to load it.
Excellent it works 
thank you!!!
asteriskCLI> module load chan_sip.so
Loaded chan_sip.so
[Mar 16 18:28:47] NOTICE[7804]: res_smdi.c:1424 load_module: No SMDI interfaces are available to listen on, not starting SMDI listener.
[Mar 16 18:28:47] WARNING[7804]: res_phoneprov.c:1232 get_defaults: Unable to find a valid server address or name.
[Mar 16 18:28:47] NOTICE[7804]: chan_skinny.c:8451 config_load: Configuring skinny from skinny.conf
[Mar 16 18:28:47] ERROR[7804]: ari/config.c:312 process_config: No configured users for ARI
[Mar 16 18:28:47] NOTICE[7804]: confbridge/conf_config_parser.c:2342 verify_default_profiles: Adding default_menu menu to app_confbridge
[Mar 16 18:28:47] NOTICE[7804]: cel_custom.c:95 load_config: No mappings found in cel_custom.conf. Not logging CEL to custom CSVs.
[Mar 16 18:28:48] WARNING[7804]: loader.c:2381 load_modules: Some non-required modules failed to load.
[Mar 16 18:28:48] ERROR[7804]: loader.c:2396 load_modules: cdr_sqlite3_custom declined to load.
[Mar 16 18:28:48] ERROR[7804]: loader.c:2396 load_modules: cel_sqlite3_custom declined to load.
Asterisk Ready.
[Mar 16 18:28:48] ERROR[7804]: logger.c:1748 logger_queue_init: Unable to create queue log: Permission denied
SIP channel loading…
[Mar 16 18:28:48] WARNING[7848]: chan_sip.c:35449 deprecation_notice: chan_sip has no official maintainer and is deprecated. Migration to
[Mar 16 18:28:48] WARNING[7848]: chan_sip.c:35450 deprecation_notice: chan_pjsip is recommended. See guides at the Asterisk Wiki:
[Mar 16 18:28:48] WARNING[7848]: chan_sip.c:35451 deprecation_notice: https://wiki.asterisk.org/wiki/display/AST/Migrating+from+chan_sip+to+res_pjsip
[Mar 16 18:28:48] WARNING[7848]: chan_sip.c:35452 deprecation_notice: https://wiki.asterisk.org/wiki/display/AST/Configuring+res_pjsip
asteriskCLI> sip show peers
Name/username Host Dyn Forcerport Comedia ACL Port Status Description
0 sip peers [Monitored: 0 online, 0 offline Unmonitored: 0 online, 0 offline]
thank you bro 
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