[asterisk-dev] AST_MODULE_LOAD_FAILURE vs AST_MODULE_LOAD_DECLINE

[George Joseph]

Over the years I've been frustrated at times where Asterisk fails to start
for absolutely no (apparent) reason.  No error message, no trap, nothing.
It just ends.  Every case of this I've tracked down has been the result of
a module returning AST_MODULE_LOAD_FAILURE when it encounters a problem but
not bothering to print an error first.   If you don't know
already, AST_MODULE_LOAD_FAILURE tells asterisk to *stop loading and exit*.
What the module should have done was actually print an error and
return AST_MODULE_LOAD_DECLINE which just tells asterisk that the module
couldn't load.

So now I'm doing an audit of module return codes to see which call FAILURE
vs DECLINE and found the modules below all call FAILURE.

The big question is...  Exactly which modules are so critical to asterisk
that asterisk should exit if the module couldn't load?  Surely not all 103
modules listed below.  If we can agree on the ones that really *should*
call FAILURE, I'll update those with error messages, then change the rest
to DECLINE and add error messages as needed.

Reply with the list intact.  Don't add, remove, split or re-order the
lines.  Just stick an "F " before the modules you think should remain
failure.

Modules returning FAILURE:

./addons/app_mysql.c
./addons/cdr_mysql.c
./addons/chan_mobile.c
./apps/app_adsiprog.c
./apps/app_agent_pool.c
./apps/app_alarmreceiver.c
./apps/app_amd.c
./apps/app_authenticate.c
./apps/app_cdr.c
./apps/app_confbridge.c
./apps/app_festival.c
./apps/app_followme.c
./apps/app_forkcdr.c
./apps/app_meetme.c
./apps/app_queue.c
./apps/app_skel.c
./apps/app_voicemail.c
./cdr/cdr_custom.c
./cel/cel_custom.c
./cel/cel_odbc.c
./channels/chan_alsa.c
./channels/chan_console.c
./channels/chan_dahdi.c
./channels/chan_iax2.c
./channels/chan_mgcp.c
./channels/chan_misdn.c
./channels/chan_motif.c
./channels/chan_nbs.c
./channels/chan_oss.c
./channels/chan_phone.c
./channels/chan_pjsip.c
./channels/chan_sip.c
./channels/chan_skinny.c
./channels/chan_unistim.c
./channels/chan_vpb.cc
./codecs/codec_a_mu.c
./codecs/codec_adpcm.c
./codecs/codec_alaw.c
./codecs/codec_g722.c
./codecs/codec_g726.c
./codecs/codec_gsm.c
./codecs/codec_ilbc.c
./codecs/codec_lpc10.c
./codecs/codec_resample.c
./codecs/codec_ulaw.c
./formats/format_g723.c
./formats/format_g726.c
./formats/format_g729.c
./formats/format_gsm.c
./formats/format_h263.c
./formats/format_h264.c
./formats/format_ilbc.c
./formats/format_jpeg.c
./formats/format_ogg_vorbis.c
./formats/format_pcm.c
./formats/format_sln.c
./formats/format_vox.c
./formats/format_wav.c
./formats/format_wav_gsm.c
./funcs/func_cdr.c
./main/loader.c
./pbx/pbx_dundi.c
./pbx/pbx_loopback.c
./pbx/pbx_realtime.c
./pbx/pbx_spool.c
./res/res_pjsip/config_transport.c
./res/res_ari.c
./res/res_ari_events.c
./res/res_ari_model.c
./res/res_calendar.c
./res/res_chan_stats.c
./res/res_clialiases.c
./res/res_config_ldap.c
./res/res_config_sqlite.c
./res/res_config_sqlite3.c
./res/res_curl.c
./res/res_endpoint_stats.c
./res/res_fax.c
./res/res_hep_rtcp.c
./res/res_http_websocket.c
./res/res_musiconhold.c
./res/res_odbc.c
./res/res_phoneprov.c
./res/res_pjsip_nat.c
./res/res_pjsip_one_touch_record_info.c
./res/res_pjsip_outbound_publish.c
./res/res_pjsip_outbound_registration.c
./res/res_pjsip_sdp_rtp.c
./res/res_pjsip_send_to_voicemail.c
./res/res_pjsip_t38.c
./res/res_smdi.c
./res/res_snmp.c
./res/res_stasis.c
./res/res_stasis_device_state.c
./res/res_stasis_playback.c
./res/res_stasis_recording.c
./res/res_stasis_test.c
./res/res_statsd.c
./res/res_timing_kqueue.c
./res/res_xmpp.c
./res/res_pjsip_pubsub.c
./tests/test_bucket.c
./tests/test_channel_feature_hooks.c

-- 
George Joseph
Digium, Inc. | Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.digium.com & www.asterisk.org
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