Can you post the message when it fails? On Mon, Mar 4, 2013 at 4:37 PM, Olivier <[email protected]> wrote:
> Hi, > > I've got a brand new Asterisk 11 setup for which I would like to keep the > number of loaded modules to a minimum. > My goal is to this setup in a pure SIP environment, for switching incoming > calls to outgoing tSIP trunks. > > When I leave autoload=yes in /etc/asterisk/modules.conf, I can handle an > incoming SIP call with a Playback app. > When I leave autoload=no in /etc/asterisk/modules.conf, it fails with with > messages I'm not familiar with. > > My question: > with autoload=no, which (efficient) method shall I use to trim the number > of modules to a minimum ? > > > Here is my modules.conf : > > [modules] > autoload=yes > > load => pbx_config.so > load => pbx_spool.so > > load => chan_local.so > load => chan_iax2.so > load => chan_sip.so > > load => app_authenticate.so > load => app_cdr.so > load => app_dial.so > load => app_dumpchan.so > load => app_echo.so > load => app_exec.so > load => app_hangup.so > load => app_macro.so > load => app_originate.so > load => app_playback.so > load => app_playtones.so > load => app_record.so > load => app_userevent.so > > load => codec_adpcm.so > load => codec_alaw.so > load => codec_a_mu.sothe number of modules to minimum ? > load => codec_g722.so > load => codec_g726.so > load => codec_gsm.so > load => codec_lpc10.so > load => codec_ulaw.so > > load => format_gsm.so > load => format_pcm.so > load => format_wav.so > load => format_wav_gsm.so > > load => res_agi.so > load => res_clioriginate.so > load => res_fax.so > load => res_musiconhold.so > load => res_timing_timerfd.so > > load => func_callerid.so > load => func_cdr.so > load => func_channel.so > load => func_cut.so > load => func_math.so > load => func_rand.so > load => func_strings.so > load => func_global.so > > load => cdr_csv.so > > > Regards > > -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > New to Asterisk? Join us for a live introductory webinar every Thurs: > http://www.asterisk.org/hello > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >
-- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
