Hey David,<br> I apologize if you thought that I'm wasting everyone's time overhere ...I'm not...I'm under a lot of frustration right now....I just want this thing to start working like it was till yesterday. I'm a newbie at asterisk so I'm discovering things as I go I figure out how to make it print error messages when i restart here'we go<br><br>version = I'm using asterisk-1.2.13 (it worked fine till last nite..when I was testing it and must've called 20, 30)<br>my version of zaptel -1.2.10<br>lumenvox7.5<br><br>this is the error I get when starting asterisk<br><br>Asterisk Dynamic Loader Starting:<br> == Parsing '/etc/asterisk/modules.conf': Found<br> [res_musiconhold.so] => (Music On Hold Resource)<br> == Registered application 'MusicOnHold'<br> == Registered application 'WaitMusicOnHold'<br> == Registered application 'SetMusicOnHold'<br> == Registered application
'StartMusicOnHold'<br> == Registered application 'StopMusicOnHold'<br> == Parsing '/etc/asterisk/musiconhold.conf': Found<br> [res_speech.so] => (Lumenvox Speech Recognition)<br>Jul 26 14:39:09 NOTICE[5770]: res_speech_lumenvox.c:758 load_module: Lumenvox SRE Connector module Copyright (C) 1999-2007 Digium, Inc.<br>Jul 26 14:39:09 NOTICE[5770]: res_speech_lumenvox.c:759 load_module: This module is supplied under a commercial license granted by Digium, Inc.<br> == Parsing '/etc/asterisk/lumenvox.conf': Found<br> -- Using server(s): 127.0.0.1<br> -- Loaded grammar 'plan'<br> -- Loaded grammar 'one_to_ten'<br> -- Loaded tweaking profile default<br>asterisk: symbol lookup error: /usr/lib/asterisk/modules/res_speech.so: undefined symbol: ast_speech_register<br><br><br><br><br><br>this is what my modules.conf looks like...<br>;<br>; Asterisk configuration file<br>;<br>; Module
Loader configuration file<br>;<br><br>[modules]<br>autoload=yes<br>;<br>; Any modules that need to be loaded before the Asterisk core has been<br>; initialized (just after the logger has been initialized) can be loaded<br>; using 'preload'. This will frequently be needed if you wish to map all<br>; module configuration files into Realtime storage, since the Realtime<br>; driver will need to be loaded before the modules using those configuration<br>; files are initialized.<br>;<br>; An example of loading ODBC support would be:<br>;preload => res_odbc.so<br>;preload => res_config_odbc.so<br>;<br>; If you want, load the GTK console right away.<br>; Don't load the KDE console since<br>; it's not as sophisticated right now.<br>;<br>noload => pbx_gtkconsole.so<br>;load => pbx_gtkconsole.so<br>noload => pbx_kdeconsole.so<br>;<br>; Intercom application is obsoleted by<br>; chan_oss. Don't load it.<br>;<br>noload => app_intercom.so<br>;<br>; The 'modem'
channel driver and its subdrivers are<br>; obsolete, don't load them.<br>;<br>noload => chan_modem.so<br>noload => chan_modem_aopen.so<br>noload => chan_modem_bestdata.so<br>noload => chan_modem_i4l.so<br>;<br>load => res_musiconhold.so<br>;<br>; Load either OSS or ALSA, not both<br>; By default, load OSS only (automatically) and do not load ALSA<br>;<br>noload => chan_alsa.so<br>load => res_speech.so<br>load => res_speech_lumenvox.so<br>;noload => chan_oss.so<br>;<br>; Module names listed in "global" section will have symbols globally<br>; exported to modules loaded after them.<br>;<br>[global]<br><br><br><br><br>this is what my lumenvox.com looks like <br><br>; Lumenvox configuration file<br><br>[general]<br>servers=127.0.0.1 ; SRE Servers to use<br>save_sound_files=no ; Set to yes to save sound files for tuner usage<br><br>; Pre-loaded
grammars<br>[grammars]<br>plan=/etc/asterisk/grammars/plan.gram<br>one_to_ten=/etc/asterisk/grammars/one_to_ten.gram<br><br>; Lumenvox tweaking profiles<br>; A tweaking profile can be used by using the SPEECH_ENGINE dialplan function. For example,<br>; to apply the default profile you would use Set(SPEECH_ENGINE(profile)=default)<br><br>; NOTE: Each option can also be set in the dialplan by using Set(SPEECH_ENGINE(name)=value)<br>; Example: Set(SPEECH_ENGINE(vad_eos_delay)=100)<br><br>; Default settings<br>[default]<br>; SNR threshold/Barge-In Level.<br>; An audio frame will be considered for voice activity only when the SNR metric is higher than this threshold.<br>; So for noisy channel, this value should be lower, so that it is easier to barge in.<br>; NOTE: this value is not a measurement in dB. It is just a relative value compared to an internal standard.<br>vad_bargein_level=40<br><br>; End-of-speech delay in ms.<br>; The initial audio used to initialize voice
activity detector.<br>; Setting this parameter to 60 means using the first 60ms as initialization data.<br>vad_eos_delay=2000<br><br>; Noise floor threshold.<br>; An audio frame will be considered for voice activity only when the average energy<br>; is higher than this threshold. The default value is 200. This parameter is<br>; particularly useful when the echo canceler doesn't work very well.<br>; Because the residual echo has all features of voice, so the only way we<br>; can filter them out is using a hard constraint on energy.<br>vad_noise_floor=200<br><br>; Wind back.<br>; The length of audio wound back from the point of voice detection.<br>; It helps in the situation of weak speech onset. The resolution of this<br>; parameter is 1/8 sec,i.e. 125ms, which means setting this value to 249ms<br>; is same as setting it to 125ms.<br>vad_wind_back=255<br><br>; Burst control threshold in ms.<br>; Barge-in will be triggered only when the duration of voice is longer<br>; than
this threshold.<br>vad_burst_threshold=100<br><br>; After barge-in, the streaming interface will flag that speech was detected<br>; if it detected it in the time frame specified by this option.<br>end_of_speech_timeout=8000<br><br>; Whether to use the out-of-vacabulary filter during decode.<br>use_oov_filter=no<br><br><p> 
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