Ok, here is something you should look into, run
echo $LVBIN
echo $LVLIV
etc and see if your environment variables have been set
Let me know
Also, this is not a development issue so please don't post
to the dev
list use the user list.
Dave
.
This problem may come from environment variables not being
set
correctly. Make sure that you have logged out and back in
after
installing the Speech Engine.
I know a user with a similar problem using 1.4b2 previously
posted the
following, which he said helped:
Added lumenvox.sh (chmod +x) to /etc/asterisk/startup.d with
the
following (pulled from root's .bashrc):
export LVBIN=/opt/lumenvox/engine_7.0/bin
export LVLIB=/opt/lumenvox/engine_7.0/lib
export LVINCLUDE=/opt/lumenvox/engine_7.0/include
export LVLANG=/opt/lumenvox/engine_7.0/Lang
export LVRESPONSE=/opt/lumenvox/engine_7.0/Lang
export LD_LIBRARY_PATH=$LVLIB:$LD_LIBRARY_PATH
export LD_RUN_PATH=$LVLIB:$LD_RUN_PATH
--
Stephen Keller
LumenVox Technical Support
P: 877-977-0707, just say "Stephen Keller" or
"Support"
F: 858-707-7072
support at LumenVox.com
www.LumenVox.com
LumenVox
On Thu, 2007-07-26 at 15:32 -0700, jazzy singh wrote:
> Hey David,
> 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
>
> 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)
> my version of zaptel -1.2.10
> lumenvox7.5
>
> this is the error I get when starting asterisk
>
> Asterisk Dynamic Loader Starting:
> == Parsing '/etc/asterisk/modules.conf': Found
> [res_musiconhold.so] => (Music On Hold Resource)
> == Registered application 'MusicOnHold'
> == Registered application 'WaitMusicOnHold'
> == Registered application 'SetMusicOnHold'
> == Registered application 'StartMusicOnHold'
> == Registered application 'StopMusicOnHold'
> == Parsing '/etc/asterisk/musiconhold.conf': Found
> [res_speech.so] => (Lumenvox Speech Recognition)
> Jul 26 14:39:09 NOTICE[5770]: res_speech_lumenvox.c:758
load_module:
> Lumenvox SRE Connector module Copyright (C) 1999-2007
Digium, Inc.
> 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.
> == Parsing '/etc/asterisk/lumenvox.conf': Found
> -- Using server(s): 127.0.0.1
> -- Loaded grammar 'plan'
> -- Loaded grammar 'one_to_ten'
> -- Loaded tweaking profile default
> asterisk: symbol lookup
> error: /usr/lib/asterisk/modules/res_speech.so:
undefined symbol:
> ast_speech_register
>
>
>
>
>
> this is what my modules.conf looks like...
> ;
> ; Asterisk configuration file
> ;
> ; Module Loader configuration file
> ;
>
> [modules]
> autoload=yes
> ;
> ; Any modules that need to be loaded before the
Asterisk core has been
> ; initialized (just after the logger has been
initialized) can be
> loaded
> ; using 'preload'. This will frequently be needed if
you wish to map
> all
> ; module configuration files into Realtime storage,
since the Realtime
> ; driver will need to be loaded before the modules
using those
> configuration
> ; files are initialized.
> ;
> ; An example of loading ODBC support would be:
> ;preload => res_odbc.so
> ;preload => res_config_odbc.so
> ;
> ; If you want, load the GTK console right away.
> ; Don't load the KDE console since
> ; it's not as sophisticated right now.
> ;
> noload => pbx_gtkconsole.so
> ;load => pbx_gtkconsole.so
> noload => pbx_kdeconsole.so
> ;
> ; Intercom application is obsoleted by
> ; chan_oss. Don't load it.
> ;
> noload => app_intercom.so
> ;
> ; The 'modem' channel driver and its subdrivers are
> ; obsolete, don't load them.
> ;
> noload => chan_modem.so
> noload => chan_modem_aopen.so
> noload => chan_modem_bestdata.so
> noload => chan_modem_i4l.so
> ;
> load => res_musiconhold.so
> ;
> ; Load either OSS or ALSA, not both
> ; By default, load OSS only (automatically) and do not
load ALSA
> ;
> noload => chan_alsa.so
> load => res_speech.so
> load => res_speech_lumenvox.so
> ;noload => chan_oss.so
> ;
> ; Module names listed in "global" section
will have symbols globally
> ; exported to modules loaded after them.
> ;
> [global]
>
>
>
>
> this is what my lumenvox.com looks like
>
> ; Lumenvox configuration file
>
> [general]
> servers=127.0.0.1 ; SRE Servers to use
> save_sound_files=no ; Set to yes to save sound files
for tuner usage
>
> ; Pre-loaded grammars
> [grammars]
> plan=/etc/asterisk/grammars/plan.gram
> one_to_ten=/etc/asterisk/grammars/one_to_ten.gram
>
> ; Lumenvox tweaking profiles
> ; A tweaking profile can be used by using the
SPEECH_ENGINE dialplan
> function. For example,
> ; to apply the default profile you would use
> Set(SPEECH_ENGINE(profile)=default)
>
> ; NOTE: Each option can also be set in the dialplan by
using
> Set(SPEECH_ENGINE(name)=value)
> ; Example: Set(SPEECH_ENGINE(vad_eos_delay)=100)
>
> ; Default settings
> [default]
> ; SNR threshold/Barge-In Level.
> ; An audio frame will be considered for voice activity
only when the
> SNR metric is higher than this threshold.
> ; So for noisy channel, this value should be lower, so
that it is
> easier to barge in.
> ; NOTE: this value is not a measurement in dB. It is
just a relative
> value compared to an internal standard.
> vad_bargein_level=40
>
> ; End-of-speech delay in ms.
> ; The initial audio used to initialize voice activity
detector.
> ; Setting this parameter to 60 means using the first
60ms as
> initialization data.
> vad_eos_delay=2000
>
> ; Noise floor threshold.
> ; An audio frame will be considered for voice activity
only when the
> average energy
> ; is higher than this threshold. The default value is
200. This
> parameter is
> ; particularly useful when the echo canceler doesn't
work very well.
> ; Because the residual echo has all features of voice,
so the only way
> we
> ; can filter them out is using a hard constraint on
energy.
> vad_noise_floor=200
>
> ; Wind back.
> ; The length of audio wound back from the point of
voice detection.
> ; It helps in the situation of weak speech onset. The
resolution of
> this
> ; parameter is 1/8 sec,i.e. 125ms, which means setting
this value to
> 249ms
> ; is same as setting it to 125ms.
> vad_wind_back=255
>
> ; Burst control threshold in ms.
> ; Barge-in will be triggered only when the duration of
voice is longer
> ; than this threshold.
> vad_burst_threshold=100
>
> ; After barge-in, the streaming interface will flag
that speech was
> detected
> ; if it detected it in the time frame specified by this
option.
> end_of_speech_timeout=8000
>
> ; Whether to use the out-of-vacabulary filter during
decode.
> use_oov_filter=no
>
>
>
>
>
____________________________________________________________
__________
> Be a better Globetrotter. Get better travel answers
from someone who
> knows.
> Yahoo! Answers - Check it out.
> _______________________________________________
> --Bandwidth and Colocation Provided by http://www.api-digital.c
om--
>
> asterisk-dev mailing list
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-dev
_______________________________________________
--Bandwidth and Colocation Provided by http://www.api-digital.c
om--
asterisk-dev mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-dev
|