problem with voxin and speakup

[Berkó Norbert]

Hy all,

I using ubuntu lucid 32 bit system.
I installed speech-dispatcher-voxin from vinux lucid ppa.
I running speech-dispatcher as a system daemon, so I removed comment 
before ibmtts line in /etc/speech-dispatcher/speechd.conf, and enabled 
port 6560.
Pulseaudio running as a system daemon.
My default module is ibmtts. I would like to use speakup with speechd-up 
and voxin.
If i running speech-dispatcher  with espeak, speakup works well.
  I would like to use espeakup, but do not work well with pulseaudio.
This is my speechd.conf file:

# Global configuration for Speech Dispatcher
# ==========================================

# -----SYSTEM OPTIONS-----

# The Port on which Speech Dispatcher should be available
# to clients.

Port 6560

# By default, the specified port is opened only for connections
# comming from localhost. If LocalhostAccessOnly is set to 0 it
# disables this access controll. It means that the port will be
# accessible from all computers on the network. If you turn off this
# option, please make sure you set up some system rules on what
# computers are and are not allowed to access the Speech Dispatcher
# port.

# LocalhostAccessOnly 1

# -----LOGGING CONFIGURATION-----

# The LogLevel is a number between 0 and 5 that specifies
# how much of the logging information should be printed
# out on the screen or in the logfile (see LogFile)
# 0 means nothing, 5 means everything (not recommended).

LogLevel  0

# The LogDir specifies where Speech Dispatcher writes its logging messages
# (status information, error messages, etc.).  Specify "stdout" for
# standard console output or a directory path. 'default' means that
# the logs are written to the default destination (e.g. a preconfigured
# system directory or the home directory if .speech-dispatcher is present)
# DO NOT COMMENT OUT THIS OPTION, SET IT TO "default" if you do not
# want to influence it.

LogDir  "default"
#LogDir  "/var/log/speech-dispatcher/"
#LogDir  "stdout"

# The CustomLogFile allows logging all messages of the given kind,
# regardless their priority, to the given destination.

#CustomLogFile "protocol" 
"/var/log/speech-dispatcher/speech-dispatcher-protocol.log"

# ----- VOICE PARAMETERS -----

# The DefaultRate controls how fast the synthesizer is going to speak.
# The value must be between -100 (slowest) and +100 (fastest), default
# is 0.

# DefaultRate  0

# The DefaultPitch controls the pitch of the synthesized voice.  The
# value must be between -100 (lowest) and +100 (highest), default is
# 0.

# DefaultPitch  0

# The DefaultVolume constrols the default volume of the voice.  It is
# a value between -100 (softly) and +100 (loudly).  Currently, +100
# maps to the default volume of the synthesizer.

DefaultVolume 100

# The DefaultVoiceType controls which voice type should be used by
# default.  Voice types are symbolic names which map to particular
# voices provided by the synthesizer according to the output module
# configuratuion.  Please see the synthesizer-specific configuration
# in etc/speech-dispatcher/modules/ to see which voices are assigned to
# different symbolic names.  The following symbolic names are
# currently supported: MALE1, MALE2, MALE3, FEMALE1, FEMALE2, FEMALE3,
# CHILD_MALE, CHILD_FEMALE

# DefaultVoiceType  "MALE1"

# The Default language with which to speak

# DefaultLanguage "en"


# ----- MESSAGE DISPATCHING CONTROLL -----

# The DefaultClientName specifies the name of a client who didn't
# introduce himself at the beginning of an SSIP session.

# DefaultClientName  "unknown:unknown:unknown"

# The Default Priority. Use with caution, normally this shouldn't be
# changed globally (at this place)

# DefaultPriority  "text"

# The DefaultPauseContext specifies by how many index marks a speech
# cursor should return when resuming after a pause. This is roughly
# equivalent to the number of sentences before the place of the
# execution of pause that will be repeated.

# DefaultPauseContext 0

# -----SPELLING/PUNCTUATION/CAPITAL LETTERS  CONFIGURATION-----

# The DefaultPunctuationMode sets the way dots, comas, exclamation
# marks, question marks etc. are interpreted.  none: they are ignored
# some: some of them are sent to synthesis (see
# DefaultPunctuationSome) all: all punctuation marks are sent to
# synthesis

# DefaultPunctuationMode "none"

# The DefaultCapLetRecognition: if set to "spell", capital letters
# should be spelled (e.g. "capital b"), if set to "icon",
# capital letters are indicated by inserting a special sound
# before them but they should be read normally, it set to "none"
# capital letters are not recognized (by default)

# DefaultCapLetRecognition  "none"

# The DefaultSpelling: if set to On, all messages will be spelled
# unless set otherwise (this is usually not something you want to do.)

# DefaultSpelling  Off

# ----- AUDIO CONFIGURATION -----------

# -- AUDIO OUTPUT --

# Chooses between three possible sound output systems:
#       "oss"   - Open Sound System
#       "alsa"  - Advanced Linux Sound System
#       "nas"   - Network Audio System
#       "pulse" - PulseAudio
# ALSA is default and recommended. The recent implementations
# support mixing of multiple streams. OSS is only provided
# for compatibility with architectures that do not include ALSA.
# NAS is an audio server with higher level of control over
# your audio stream, with the possibility to stream your audio
# over the network to a different computer and other advanced
# features. (The NAS backend is not very well tested however.)
# PulseAudio is a sound server for POSIX and WIN32 systems.
#

# AudioOutputMethod "pulse,alsa"

# What ALSA device to use when Advanced Linux Sound Architecture is
# chosen for the audio output.

#AudioALSADevice "default"

# -- PulseAudio parameters --

#AudioPulseServer "default"

# Maximum length of the buffer

#AudioPulseMaxLength -1

# Target length of the buffer
# The server tries to assure that at least FestivalPulseTargetLength
# bytes are always available in the buffer

#AudioPulseTargetLength 4410

# Pre-buffering
# The server does not start with playback before at least
# FestivalPulsePrebuffering bytes are available in the buffer

#AudioPulsePreBuffering -1

# Minimum request
# The server does not request less than FestivalPulseMinRequest bytes
# from the client, instead waits until the buffer is free enough to
# request more bytes at once

#AudioPulseMinRequest -1

# -- OSS parameters --

# What OSS device to use when Open Sound System is
# chosen for the audio output.

#AudioOSSDevice "/dev/dsp"

# -- NAS parameters --

# Route to the Network Audio System server when NAS
# was chosen for the audio output. Note that NAS
# server doesn't need to run on your machine,
# you can use it also over network (for instance
# when working on remote machines).

#AudioNASServer "tcp/localhost:5450"



# -----OUTPUT MODULES CONFIGURATION-----

# Each AddModule line loads an output module.
#  Syntax: AddModule "name" "binary" "configuration" "logfile"
#  - name is the name under which you can acces this module
#  - binary is the path to the binary executable of this module,
#    either relative (to lib/speech-dispatcher-modules/) or absolute
#  - configuration is the path to the config file of this module,
#    either relative (to etc/speech-dispatcher/modules/) or absolute

AddModule "espeak"       "sd_espeak"   "espeak.conf"
#AddModule "festival"     "sd_festival"  "festival.conf"
#AddModule "flite"        "sd_flite"     "flite.conf"
#AddModule "ivona"     "sd_ivona"    "ivona.conf"
#AddModule "espeak-generic" "sd_generic" "espeak-generic.conf"
#AddModule "espeak-mbrola-generic" "sd_generic" "espeak-mbrola-generic.conf"
#AddModule "swift-generic" "sd_generic" "swift-generic.conf"
#AddModule "epos-generic" "sd_generic"   "epos-generic.conf"
#AddModule "dtk-generic"  "sd_generic"   "dtk-generic.conf"
AddModule "ibmtts"       "sd_ibmtts"    "ibmtts.conf"
#AddModule "cicero"        "sd_cicero"     "cicero.conf"

# DO NOT REMOVE the following line unless you have
# a specific reason -- this is the fallback output module
# that is only used when no other modules are in use
AddModule "dummy"         "sd_dummy"      ""

# The output module testing doesn't actually connect to anything. It
# outputs the requested commands to standard output and reads
# responses from stdandard input. This way, Speech Dispatcher's
# communication with output modules can be tested easily.

# AddModule "testing"

# The DefaultModule selects which output module is the default.  You
# must use one of the names of the modules loaded with AddModule.

DefaultModule ibmtts

# The LanguageDefaultModule selects which output modules are prefered
# for specified languages.

#LanguageDefaultModule "en"  "espeak"
#LanguageDefaultModule "cs"  "festival"
#LanguageDefaultModule "es"  "festival"

# -----CLIENT SPECIFIC CONFIGURATION-----

# Here you can include the files with client-specific configuration
# for different types of clients. They must contain one or more sections 
with
# this structure:
#     BeginClient "emacs:*"
#          DefaultPunctuationMode "some"
#          ...and/or some other settings
#     EndClient
# The parameter of BeginClient tells Speech Dispatcher to which clients
# it should apply this settings (it does glob-style matching, you can use
# * to match any number of characters and ? to match one character)

# There are some sample client settings

Include "/etc/speech-dispatcher/clients/*.conf"
Include "clients/*.conf"

# This line below is to enable autospawning, without breaking everything 
else read by dotconf.
# AutoSpawn

What is the problem in my system?

Thank you for your help, and sorry for my mistakes.

Norbert