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Yeah, you suggested to use a silence prompt. Sorry I didn't know
the format to specify it and didn't realise <br>
there is an existing specific way to specify it. Now I know :-)<br>
<br>
Thanks!<br>
Xing<br>
<br>
<br>
On 17/11/11 13:37, Christopher Rienzo wrote:
<blockquote
cite="mid:CAD-qRkiE4p3LqfhHM431X6CVq-vzXX=djiu-Vt08JUE8tJDLng@mail.gmail.com"
type="cite"><br>
<br>
<div class="gmail_quote">
<blockquote class="gmail_quote" style="margin:0 0 0
.8ex;border-left:1px #ccc solid;padding-left:1ex;">
<div text="#000000" bgcolor="#FFFFFF"> <br>
After some more tests, I found followings:<br>
<br>
1. Testing from dialplan, the log output is the string of
"CRIT ${speech_detect_result}" rather than the recognition
results.<br>
</div>
</blockquote>
<div><br>
oops <br>
<br>
</div>
<blockquote class="gmail_quote" style="margin: 0pt 0pt 0pt
0.8ex; border-left: 1px solid rgb(204, 204, 204);
padding-left: 1ex;">
<div text="#000000" bgcolor="#FFFFFF"> <br>
2. Regarding to non barge in, I first send "speak" command
then send play_and_detect_speech with parameter:<br>
detect:unimrcp:nuance5-mrcp1-1
{start-input-timers=false,no-input-timeout=25000,recognition-timeout=25000}dudeYNNC_Nuance<br>
<br>
in which I removed the "say:" part.<br>
<br>
But very soon I received event
CHANNEL_EXECUTE_COMPLETE for play_and_detect_speech before
the text playing finishes,<br>
and speech_detect_result is null (actually the event
header does not contain this variable)<br>
<br>
I tried with "say:" part with empty text like
"say:unimrcp:en-GB: " but it doesn't work, see issue 3
below.<br>
</div>
</blockquote>
<div><br>
Try with silence as I originally suggested:<br>
<br>
silence_stream://1000 detect:unimrcp:nuance5-mrcp1-1
{start-input-timers=false,no-input-timeout=25000,
recognition-timeout=25000}dudeYNNC_Nuance <br>
</div>
<blockquote class="gmail_quote" style="margin: 0pt 0pt 0pt
0.8ex; border-left: 1px solid rgb(204, 204, 204);
padding-left: 1ex;">
<div text="#000000" bgcolor="#FFFFFF"> <br>
3. It seems that I can not send "speak" command twice
before the first one finishes. In my case I run one port TTS
server on one machine.<br>
<br>
If I send the command twice FS will give me Synthesizer
Error/Invalid TTS Module.<br>
I thought the TTS request would be queued rather than
it immediately looks for the TTS resource.<br>
<br>
If I send the second command to another TTS machine, no
error occurs but I can only hear one utterance being spoken,<br>
it looks like one utteraance was dropped somehow.<br>
</div>
</blockquote>
<div><br>
Wait for speak to finish before starting a new one.<br>
</div>
<blockquote class="gmail_quote" style="margin: 0pt 0pt 0pt
0.8ex; border-left: 1px solid rgb(204, 204, 204);
padding-left: 1ex;">
<div text="#000000" bgcolor="#FFFFFF">
<div class="im"> <br>
On 16/11/11 16:51, xl127 wrote: </div>
<blockquote type="cite">
<div class="im"> Hi Christopher,<br>
<br>
The questions are cleared to me now. Many thanks for
your explanations!<br>
<br>
Best regards,<br>
<br>
Xing<br>
<br>
<br>
On 16/11/11 15:52, Christopher Rienzo wrote: </div>
<blockquote type="cite"><br>
Responses inline<br>
<br>
<br>
<div class="gmail_quote">
<blockquote class="gmail_quote" style="margin:0 0 0
.8ex;border-left:1px #ccc solid;padding-left:1ex">
<div text="#000000" bgcolor="#FFFFFF">
<div class="im"> Now it works in my ESL app though
I am just able to do one dialogue ( I need to
add the event catching for furthur dialgoues).<br>
<br>
I have a couple of questions here:<br>
<br>
1. In the first try, my Nuance server was able
to be accessed somehow (FS says the MRCP is not
responding in 5000ms, <br>
</div>
something like that), then FS says: [WARNING]
rtsp_client.c:386 () Failed to Connect to RTSP
Server <a moz-do-not-send="true"
href="http://99.185.85.31:554" target="_blank"><font
color="red"><b>MailScanner has detected a
possible fraud attempt from
"99.185.85.31:554" claiming to be</b></font>
<font color="red"><b>MailScanner warning:
numerical links are often malicious:</b></font>
99.185.85.31:554</a>,
<div class="im"><br>
later FS says: <br>
[ERR] mod_unimrcp.c:1860 (TTS-6)
SYNTHESIZER channel error!<br>
[ERR] switch_ivr_play_say.c:2439 Invalid
TTS module! <br>
<br>
The SYNTHESIZER channel error and Invalid TTS
module error are obvious.<br>
<br>
</div>
What I don't understand is why it went to this
stange address: <a moz-do-not-send="true"
href="http://99.185.85.31:554" target="_blank"><font
color="red"><b>MailScanner has detected a
possible fraud attempt from
"99.185.85.31:554" claiming to be</b></font>
<font color="red"><b>MailScanner warning:
numerical links are often malicious:</b></font>
99.185.85.31:554</a>?<br>
</div>
</blockquote>
<div>
<div class="h5">
<div><br>
check your unimrcp configuration. Make sure the
default TTS and ASR profiles are set to actual
servers.<br>
</div>
<blockquote class="gmail_quote" style="margin:0pt
0pt 0pt 0.8ex;border-left:1px solid rgb(204,
204, 204);padding-left:1ex">
<div text="#000000" bgcolor="#FFFFFF"> 2. I
specified TTS engine in play_and_detect_speech
as<br>
"say:unimrcp:nuance5-mrcp1-1: the
text to speak"<br>
It works though I didn't specify the TTS
voice.<br>
<br>
How do I specify the TTS voice? In the
mrcp profile (how?)? or something like:<br>
"say:unimrcp:nuance5-mrcp1-1:Serena:
the text to speak" (this seems not right.)<br>
</div>
</blockquote>
<div><br>
That won't work. Set the tts_engine variable as
I explained previously, or use
say:unimrcp:voice:text to speak with the desired
voice and the correct default TTS profile
defined in unimrcp.conf.xml. This is a
limitation of the say: notation. Alternatively,
the voice can be defined with the tts_voice
channel variable.<br>
<br>
</div>
<blockquote class="gmail_quote" style="margin:0pt
0pt 0pt 0.8ex;border-left:1px solid rgb(204,
204, 204);padding-left:1ex">
<div text="#000000" bgcolor="#FFFFFF"> 3. The
barge-in works well, thanks!. Is the barge-in
configurable? In some scenarios, we might not
allow barge-in.<br>
</div>
</blockquote>
<div><br>
If you don't want to barge in, just do "playback
(or speak)" first, then "play_and_detect_speech"
with a silence prompt.<br>
</div>
<blockquote class="gmail_quote" style="margin:0pt
0pt 0pt 0.8ex;border-left:1px solid rgb(204,
204, 204);padding-left:1ex">
<div text="#000000" bgcolor="#FFFFFF"> <br>
4. How could I get the text which has spoken
to the user when barge-in occurs?<br>
Or Could I get the time when barge-in
occurs? If I know the barge-in time and rough
totale time for the whole text<br>
to be spoken I can figure out the spoken
text by manually checking the recorded audio
file later, which would be painful.<br>
</div>
</blockquote>
<div><br>
If this is necessary, you might want to use the
lower-level functions instead to watch for the
begin-speaking event.<br>
</div>
<blockquote class="gmail_quote" style="margin:0pt
0pt 0pt 0.8ex;border-left:1px solid rgb(204,
204, 204);padding-left:1ex">
<div text="#000000" bgcolor="#FFFFFF"> <br>
5. when I use "speak" and "detect_speech"
apps in ESL, I can catch event:
DETECTED_SPEECH and speech-type:
begin-speaking<br>
and "detected-speech", then I do the
recognition results processing.<br>
<br>
The new app play_and_detect_speech seems
not generate these events any more. The way
that I can think of to get the results<br>
is to catch event:CHANNEL_EXECUTE_COMPLETE
then check if
variable_current_application=play_and_detect_speech,
then get<br>
the results from
variable_detect_speech_result.<br>
<br>
Is this the proper way to get the results
in ESL app? Or will play_and_detect_speech
later on be consistent with detect_speech <br>
in term of ASR events?<br>
</div>
</blockquote>
<div><br>
play_and_detect_speech is a higher level
abstraction to simplify things. If you want to
have more control, go back to using the ESL
events. Reading the code in mod_dptools and
switch_ivr_async will give you hints about how
to do it correctly.<br>
</div>
<blockquote class="gmail_quote" style="margin:0pt
0pt 0pt 0.8ex;border-left:1px solid rgb(204,
204, 204);padding-left:1ex">
<div text="#000000" bgcolor="#FFFFFF"> <br>
6. I'd like to set start-input-timers=false
in the initial request then start the
recognition timers (start-input-timers=true)<br>
after the TTS finishes.<br>
How possibly could I do this?<br>
</div>
</blockquote>
<div><br>
This is automatically done in the
switch_ivr_play_and_detect_speech() function.
You just need to specify
start-input-timers=false in the beginning.<br>
</div>
</div>
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