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I want to specify <grammar> for play_and_detect_speech using
URI format.<br>
<br>
I am running a Tomcat server that hosts the grammars and I am sure
it can be accessed from local machine or from a different machine.<br>
<br>
If I use <a class="moz-txt-link-freetext" href="http://localhost:8084/grammars/FS_Nuance/myGrammar.gram">http://localhost:8084/grammars/FS_Nuance/myGrammar.gram</a><br>
or <a class="moz-txt-link-freetext" href="http://127.0.0.1:8084/grammars/FS_Nuance/myGrammar.gram">http://127.0.0.1:8084/grammars/FS_Nuance/myGrammar.gram</a><br>
<br>
FS got grammar loading failure (MRCP/1.0 1 407 COMPLETE)<br>
<br>
If I change localhost to my concrete external IP, it works for
Nuance server, but not for Voxeo Prophecy server.<br>
<br>
I also have difficulties to specify URI grammars for detect_speech.<br>
<br>
Any suggestions please?<br>
<br>
Thanks,<br>
Xing<br>
<br>
<br>
On 17/11/11 16:21, xl127 wrote:
<blockquote cite="mid:4EC5348A.6030409@hw.ac.uk" type="cite">
<meta content="text/html; charset=ISO-8859-1"
http-equiv="Content-Type">
Yeah, I see, it should be set to true there. Thanks for reminding
me this.<br>
<br>
<br>
On 17/11/11 15:48, Christopher Rienzo wrote:
<blockquote
cite="mid:CAD-qRkgSYZbK8bT_WTcsH=1BEgTwg-5rPB5bxf1MWWB6KRr9uQ@mail.gmail.com"
type="cite">Actually in that silence prompt suggestion I gave,
use {start-input-timers=true}<br>
<br>
<br>
<div class="gmail_quote">On Thu, Nov 17, 2011 at 8:37 AM,
Christopher Rienzo <span dir="ltr"><<a
moz-do-not-send="true" href="mailto:cmrienzo@gmail.com">cmrienzo@gmail.com</a>></span>
wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0
.8ex;border-left:1px #ccc solid;padding-left:1ex;"><br>
<br>
<div class="gmail_quote">
<div class="im">
<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>
<div><br>
oops <br>
<br>
</div>
<div class="im">
<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>
<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>
<div class="im">
<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>
<div><br>
Wait for speak to finish before starting a new one.<br>
</div>
<div>
<div class="h5">
<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> <br>
On 16/11/11 16:51, xl127 wrote: </div>
<blockquote type="cite">
<div> 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> 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><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>
<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>
</div>
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