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    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">&lt;<a
            moz-do-not-send="true" href="mailto:cmrienzo@gmail.com">cmrienzo@gmail.com</a>&gt;</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"> &nbsp; <br>
                  &nbsp; After some more tests, I found followings:<br>
                  <br>
                  &nbsp; 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>
              &nbsp; oops <br>
              &nbsp;<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>
                  &nbsp; 2. Regarding to non barge in, I first send "speak"
                  command then send play_and_detect_speech with
                  parameter:<br>
                  &nbsp;&nbsp;&nbsp;&nbsp;&nbsp; detect:unimrcp:nuance5-mrcp1-1
{start-input-timers=false,no-input-timeout=25000,recognition-timeout=25000}dudeYNNC_Nuance<br>
                  <br>
                  &nbsp;&nbsp;&nbsp;&nbsp;&nbsp; in which I removed the "say:" part.<br>
                  &nbsp;&nbsp;&nbsp;&nbsp;&nbsp; <br>
                  &nbsp;&nbsp;&nbsp;&nbsp;&nbsp; But very soon I received event
                  CHANNEL_EXECUTE_COMPLETE for play_and_detect_speech
                  before the text playing finishes,<br>
                  &nbsp;&nbsp;&nbsp;&nbsp;&nbsp; and speech_detect_result is null (actually the
                  event header does not contain this variable)<br>
                  <br>
                  &nbsp;&nbsp;&nbsp;&nbsp;&nbsp; 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>
                  &nbsp; 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>
                  &nbsp;&nbsp;&nbsp;&nbsp; If I send the command twice FS will give me
                  Synthesizer Error/Invalid TTS Module.<br>
                  &nbsp;&nbsp;&nbsp;&nbsp; I thought the TTS request would be queued rather
                  than it immediately looks for the TTS resource.<br>
                  <br>
                  &nbsp;&nbsp;&nbsp;&nbsp; If I send the second command to another TTS
                  machine, no error occurs but I can only hear one
                  utterance being spoken,<br>
                  &nbsp;&nbsp;&nbsp;&nbsp; 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>
              &nbsp;</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>
                                &nbsp; 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>
                              &nbsp;&nbsp;&nbsp; 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>
                                &nbsp;&nbsp;&nbsp; later FS says: <br>
                                &nbsp;&nbsp;&nbsp;&nbsp; [ERR] mod_unimrcp.c:1860 (TTS-6)
                                SYNTHESIZER channel error!<br>
                                &nbsp;&nbsp;&nbsp;&nbsp; [ERR] switch_ivr_play_say.c:2439
                                Invalid TTS module!&nbsp; <br>
                                &nbsp;&nbsp;&nbsp; <br>
                                &nbsp;&nbsp; The SYNTHESIZER channel error and
                                Invalid TTS module error are obvious.<br>
                                <br>
                              </div>
                              &nbsp;&nbsp;&nbsp; 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.&nbsp; Make
                                sure the default TTS and ASR profiles
                                are set to actual servers.<br>
                                &nbsp;</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"> &nbsp;
                                  2. I specified TTS engine in
                                  play_and_detect_speech as<br>
                                  &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp; "say:unimrcp:nuance5-mrcp1-1:
                                  the text to speak"<br>
                                  &nbsp;&nbsp;&nbsp;&nbsp; It works though I didn't specify
                                  the TTS voice.<br>
                                  <br>
                                  &nbsp;&nbsp;&nbsp;&nbsp; How do I specify the TTS voice?
                                  In the mrcp profile (how?)? or
                                  something like:<br>
                                  &nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;&nbsp;
                                  "say:unimrcp:nuance5-mrcp1-1:Serena:
                                  the text to speak" (this seems not
                                  right.)<br>
                                </div>
                              </blockquote>
                              <div><br>
                                That won't work.&nbsp; 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.&nbsp; This is a limitation
                                of the say: notation.&nbsp; Alternatively,
                                the voice can be defined with the
                                tts_voice channel variable.<br>
                                <br>
                                &nbsp;</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"> &nbsp;
                                  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>
                                &nbsp;</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>
                                  &nbsp; 4. How could I get the text which
                                  has spoken to the user when barge-in
                                  occurs?<br>
                                  &nbsp;&nbsp;&nbsp;&nbsp; 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>
                                  &nbsp;&nbsp;&nbsp;&nbsp; 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>
                                &nbsp;</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>
                                  &nbsp; 5. when I use "speak" and
                                  "detect_speech" apps in ESL, I can
                                  catch event: DETECTED_SPEECH and
                                  speech-type: begin-speaking<br>
                                  &nbsp;&nbsp;&nbsp;&nbsp; and "detected-speech", then I do
                                  the recognition results processing.<br>
                                  &nbsp;&nbsp;&nbsp;&nbsp; <br>
                                  &nbsp;&nbsp;&nbsp; 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>
                                  &nbsp;&nbsp;&nbsp; is to catch
                                  event:CHANNEL_EXECUTE_COMPLETE then
                                  check if
                                  variable_current_application=play_and_detect_speech,
                                  then get<br>
                                  &nbsp;&nbsp;&nbsp; the results from
                                  variable_detect_speech_result.<br>
                                  <br>
                                  &nbsp;&nbsp;&nbsp; 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>
                                  &nbsp;&nbsp;&nbsp; in term of ASR events?<br>
                                </div>
                              </blockquote>
                              <div><br>
                                play_and_detect_speech is a higher level
                                abstraction to simplify things.&nbsp; If you
                                want to have more control, go back to
                                using the ESL events.&nbsp; Reading the code
                                in mod_dptools and switch_ivr_async will
                                give you hints about how to do it
                                correctly.<br>
                                &nbsp;</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>
                                  &nbsp; 6. I'd like to set
                                  start-input-timers=false in the
                                  initial request then start the
                                  recognition timers
                                  (start-input-timers=true)<br>
                                  &nbsp;&nbsp;&nbsp;&nbsp; after the TTS finishes.<br>
                                  &nbsp;&nbsp;&nbsp;&nbsp; How possibly could I do this?<br>
                                </div>
                              </blockquote>
                              <div><br>
                                This is automatically done in the&nbsp;
                                switch_ivr_play_and_detect_speech()
                                function.&nbsp; You just need to specify
                                start-input-timers=false in the
                                beginning.<br>
                              </div>
                            </div>
                          </div>
                        </div>
                        <div>
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                      </blockquote>
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                  <br>
                  FreeSWITCH-powered IP PBX: The CudaTel Communication
                  Server<br>
                  <a moz-do-not-send="true"
                    href="http://www.cudatel.com" target="_blank">http://www.cudatel.com</a><br>
                  <br>
                  Official FreeSWITCH Sites<br>
                  <a moz-do-not-send="true"
                    href="http://www.freeswitch.org" target="_blank">http://www.freeswitch.org</a><br>
                  <a moz-do-not-send="true"
                    href="http://wiki.freeswitch.org" target="_blank">http://wiki.freeswitch.org</a><br>
                  <a moz-do-not-send="true"
                    href="http://www.cluecon.com" target="_blank">http://www.cluecon.com</a><br>
                  <br>
                  FreeSWITCH-users mailing list<br>
                  <a moz-do-not-send="true"
                    href="mailto:FreeSWITCH-users@lists.freeswitch.org"
                    target="_blank">FreeSWITCH-users@lists.freeswitch.org</a><br>
                  <a moz-do-not-send="true"
                    href="http://lists.freeswitch.org/mailman/listinfo/freeswitch-users"
                    target="_blank">http://lists.freeswitch.org/mailman/listinfo/freeswitch-users</a><br>
                  UNSUBSCRIBE:<a moz-do-not-send="true"
                    href="http://lists.freeswitch.org/mailman/options/freeswitch-users"
                    target="_blank">http://lists.freeswitch.org/mailman/options/freeswitch-users</a><br>
                  <a moz-do-not-send="true"
                    href="http://www.freeswitch.org" target="_blank">http://www.freeswitch.org</a><br>
                  <br>
                </blockquote>
              </div>
            </div>
          </div>
          <br>
        </blockquote>
      </div>
      <br>
      <br>
      <fieldset class="mimeAttachmentHeader"></fieldset>
      <br>
      <pre wrap="">_________________________________________________________________________
Professional FreeSWITCH Consulting Services:
<a class="moz-txt-link-abbreviated" href="mailto:consulting@freeswitch.org">consulting@freeswitch.org</a>
<a class="moz-txt-link-freetext" href="http://www.freeswitchsolutions.com">http://www.freeswitchsolutions.com</a>

FreeSWITCH-powered IP PBX: The CudaTel Communication Server
<a class="moz-txt-link-freetext" href="http://www.cudatel.com">http://www.cudatel.com</a>

Official FreeSWITCH Sites
<a class="moz-txt-link-freetext" href="http://www.freeswitch.org">http://www.freeswitch.org</a>
<a class="moz-txt-link-freetext" href="http://wiki.freeswitch.org">http://wiki.freeswitch.org</a>
<a class="moz-txt-link-freetext" href="http://www.cluecon.com">http://www.cluecon.com</a>

FreeSWITCH-users mailing list
<a class="moz-txt-link-abbreviated" href="mailto:FreeSWITCH-users@lists.freeswitch.org">FreeSWITCH-users@lists.freeswitch.org</a>
<a class="moz-txt-link-freetext" href="http://lists.freeswitch.org/mailman/listinfo/freeswitch-users">http://lists.freeswitch.org/mailman/listinfo/freeswitch-users</a>
UNSUBSCRIBE:<a class="moz-txt-link-freetext" href="http://lists.freeswitch.org/mailman/options/freeswitch-users">http://lists.freeswitch.org/mailman/options/freeswitch-users</a>
<a class="moz-txt-link-freetext" href="http://www.freeswitch.org">http://www.freeswitch.org</a>
</pre>
    </blockquote>
    <br>
  <br>
<hr>
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&nbsp; Scottish University of the Year 2011-2012</b>
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&nbsp; Heriot-Watt University is a Scottish charity<br>
&nbsp; registered under charity number SC000278.
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