[Freeswitch-users] Where i can get help with freeswitch?

[patrick]

I use dialplan and this extensions:

<extension name="my_ivr_01">
<condition field="destination_number" expression="^777$">
<action application="answer"/>

<action application="set" data="tts_engine=unimrcp:vn-mrcp-v1"/>
<action application="set" data="tts_voice=Anna8000"/>
<action application="speak" data="Here is some text for synthesis"/>

<action application="detect_speech" data="unimrcp 
http://192.168.22.116:8080/vxml/tstgrammar.xml vn-mrcp-v1"/>
</condition>
</extension>


<extension name="my_ivr_02">
<condition field="destination_number" expression="^778$">
<action application="answer"/>

<action application="set" data="playback_terminators=#"/>
<action application="playback" data="ponce-preludio-in-e-major.wav"/>

<action application="detect_speech" data="unimrcp 
http://192.168.22.116:8080/vxml/tstgrammar.xml vn-mrcp-v1"/>
</condition>
</extension>



And I need to barge-in and start "detect_speech" in both extensions, 
when synthesis or playback is going on.

P.S. Thank you for answer!


Christopher Rienzo пишет:
> Barge-in will work out of the box for digits... to make it work for 
> ASR is a bit more complicated.
>
> I don't know what method you are using to do TTS, but it this is the 
> general idea:
>
> 1. set up a handler to deal with input callbacks
> 2. on DTMF or start of speech, return "break" to cause barge-in.
>
> My help can be more specific if you tell me more about what method 
> (dialplan, Lua, javascript, etc) you are using to execute TTS and ASR.
>
> Can't help you on the noise issue... someone else needs to chime in.
>
>
>
> On Fri, Apr 23, 2010 at 5:41 AM, patrick <patrick at speechpro.com 
> <mailto:patrick at speechpro.com>> wrote:
>
>     Hello from St.Petersburg!
>     My name is Patrick.
>     I try to realise IVR with ASR & TTS.
>     Platform win32. Soft: Freeswith, Unimrcp mod (client), and some local
>     product "Voicenavigator" (mrcp server, ASR, TTS).
>
>     I have 2 problems:
>
>     1. How to realise "barge in" for playback and TTS? Does freeswitch
>     allow
>     that?
>     I need to start playback, or TTS and break it when some speech is
>     detected (by ASR)...
>
>     2. When I call from Asterisk to Freeswitch and bridge my call back to
>     Asterisk (to another number), there only noise in first asterisk
>     abonent's phone...
>
>     I would be grateful for any help ;-)
>
>     --
>     С уважением,
>
>     Равальдини Патрик Стефанович
>     Инженер по тестированию
>     ООО «Центр речевых технологий»
>     Тел.: (812) 325-8848, доб. 6225
>     Факс: (812) 327-9297
>     E-mail: patrick at speechpro.com <mailto:patrick at speechpro.com>
>     http://www.speechpro.ru
>
>
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>
>
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>   

-- 
С уважением,

Равальдини Патрик Стефанович
Инженер по тестированию
ООО «Центр речевых технологий»
Тел.: (812) 325-8848, доб. 6225
Факс: (812) 327-9297
E-mail: patrick at speechpro.com
http://www.speechpro.ru