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

[Christopher Rienzo]

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> 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
> http://www.speechpro.ru
>
>
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