Barge-in will work out of the box for digits... to make it work for ASR is a bit more complicated.<br><br>I don't know what method you are using to do TTS, but it this is the general idea:<br><br> 1. set up a handler to deal with input callbacks<br>
2. on DTMF or start of speech, return "break" to cause barge-in.<br><br>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.<br>
<br>Can't help you on the noise issue... someone else needs to chime in.<br><br><br><br><div class="gmail_quote">On Fri, Apr 23, 2010 at 5:41 AM, patrick <span dir="ltr"><<a href="mailto:patrick@speechpro.com">patrick@speechpro.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="border-left: 1px solid rgb(204, 204, 204); margin: 0pt 0pt 0pt 0.8ex; padding-left: 1ex;">Hello from St.Petersburg!<br>
My name is Patrick.<br>
I try to realise IVR with ASR & TTS.<br>
Platform win32. Soft: Freeswith, Unimrcp mod (client), and some local<br>
product "Voicenavigator" (mrcp server, ASR, TTS).<br>
<br>
I have 2 problems:<br>
<br>
1. How to realise "barge in" for playback and TTS? Does freeswitch allow<br>
that?<br>
I need to start playback, or TTS and break it when some speech is<br>
detected (by ASR)...<br>
<br>
2. When I call from Asterisk to Freeswitch and bridge my call back to<br>
Asterisk (to another number), there only noise in first asterisk<br>
abonent's phone...<br>
<br>
I would be grateful for any help ;-)<br>
<br>
--<br>
С уважением,<br>
<br>
Равальдини Патрик Стефанович<br>
Инженер по тестированию<br>
ООО «Центр речевых технологий»<br>
Тел.: (812) 325-8848, доб. 6225<br>
Факс: (812) 327-9297<br>
E-mail: <a href="mailto:patrick@speechpro.com">patrick@speechpro.com</a><br>
<a href="http://www.speechpro.ru" target="_blank">http://www.speechpro.ru</a><br>
<br>
<br>
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</blockquote></div><br>