[asterisk-dev] Speech Recognition Problems
pbx.kumar at gmail.com
Thu May 15 00:11:34 CDT 2008
-- I posted this in speech-rec forum but I see that its pretty inactive.
Kudos to the asterisk community.
I am trying to integrate nuance speech recognition and microsoft speech
server with asterisk and was partly successful. The present solution records
voice with silence detection set to 2 seconds and then sends the recorded
file to the speech servers with the voice grammar and then acts on the
result. Everything is done in agi. We use cajo for load distribution.
- the present solution has a big drawback. The users have to wait till the
end of the prompt before they can start recording. They cannot speak in
between and this is causing a problem. Is there a way to stop playback of
prompt when the speech is detected from the callee end.
- I see that speech api is available and connectors can be written but there
is no proper documentation. Can we check how lumenvox has done the
connector? Since its GPL licensed - I am assuming it should be shared.
- backgrounddetect does stop during play but it jumps to talk extension. we
want the entire speech to be recorded and leaving out the first fragment
which triggered to jump to talk extension will not server the purpose in
speech detection. further, since its agi , its not extension driven.
Any pointers will be appreciated. Thanks.
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