[Asterisk-Dev] <<< SPHINX INTEGRATION >>>

[Storer, Darren]

Hi,

I was beginning to think that no one was interested in speech recognition
for Asterisk; having recently asked for a "must have" and a wish list in the
asterisk-users mailing list, the silence was deafening.

Tests this week have shown that speech recognition over PRI via Asterisk is
much more accurate than via VoIP (even when compared to a 64k skinny
connection from a Cisco VIP 30+). A rough estimate is a degradation of 20%
(loss of accuracy) when good quality VoIP is used instead of ISDN.

Work is progressing nicely on a commercial speech rec. add-on for Asterisk
but the finer details of API support, size of vocabulary etc. have not yet
been thrashed out. If anyone in the dev list has a "must have" and a "nice
to have" features list please shout up.

Regards

Darren
--
Comgate
Telco>Internet<Broadcast

-----Original Message-----
From: asterisk-dev-admin at lists.digium.com
[mailto:asterisk-dev-admin at lists.digium.com]On Behalf Of Eric Wieling
Sent: 10 April 2004 23:48
To: asterisk-dev at lists.digium.com
Subject: Re: [Asterisk-Dev] <<< SPHINX INTEGRATION >>>


jeff quade wrote:
> Howdie:
> Thought Id just post to see if anyone has had any success with
> integrating CMUs SPHINX into Asterisk?
> I see the EAGI in the source/build, and a couple of posts to
> Asterisk-Users, but no useable info.
> Seems like a DEAD issue, waiting for a re-visit.
> Before I pull the code apart and struggle with deciphering the source--
> I thought someone might have pointers or suggestions.

Someone, I don't remember who, on IRC #Asterisk got it working, but only
with ULAW/ALAW codec.  Apparently the sound quality on the compressed
codec was not quite good enough for Sphinx.  I don't know if he ever
resolved the problem.  I imagine you could tune Sphinx to work better
with compressed codecs.
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