[asterisk-users] Speech/DTMF mix?

[Danny Nicholas]

All input (DTMF and voice) processed by SpeechBackground are returned to the
dialplan in SPEECH_TEXT(0).  The grammar controls how speech is returned.
When you press DTMF instead of speaking you typically get a speech score of
999 or 1000 (a quick and dirty way of knowing that you pressed one instead
of saying 1).  
For question 2, here's how I do it (Vestec since you are using that)
exten => s,1,SpeechCreate(Vestec ASR Engine)
exten => s,n,Set(usedtmf=${ERROR})
exten => s,n,Set(SPEECH_ERRORS=0)
exten => s,n,Gotoif($["${usedtmf}" = "1"]?novr)
exten => s,n,Set(THRESHOLD=710)
exten => s,n,Set(VPVAL=voicegood)
exten => s,n,Goto(start,s,1)
exten => s,n(novr),Set(VPVAL=voicebad)
exten => s,n,Goto(start,s,1)

When processing for speech or DTMF, I check ${usedtmf} before trying speech
function, if it is 1 (error) I use read or Waitexten instead of
SpeechBackground.
 
(for anybody using Lumenvox, here's the same snippet)
exten => 372,1,SpeechCreate()
exten => 372,n,Set(usedtmf=${ERROR})
exten => 372,n,Gotoif($["${usedtmf}" = "1"]?nolumena)
exten => 372,n,Set(VPVAL=voicegood)
exten => 372,n,Goto(start|s|1)
exten => 372,n(nolumena),Set(VPVAL=voicebad)
exten => 372,n,Goto(start|s|1)

-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Richard Kenner
Sent: Tuesday, May 11, 2010 8:25 AM
To: asterisk-users at lists.digium.com
Subject: Re: [asterisk-users] Speech/DTMF mix?

> Here is a snippet from my lumenvox dialplan (works pretty much the same
for
> Vestec)

Thanks for the confirmation and sample.

Sorry to be dense, but you're saying that the DTMF comes back in
SPEECH_TEXT(0)?  What about SPEECH_SCORE in that case?  And what's the
exact difference with Vestec since I'll be using it?

Also, one (somewhat unrelated) question: what happens when I hit the
port limit?  Where do I detect that in the dialplan, so I can revert
to dtmf-only?

-- 
_____________________________________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
New to Asterisk? Join us for a live introductory webinar every Thurs:
               http://www.asterisk.org/hello

asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
   http://lists.digium.com/mailman/listinfo/asterisk-users