[asterisk-speech-rec] DTMF Delay CRM:00170810

Stephen Keller StephenKeller at LumenVox.com
Fri Mar 16 15:37:25 MST 2007


Coincidentally, I just had a customer contact me directly with the DTMF
issue, and I forwarded him this discussion.

He just responded with:

"Sweet - this was perfect - I went into app_speech_utils.c at line 668
and changed it to '2' instead of '5' recompiled and everything was
good!"

I haven't tried it on a test system here yet, but you may want to check
that you're setting the same value.

Stephen Keller
LumenVox Technical Support
P: 877-977-0707, just say "Stephen Keller" or "Support"
F: 858-707-7072
support at LumenVox.com
www.LumenVox.com

LumenVox

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-----Original Message-----
From: asterisk-speech-rec-bounces at lists.digium.com
[mailto:asterisk-speech-rec-bounces at lists.digium.com] On Behalf Of Scott
Higginbotham
Sent: Friday, March 16, 2007 3:09 PM
To: Use of speech recognition in Asterisk
Subject: RE: [asterisk-speech-rec] DTMF Delay

I'll give it a shot when calls die down on the platform and post my
findings.  Thanks!

Scott Higginbotham
Systems / Network Operations Manager
215.259.2185 or 1.800.835.5710 ext 2185
scotth at voicenet.com  

-----Original Message-----
From: asterisk-speech-rec-bounces at lists.digium.com
[mailto:asterisk-speech-rec-bounces at lists.digium.com] On Behalf Of jeff
quade
Sent: Friday, March 16, 2007 1:21 PM
To: asterisk-speech-rec at lists.digium.com
Subject: RE: [asterisk-speech-rec] DTMF Delay

Hey-
The culprit is most likely somewhere in the precompiled (blackbox)
"connector"-- which cant be tinkered with.

Ive noticed that it SOMETIMES takes several seconds for SpeechDestroy()
to work.

Perhaps there is an issue within the connector with killing the "speech
object" if it has NOT ACTUALLY received a valid decode from the speech
engine. This would account for moving on in the dialplan rapidly after
ASR but slowly after DTMF.

If you are going to use the speech engine "later" in the dialplan-- Id
suggest unloading the active grammars AND skipping the call to
SpeechDestroy()-- until the end of the "session."

Please post results if you try above. Thanks.

Cheers-
jjq


>From: "Scott Higginbotham" <scotth at voicenet.com>
>Reply-To: Use of speech recognition in 
>Asterisk<asterisk-speech-rec at lists.digium.com>
>To: "Use of speech recognition in
>Asterisk"<asterisk-speech-rec at lists.digium.com>
>Subject: RE: [asterisk-speech-rec] DTMF Delay
>Date: Fri, 16 Mar 2007 11:51:46 -0400
>
>Interesting.
>
>I modified the app_speech_utils.c file to reduce the DTMF timeout from
>5 seconds to 1 second - the time it takes from the time a DTMF is 
>pressed till the next appropriate action is reduced, but still taking 
>about 6-7 seconds longer than if speech is utilized to make your 
>selection instead of DTMF.
>
>Is there any other time-out settings that may be hidden with code 
>somewhere?
>
>Scott Higginbotham
>Systems / Network Operations Manager
>215.259.2185 or 1.800.835.5710 ext 2185 scotth at voicenet.com
>
>
>-----Original Message-----
>From: asterisk-speech-rec-bounces at lists.digium.com
>[mailto:asterisk-speech-rec-bounces at lists.digium.com] On Behalf Of jeff

>quade
>Sent: Thursday, March 15, 2007 1:18 PM
>To: asterisk-speech-rec at lists.digium.com
>Subject: RE: [asterisk-speech-rec] DTMF Delay
>
>Hey-
>The way dialplan command SpeechBackground(Sound File|Timeout) is 
>written there is ALWAYS a 5 second digit time-out-- AFTER a key is 
>pressed. You can see this in app_speech_utils.c at line 668.
>
>The whole app_speech_utils.c DTMF capture is a bit wonky. The speech 
>engine does NOT interpret any DTMF--  app_speech_utils.c simply creates

>a string of valid DTMF, and then populates speech->results AFTER the 5 
>second digit time-out.
>
>This method of DTMF capture BYPASSES the speech engine AND subsequently

>ALL semantic command interpretation from within the decoder-- this is 
>NOT in the docs.
>
>Cheers-
>jjq
>
>
>
>
> >From: "Scott Higginbotham" <scotth at voicenet.com>
> >Reply-To: Use of speech recognition in 
> >Asterisk<asterisk-speech-rec at lists.digium.com>
> >To: "Use of speech recognition in
> >Asterisk"<asterisk-speech-rec at lists.digium.com>
> >Subject: RE: [asterisk-speech-rec] DTMF Delay
> >Date: Thu, 15 Mar 2007 12:11:00 -0400
> >
> >Anyone have any thoughts on the delay I'm experiencing?  Is there 
> >anything I'm overlooking in the dial plan from what anyone can see?
> >
> >Scott Higginbotham
> >Systems / Network Operations Manager
> >215.259.2185 or 1.800.835.5710 ext 2185 scotth at voicenet.com
> >
> >-----Original Message-----
> >From: asterisk-speech-rec-bounces at lists.digium.com
> >[mailto:asterisk-speech-rec-bounces at lists.digium.com] On Behalf Of 
> >Scott Higginbotham
> >Sent: Sunday, March 11, 2007 8:40 PM
> >To: asterisk-speech-rec at lists.digium.com
> >Subject: [asterisk-speech-rec] DTMF Delay
> >
> >I've got an odd situation with DTMF selections being given a delay 
> >compared to the responsiveness of a voice selected menu option.
> >
> >I load both grammars, one for selecting options 1-9 via Voice 
> >selection
>
> >and one for DTMF selection so that the user can simply press 1-9 if 
> >they don't like to speak the menu option.
> >
> >Both work; however selecting the menu option with DTMF causes a good 
> >5-10 second delay before you are taken to the next context after 
> >selecting your option, whereas speaking the command is almost 
> >instantaneously taking you to the next context without the 5-10 
> >second delay.
> >
> >A snippet of my code looks like:
> >
> >exten => s,1,Playback(custom/default/welcome_0)
> >exten => s,2,SpeechCreate(lumenvox)
> >exten => s,3,SpeechActivateGrammar(numbers)
> >exten => s,4,SpeechActivateGrammar(choicetone)
> >exten => s,5,NoOp(Make it past grammar load) exten => s,n,SpeechStart

> >exten => s,n,SpeechBackground(custom/default/menu_main|3)
> >exten => s,n,NoOP(${SPEECH_TEXT(0)})
> >exten => s,n,Set(SELECTION=${SPEECH_TEXT(0)})
> >exten => s,n,SpeechDeactivateGrammar(numbers)
> >exten => s,n,SpeechDeactivateGrammar(choicetone)
> >exten => s,n,SpeechDestroy
> >exten => s,n,NoOp(Received a response of ${SELECTION}) exten => 
> >s,n,GotoIf($["${SELECTION}" = "0"]?,3,1) exten => 
> >s,n,GotoIf($["${SELECTION}" = "1"]?region-1,s,2) exten => 
> >s,n,GotoIf($["${SELECTION}" = "2"]?region-2,s,2) exten => 
> >s,n,GotoIf($["${SELECTION}" = "3"]?region-3,s,2) exten => 
> >s,n,GotoIf($["${SELECTION}" = "4"]?region-4,s,2) exten => 
> >s,n,GotoIf($["${SELECTION}" = "5"]?,3,1) exten => 
> >s,n,GotoIf($["${SELECTION}" = "6"]?,3,1) exten => 
> >s,n,GotoIf($["${SELECTION}" = "7"]?advanced,s,1) exten => 
> >s,n,GotoIf($["${SELECTION}" = "8"]?,3,1) exten => 
> >s,n,GotoIf($["${SELECTION}" = "9"]?,3,1) exten =>
> >s,n,Playback(custom/default/no_response_1)
> >exten => s,n,Wait(2)
> >exten => s,n,Goto(new-caller-intro-nresp,s,1)
> >exten => 1,1,Goto(region-1,s,2)
> >exten => 3,1,Playback(invalid)
> >exten => 3,2,Goto(new-caller-intro,s,1)
> >
> >Any thoughts?
> >
> >Scott Higginbotham
> >Systems / Network Operations Manager
> >215.259.2185 or 1.800.835.5710 ext 2185 scotth at voicenet.com 
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