[asterisk-users] ConfBridge: Identifying troublemakers

2019-01-16 Thread Markus

Hi list,

imagine a ConfBridge conference with 10 participants. Now, one of them 
suddenly starts to yell and scream.


Is there any built-in functionality (maybe not in ConfBridge, but in 
Asterisk itself) to identify the "loudest" caller?


Or maybe already built-in functionality to automatically lower his 
speech volume in a case where he starts to scream, respectively adjust 
it to the level of the other participants?


How about a scenario where that troublemaker does not scream, but 
instead starts to insult the other participants. He's not loud, just rude.


I imagine a solution where a ConfBridge moderator/admin has to press a 
DTMF key and the ConfBridge will offer him a menu of that particular 
person who's talking at that very second. E.g...


Caller: "You're all stupid! I don't like y..."
- Moderator presses "5" in that very second
ConfBridge talks only to Moderator:
"Welcome to the moderator menu. The person who was talking in that very 
second is on channel SIP/something-123456

Please press 1 to kick him out.
Please press 2 to ban him.
Please press 3 to return to the conference."

Something like that...

Thanks for any pointers & tips! :)

Regards
Markus

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[asterisk-users] Simple one-word offline free speech recognition in Asterisk (or as an AGI)?

2019-01-16 Thread Jonathan H
When I last looked into this a couple of years ago, simple one-word speech
recognition was rather complex and slow.

At the moment, I use Google Speech Recognition which uses no local
processing power, and is very accurate and fast, allowing me to run on a
very low end VPS.

However, with the minimum of 15 seconds, numbers and words like "yes, no"
soon eat up the 60 minute free allowance.

Have things changed much in the last couple of years? I see a couple of new
"standalone" projects even from the likes of Facebook and Mozilla, but they
require a degree in C++ and, apparently, about 24 hours to build a voice
model on a high-end box with the latest graphics cards (for the number
crunching). Also, unless I'm reading it wrong, each second of speech takes
4 seconds to recognise on a low end machine with this standalone offerings
and similar ones.

https://github.com/facebookresearch/wav2letter
https://voice.mozilla.org/en

In fact, come to think of it, I really only need offline fast recognition
of numbers 1 to 20, yes, no, menu and help.
For voicemail transcription I'm happy to stick with Google's paid service
as it's remarkably accurate with phone quality speech (beats Microsoft and
Amazon Transcribe hands down from what I can tell).

Oh, and UniMRPC seems rather complex and the licensing doesn't suit - 99%
of the time I have one channel (caller) but it can jump to 10 - I don't
want to have to buy a 10 channel license for that 1 hour a month!

Any ideas? Thanks
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[asterisk-users] CDR/CEL Radius features

2019-01-16 Thread Olivier
Hello,

Page [1] gathers information on how to configure Asterisk CDR Radius
backend.
I'm not familiar at all with Radius in IP Telephony.

1. Would a Radius database and its associated tools allow live call
accounting data displaying of an Asterisk instance powered by such CDR
Radius backend ? Would it scale to several Asteruisk instances ?

2. I know Freeradius SimultaneousUse setting.
Could this be used to control the total call count of a given accountcode
when these calls are processed by several Asterisk boxes ?

Best regards

[1] https://wiki.asterisk.org/wiki/display/AST/RADIUS+CDR+Backend
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