[asterisk-users] Simple, fast single-word offline free speech recognition in Asterisk (or as an AGI)?
I'm after fast, native recognition of the numbers 1 to 20, yes, no, menu and help. At the moment, I use Google Speech Recognition which uses no local processing power, and is very accurate, 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. I was hoping I could use "local", with a fallback to Google speech rec if it was uncertain. Any ideas? Thanks Yes, I know I post similar back in January, but there was no response back then and I was hoping things might have changed :) On Wed, 16 Jan 2019 at 17:42, Jonathan H wrote: > 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 > -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] multiple softphone clients and same/different account credentials
(I'm new to Asterisk, after having started VOIP with vat on the mbone in the 90s.) I am setting up my first Asterisk system, and trying to read docs/guidance and follow best practices. I have read the 5th Edition of "Asterisk: The Definitive Guide" and like the 3rd Edition on the web it recommends that hardphones and softphones both have a unique name distinct from any concept of extension. http://asteriskdocs.org/en/3rd_Edition/asterisk-book-html-chunk/DeviceConfig_id283201.html Then the 5th edition goes on to give an example with a hardphone and a softphone associated with one individual, where the hardphone is named by MAC address and the softphone by JIM_VANM_SOFT (p. 61). Despite talking about separating extensions, phone names, and people, it seems clear that a softphone is usually personal to a person (unless it's a desk phone via a computer, but I'm talking about the personal type). THe book does not address the notion that a user might be given credentials and then configure them on a number of softphone-type devices simultaneously, e.g. a smartphone, a tablet, and two laptops. When getting service from an ITSP, it seems there are credentials and they don't want to know the details of how many softphones you are using. So which option is preferred? A) Have a softphone aor/auth_user/password for a particular human, and expect them to configure it on multiple devices. Do not worry that 1) multiple are registered at once (because that's normal in SIP) and 2) asterisk has no idea which is which (because the intent is to place a call to that person) B) issue credentials per device and keep them all separate. Use extensions.conf to ring them all Having written the question out carefully, it seems obvious that A is the way to do this, but it's sort of contrary to the advice in the book so I thought I would ask. Thanks, Greg -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] bug in pjsip trust_id_outpound?
Hi Team I'm still struggling to get privacy settings passed on correctly. The Asterisk is sitting between customers and IC trunks. On the customer face, of course I have those settings: trust_id_inbound=yes trust_id_outbound=no This ensures that presentation is set to probibited, if the customer is setting Privacy: ID. It also ensures that the From: header is set to anonymous, hiding the callerID if the caller requested presentations prohib. So far, towards the customer side, this works as expected. Towards the IC, we need to correctly populate the Request, From, P-Asserted Identity and Privacy header. Sending From: anonymous is not allowed. So I set: trust_id_inbound=no trust_id_outbound=yes Unfortunately I have to set inbound trust to no, to make sure the Asterisk takes callerID from the From: header and NOT from the P-Asserted Identity Header. I then call pull the Privacy: Header with PJSIP_HEADER and set caller presentation correctly. But with outbound calls I am facing a HUGE problem. I have set trust_id_outbound=yes. So I am expecting Asterisk to correctly set the From: header to CallerID(number) and if CallerID(num-pres) ist set to prohib, to add a Privacy: ID header. This is not happening. From: is set to anonymous with missing Privacy: header with the result, that the IC partner on the other side is blocking this call. Any idea how I could deal with this? Why is trust_id_outbound=yes not behaving as expected? Mit freundlichen Grüssen -Benoît Panizzon- -- I m p r o W a r e A G-Leiter Commerce Kunden __ Zurlindenstrasse 29 Tel +41 61 826 93 00 CH-4133 PrattelnFax +41 61 826 93 01 Schweiz Web http://www.imp.ch __ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
