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Matt Riddell wrote: > >> On 17/10/2016, at 3:43 PM, Luca Pradovera <luca.pradovera at gmail.com >> <mailto:luca.pradovera at gmail.com>> wrote: >> >> I have been working on designs for two different projects, where both >> of them would need to use the IBM Watson streaming ASR service. >> >> Would it be possible to w...

Hello, (sorry for not continuing the thread, I had set the list to digest). Would UnicastRTP be able to output u-law frames directly? If so, I think that is all I need. Does anyone know what the EAGI output is? Raw RTP? Best regards, Luca -------------- next part -------------- An HTML attachment was scrubbed... URL:

Hello, I have been working on designs for two different projects, where both of them would need to use the IBM Watson streaming ASR service. Based on our discussion at AstriDevCon, I know there is currently no support for that. However, there may be some workarounds I am not aware of. Would it be possible to write out the audio frames as they get recorded? Watson supports 16 bit signed little

...n application call being dropped, because the leg between the PBX and the app server is terminated by the PBX following the REFER. Is there a way to configure Asterisk 11 to proxy a refer across a bridge instead of following it, so the application server can follow it instead? Best regards, Luca Pradovera -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20150427/fd82ca31/attachment.html>

...e MRCP box so it can use them during the play/ASR. Or would some form of bridging be used for the play on Asterisk while MRCP would perform only the ASR? Have a great day! Dan From: asterisk-users-bounces at lists.digium.com [mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Luca Pradovera Sent: Thursday, February 23, 2017 5:39 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Looking for Speech Recognition (ASR) suggestions Hello, you are correct. Very large grammars are only feasible with UniMRCP, and there is recognition quality to keep...

Hello, as a variation of our issues with Adhearsion calls dropping when an INVITE comes in for a bridged call, I now have a new issue to contend with. Our call is in an AsyncAGI application, and has been bridged to another channel. The provider that supplies the DID sends a polling reINVITE every 15 minutes (it's a documented Metaswitch behavior amongst others). The reINVITE is seen as a new

-- Luca Pradovera luca.pradovera at gmail.com Hello, sorry, I managed to lose the reply amidst the traffic. What we have here is our application server APP with leg A in AsyncAGI in an Adhearsion application, which after some magic dials leg B on the office PBX through a configured peer. Leg B then decides that...

Hello, I need to have an extension on a SwitchVox server dial out to one on an Asterisk (FreePBX actually) box which will host a voice directory. The Asterisk server will then need to dial one of the SwitchVox extensions if it gets a voice match. Anyone has done that, and could let me know how? So far it looks like IAX peering (what SW calls "SwitchVox peering") could work? Thanks in

Is it correct that the unimrcp is the best approach for Asterisk and ASR/TTS? Could anyone provide pros/cons for the various ASR options for Asterisk? We need the ability for very large grammars (over 100,000 options). Because of this, my initial thought is Nuance or Lumenvox. Does this sound correct? Have a great day! Dan -------------- next part -------------- An HTML attachment was