I want to be able to capture both parties speech to text continuously in a call and send those strings off to be translated in real-time and then use twiml.say to speak the text back. I have not been having much luck with this and wondering how I should go about doing this.
The one user will make a call from their phone to the other support person which is at a web browser. I have the call setup and working fine, however I cannot find any documentation anywhere that is aligned with what I am wanting to do and wondering if it is possible or if I need to be looking down a different route.
Should anyone have any advice or has seen samples similar to this I would love to see them. Thanks!
Twilio developer evangelist here.
It's not currently possible to capture a two legged conversation with <Gather> and speech recognition. So you might need to look somewhere else for this functionality.
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I want to create a real-time audio-calling application with nodejs. It will have a feature like Omegle (randomly) but only audio calling. A user also should be able to call people who he/she talked to before.
I found that Twillo and other sites have services like this but I want it to be free. Could you please suggest to me what to use and how to implement it? Any links or videos? Thanks!
I want to test how Google Home transforms vocal commands to text by sending voice commands and storing the result returned. I have already done the storage part, but now, I can't find in the documentation how to send automatically voice commands to google home, the only apparent way is to speak to it directly, which not very practical if you want to test a long list of commands, 50 times for each command!
Edited: To make it clearer, I want to write a function that sends voice files (mp3, or any other format) to google assistant, instead of having to say/pronounce the command in a human way.
Do you know if it is possible to make this process automatic ?
It sounds like you might want the Assistant SDK, which will let you send an audio stream/file or text to be processed by the Assistant and return the result.
You're unclear about exactly what you're trying to do, and how you're trying to do it, but this table should help you understand what features are available for the various methods of using the Assistant SDK. In general, you'll be able to send an audio file or stream (either using the python library or a gRPC library for your language of choice) and get a response back.
I have a google home speaker, and I can issue commands like what's the time or play some music, but I'd like to be able to define my own responses to certain commands, like
how many appointments do I have today
or
are there any cancellations
I would like the above commands to a run a script where I can either run a web-service, or pull information from my SmartThings hub (that bit is optional) and respond with an appropriate response.
I've done a bit of research, and it seems that IFTTT, can do something similar, but I don't really want to be dependent on a 3rd party app, and if this can be done directly with Google.
I guess I'm looking for something similar to Groovy for SmartThings, where I can write Smart Apps.
The API to develop your own commands is known as Actions on Google. Broadly speaking, Actions will send JSON to a webhook that you control, and you can have it do whatever you wish at that point.
As of now (is using api.ai) what I see is, I get the string format of what the user speaks.
I would like to access the raw audio file the user speaks to interact with Google Assistant using the api.ai platform.
Is there a way to get the audio file ?
[UPDATE]:
We are aiming to evaluate the quality of speech of the user hence we would need to run the algorithms on the audio.
No, there is currently no way to get the audio content of what has been sent.
(However, the team is looking to understand the use cases of why you might want or need this feature, so you may want to elaborate on your question further.)
I am a college student and a complete newbie to asterisk.
I'm currently working on a project 'email to voice call'.
Using python i'v extracted the email & converted it into speech and saved in a WAV file.
Now using asterisk i want to generate call to the mobile of the user through my system.
I have read the book 'Asterisk: The Future Of Telephony' as suggested by many. But i'm still not able to understand what all things i need to setup to generate a call to mobile.
What i understood is that i need to configure two files i.e. sip.conf where i need to give the details of VoIP provider and extensions.conf for dial-plan. Asterisk will tell the VoIP provider to generate a call.
Now can anyone please tell me what things i need to setup other than these? Also can you help me in the configuration of these two files??
Please help. Any information will be appreciated.
Thank You.
I am not fully understand your case.
You want to generate call on user's mobile (cellular telephony)?
Or
You want to generate call on user's mobile voip application?
If you case is second then you only need to configure 2 files as you have mentioned. And you can easily get pre-configured files from internet (voip-info.org).
But if you fall in to case one then you need following apart from those 2 files.
Additional hardware FXO card.
Need to configure zaptel driver for that FXO card.
One telephone line.