Open juansalvatella opened 5 years ago
The original app is still in the repo, and it uses DTMF instead of AudioSocket. You can find it here.
audiosocket is really something of a proof of concept right now. We built it for a specific purpose and customer, with an eye toward the more general need to getting streaming audio into and out of Asterisk. I am working on a Görtzel implementation in Go for another project, which is the algorithm Asterisk (and most telephony systems) uses for DTMF detection, but ultimately, I want to implement audiosocket in a channel interface or similar, which will allow us to use it fairly seamlessly with ARI, among other things (like getting DTMF detection).
Amazing! If there is any way that I can contribute to that please let me know.
On Mon, 12 Nov 2018 at 19:48, Seán C. McCord notifications@github.com wrote:
The original app is still in the repo, and it uses DTMF instead of AudioSocket. You can find it here https://github.com/CyCoreSystems/asterisk-k8s-demo/tree/master/live-demo/app .
audiosocket is really something of a proof of concept right now. We built it for a specific purpose and customer, with an eye toward the more general need to getting streaming audio into and out of Asterisk. I am working on a Görtzel https://en.wikipedia.org/wiki/Goertzel_algorithm implementation in Go for another project, which is the algorithm Asterisk (and most telephony systems) uses for DTMF detection, but ultimately, I want to implement audiosocket in a channel interface or similar, which will allow us to use it fairly seamlessly with ARI, among other things (like getting DTMF detection).
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Thanks; I'll update this issue when I have more information on either.
I added DTMF signalling. I just started using GITHUB, please come and see it. By Google Translate. :D https://github.com/rainbow-cnay/audiosocket
Thanks @rainbow-cnay
Would you mind submitting it as a Pull Request?
Congratulations on this amazing demo! It's truly inspiring. I have been wondering on how this could be improved and thought that in case that the Google Speech Recognition does not work properly because I have a speech impediment or it's just too noisy, it would be useful to use the keyboard to state how many Asterisk replicas I want.
What would be the best approach to handle this? Maybe some logic should be pushed to Asterisk in order to determine when it should send the audio to the audiosocket or when to wait for DTMF. Looking forward to know your opinion on this.