[asterisk-dev] Audio to/from Asterisk

[marek]

is there someone who can write/share small HOWTO test it with 
https://cloud.google.com/speech-to-text/ ?

Dne 01/08/2019 v 16:54 George Joseph napsal(a):
>
>
> On Thu, Aug 1, 2019 at 7:36 AM Joshua C. Colp <jcolp at digium.com 
> <mailto:jcolp at digium.com>> wrote:
>
>     On Thu, Aug 1, 2019, at 10:28 AM, George Joseph wrote:
>     > So here's where we're at with adding this capability...
>     >
>     > Initial release:
>     >  * Two new ARI endpoints, one on channel and one on bridge:
>     >    * /channels/<channel_id>/externalMedia
>     >    * /bridges/<bridge_id>/externalMedia
>
>     What do these return? How do you stop external media at a future time?
>
>
> They'd return an ExternalMedia object which would contain an ID along 
> with other pertinent data that can be gleaned from the underlying 
> provider.  For chan_rtp, it could be the local IP address and local 
> port.  To stop the streaming, you'd make a DELETE  request on the 
> ExternalMedia resource.
>
> This is similar to how we do Playback and Record today.
>
>
>
>     -- 
>     Joshua C. Colp
>     Digium - A Sangoma Company | Senior Software Developer
>     445 Jan Davis Drive NW - Huntsville, AL 35806 - US
>     Check us out at: www.digium.com <http://www.digium.com> &
>     www.asterisk.org <http://www.asterisk.org>
>
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>
> -- 
> *George Joseph*
> Digium - A Sangoma Company | Software Developer | Software Engineering
> 445 Jan Davis Drive NW - Huntsville, AL 35806 - US
> direct/fax: +1 256 428 6012
> Check us out at: https://digium.com <https://digium.com/> · 
> https://sangoma.com <https://sangoma.com/>
>
>
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