its fairly painless now
https://github.com/nimbleape/asterisk-dialogflow-rtp-audioserver
https://github.com/nimbleape/asterisk-dialogflow-ari-bridge
Theres 2 repos - one for the ari bridge - 1:1 call -> external media and
another for talking to dialogflow but theres no reason that
Thanks Dan - might have to scratch my head over that one for a while!
The phrase "you make your own RTP server" has made me all twitchy ;)
Jonathan
On Wed, 6 May 2020 at 07:21, Dan Jenkins wrote:
> Hi Jonathan,
>
> I'd probably go down the external media route in the ARI now - you make
> your
Hi Jonathan,
I'd probably go down the external media route in the ARI now - you make
your own RTP server and provide your own RTP back to asterisk
On Sun, 3 May 2020, 13:07 Jonathan H, wrote:
> Way back in 2016 the only way to allow callers to listen in to a stream
> "at will" was to do the
Way back in 2016 the only way to allow callers to listen in to a stream "at
will" was to do the following:
moh.conf
[radio]
mode=custom
application=/usr/bin/mplayer https://example.com/stream.mp3 -quiet -ao
pcm:file=/dev/stdout -af volume=5,resample=8000,channels=1,format=alaw
extensions.conf
