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 couldnt go out to a radio station stream for example...... On Wed, May 6, 2020 at 8:55 PM Jonathan H <lardconce...@gmail.com> wrote: > 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 <d...@nimblea.pe> wrote: > >> 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, <lardconce...@gmail.com> wrote: >> >>> 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 >>> >>> exten => radio,1,Verbose(1, Entered radio context) >>> same => n,Set(VOLUME(TX)=1) >>> same => n,WaitExten(27006,m(radio)) >>> same => n,Goto(#,1) >>> >>> It kind of works, but two problems here: >>> It's pulling data 24x7, giving the radio host artificial stats - all >>> rather needless as maybe one or two people might listen for 10 mins each in >>> a day. >>> And even though mplayer seems to stay up and running all the time, >>> sometimes Asterisk will stop listening on that pipe and everything needs a >>> restart (random, less than once a week). >>> >>> Is there a more modern/sensible way of achieving the same, just ensuring >>> that stream plays if someone listens, isn't playing when no-one is >>> listening, and listening can be exited with a specified key? >>> >>> Thanks! >>> >> -- > _____________________________________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > Check out the new Asterisk community forum at: > https://community.asterisk.org/ > > New to Asterisk? Start here: > https://wiki.asterisk.org/wiki/display/AST/Getting+Started > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users
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