[Freeswitch-users] Webrtc Verto jsapi

Steve Mathers stevem at singlecomm.com
Wed Jun 1 23:59:18 MSD 2016


Thanks, I did see this. But not what i'm looking for. I still want to be
able to send commands to the backend. In my case I want to broadcast a
specific event to a specific channel.

On Wed, Jun 1, 2016 at 1:46 PM, Ítalo Rossi <italo at freeswitch.org> wrote:

> http://evoluxbr.github.io/verto-docs/tut/holding-a-call.html
>
>
> > On 1 de jun de 2016, at 13:09, stevem <stevem at singlecomm.com> wrote:
> >
> > I was not able to find any documentation on this. Basically all I want
> to do
> > from my frontend. Is post an event to the backend. Basically run the
> > command: uuid_broadcast 69d58330-7f7f-4e52-bb2c-9e25d9f5238b
> > event::Event-Name=HOLD.
> >
> > I'm using FreeSWITCH Version 1.4.21~64bit
> >
> > I got the verto/webrtc working using the demo located here:
> > https://webrtc.freeswitch.org/verto/index.html#page-main
> >
> > I'm able to connect to the dialplan and drop the caller (PSTN/ITSPs) and
> > verto client into a conference and the audio is flowing. I just want to
> send
> > a message to the backend to tell it to do things. In my example above I
> want
> > to set the caller on hold.
> >
> > My verto connection in javascript looks like this:
> >
> >    $.verto.init({}, bootstrap);
> >
> >    function bootstrap(status) {
> >      that.vertoHandle = new jQuery.verto({
> >        login: userId + '@' + host,
> >        passwd: password,
> >        tag: "vertoAudio",
> >        socketUrl: 'wss://' + host + ':8082',
> >        audioParams: {
> >          googAutoGainControl: false,
> >          googNoiseSuppression: false,
> >          googHighpassFilter: false
> >        },
> >        iceServers: true
> >      }, vertoCallbacks);
> >    }
> >
> >    var vertoCallbacks = {
> >      onWSLogin: onWSLogin,
> >      onWSClose: onWSClose,
> >      onDialogState: onDialogState
> >    };
> >
> >
> > On successful login i make a call:
> >
> >     that.call= that.vertoHandle.newCall({
> >          destination_number: that.liveCall.ani,
> >          caller_id_name: user.firstName + ' ' + user.lastName,
> >          caller_id_number: userId,
> >          useVideo: false,
> >          useStereo: false,
> >          userVariables: {}
> >        });
> >
> >
> > Thanks alot. This is my first post and I've only been doing Freeswitch
> for a
> > few months. I have alot of experience in Asterisk though. Let me know if
> > more data is needed.
> >
> >
> >
> > --
> > View this message in context:
> http://freeswitch-users.2379917.n2.nabble.com/Webrtc-Verto-jsapi-tp7596220.html
> > Sent from the freeswitch-users mailing list archive at Nabble.com.
> >
> > _________________________________________________________________________
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> >
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> Ítalo Rossi
> italo at freeswitch.org
>
>
>
>
> _________________________________________________________________________
> Professional FreeSWITCH Consulting Services:
> consulting at freeswitch.org
> http://www.freeswitchsolutions.com
>
> Official FreeSWITCH Sites
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>
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