----- Original Message ----- > From: "Markus" <[email protected]> > To: "Asterisk Users Mailing List - Non-Commercial Discussion" > <[email protected]> > Sent: Sunday, August 26, 2012 6:43:31 PM > Subject: [asterisk-users] One leg in a conference and adjusting stream > volume of other leg > > Hi all, >
<snip> > A SIP caller dials into to my Asterisk 10. He will automatically > listen > to a specific MP3 stream. As you're using Asterisk 10, I'm going to assume you're using ConfBridge. <snip> > When they adjust the volume of the stream, if effects only their > stream, > and not the volume of the stream of the other callers. > > In short: All callers at all times are *always* in the same > conference, > but each caller is able to increase or decrease the volume of "their" > MP3 stream individually. You can use ConfBridge's DTMF menus to allow a participant to change their listening volume. This should only affect the audio that the participant hears, and not the other participants in the conference - regardless of whether or not the audio originates from the same source. [bridge_user_menu] *1=increase_listening_volume 1=increase_listening_volume *2=decrease_listening_volume 2=decrease_listening_volume For more information on setting up DTMF menus and associating them with bridge users, see the ConfBridge article on the Asterisk wiki: https://wiki.asterisk.org/wiki/display/AST/ConfBridge+10 > If I'm right the MP3 stream cannot come from inside conference > (MeetMe > or ConfBridge with MOH) because there is no functionality to control > the > volume individually. <snip> No, that's fine. ConfBridge allows you to control the listening/speaking volume of each participant. See above. <snip> > > I don't know where to start. Queue? Local channel? ... > If you wanted to stream your sound from a source other than MOH, using a Local channel may be appropriate. I'm not sure how a Queue would help here. -- Matthew Jordan Digium, Inc. | Engineering Manager 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at: http://digium.com & http://asterisk.org -- _____________________________________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
