Have a look here: http://www.voip-info.org/wiki-Asterisk+config+features.conf .Specifically at applicationfaturemap.
----- Original Message ----- From: "Atis Lezdins" <[EMAIL PROTECTED]> To: <[email protected]> Sent: Monday, October 08, 2007 2:10 PM Subject: Re: [asterisk-users] Injecting a sound file into a bridged call > On 08/10/2007, Girts Graudins <[EMAIL PROTECTED]> wrote: >> > I'm looking for a way to play a sound file to an already established >> > bridged call. It is meant for one party, but it's ok if both parties >> > would hear it. Ideally, I'd like to be able to trigger this from the >> > Management Interface with something like: > > I'm also in need for such functionality, the only difference is that i > need > for both channels to hear the message. As i have read press releases, > there > will be something similar available in 1.6. If you succeed, please give us > a > note - how it can be done. > >> > 2) I've seen "whisper"-type of functionality associated with >> > meetme >> > rooms, but I'd rather not set up a dynamic meetme room for each call >> > I'm >> > bridging; > > Well, you can create conference dynamically whenever you need to play the > file. I started working on this, and have found several bugs regarding > this, > but they should be fixed in 1.4.12 > > Idea is to Redirect() trough AMI both channels to dynamical conference, > and > then attach call with Playback() to the same conference. For now, the > Redirect() part is working fine, but due to lack of time, i haven't got > further. > > On Monday 08 October 2007 14:13:38 Jaswinder Singh wrote: >> See chanspy in asterisk 1.4 , it also has a whisper mode and you can talk >> to one party http://www.voip-info.org/wiki/view/Asterisk+cmd+ChanSpy . >> But >> i dont know how to play a recorded file in it . > > My collegues tried this but unsuccessfully. The basic idea is to use local > channels - one is bridged to Chanspy() and second to Playback(). I'm not > sure > what is the problem, but theoretically also this should work. > > Regards, > Atis > > -- > Atis Lezdins > VoIP Developer, > IQ Labs Inc. > [EMAIL PROTECTED] > Skype: atis.lezdins > Cell Phone: +371 28806004 > Work phone: +1 800 7502835 > > _______________________________________________ > --Bandwidth and Colocation Provided by http://www.api-digital.com-- > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users > _______________________________________________ --Bandwidth and Colocation Provided by http://www.api-digital.com-- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
