Hi guys, I am trying to configure an asterisk server for our office. Asterisk 1.4.17 SIP only
The problem appears when the call comes from external point to our internal network. So when the server receives the call the channel is answered and the remote user hears prompt which invite him to enter internal private number. After that the server starts to wait the extension. After timeout the server executes Dial application and sends invite to sip client from our internal network. The problem is in this point. I want to play ringback tone to remote user when he waits internal user to pick up his phone but I could not instruct Asterisk to generate fake ringback in rtp stream . Is there a solution for this? Thanks in advance. -- Best Regards eng. Anatoli Marinov _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 Phoenix, Arizona Register Now: http://www.astricon.net asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
