On 21/04/2008, Benjamin Jacob <[EMAIL PROTECTED]> wrote: > > Hello ppl, > Any way to do a re-invite and make RTP bypass Asterisk, after call > establishment. > In other words, I would like to control when to do the bypass work for > peer-peer RTP flow. > The issue is that I need to send DTMFs after dialing the user because > most of the users are behind PBXes (having individual extensions) > themselves and almost all of the PBXes send a 200 OK and then play out the > PBX messages. > So I need to send the extension DTMFs first, bridge the calls and then > re-invite users for them to do a peer-peer rtp conversation. > > TiA, > - Ben.
Is there an echo? ;-) I answered this an hour ago. Regards, Steve _______________________________________________ -- 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
