SIP wrote: > When calling from our SIP proxy through Asterisk to the PSTN provider, > we support reINVITES which tend to work seamlessly. > > However, when creating a call file which essentially connects a call > from the SIP proxy to the SIP proxy, Asterisk wants to stay in the RTP > media path. I understand that this is sort of the idea behind a bridged > channel, but is there any way to avoid it? Is there any way to say > "Connect this number and this number and then get out of the way," or > is this a design limitation? > > N. > >
No ideas on this one? I've tried everything I can think of and then some and still can't get Asterisk out of the media path. I can do it if I don't originate the call with Asterisk, but only use Asterisk to connect one leg of the call, but if I use Asterisk to connect both legs, no luck. Going about this the wrong way? N. _______________________________________________ -- 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
