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. _______________________________________________ -- 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
