On Fri, 22 Feb 2008 18:50:16 +0800, Ron <[EMAIL PROTECTED]> wrote: >If i set, canreinvite=yes on all ext, assuming all ip phones have the >same codec, if 100 calls 101, or vice versa will rtp still go thru >asterisk? and same scenario for 200 to 202 or vice versa.
... and I'd like to add to this question: If the phones have the option "Enable NAT", I expected them to be able to talk to each other directly, but they didn't, and I had to set them to "canreinvite=no" in sip.conf. Why is that? _______________________________________________ -- 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
