On Wed, 20 Aug 2003, Jamie Carl wrote:
> > Seeing as no one else has replied, I figured I may give it > a shot. At least it'll start something. > > Now, correct me if I'm wrong someone, but as far as I > understand in this situation you can do both. Normally > the RTP packets would be swtiched through *, but you can > set in you sip.conf file the 'canreinvite=yes' option > which will allow the RTP stream to be direct if a > compatible codec is negotiated. > > I'll double check if I ever get my server up and running > again. > > J > > On Tue, 19 Aug 2003 11:17:20 -0500 > "Jorge Cisneros Flores" <[EMAIL PROTECTED]> wrote: > >Hi > > > > > > Is posible to make a call from site A to Site C, and > >my question is, the rtp data is from A to C or is from A > >to B to C > > > > > > > > > > Site A Site B > > Site C > > ata186<-------->FW<--------->Asterisk<--------->FW<----------->ata186 > > > >Thanks > > Regards, > > Jamie Carl > Jazz Inc. > Email: [EMAIL PROTECTED] > Web: www.jazz-inc.net > Phone: +61-414-365-466 > Jabber: [EMAIL PROTECTED] > _______________________________________________ > Asterisk-Users mailing list > [EMAIL PROTECTED] > http://lists.digium.com/mailman/listinfo/asterisk-users > _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users
