> In wiki pages it is stated that The audio channels (RTP) may go directly > from phone to phone or may go through Asterisk's media bridge. > > Currently with my settings, I notice that all rtps are passing through > my asterisk. How could I achieve that they go directly from phone to > phone? I assume this way, my machine will have less load and therefore > could handle more calls.
As bkw pointed out, use canreinvite=yes for each sip phone definition. But, that will only work if the phones can reach each other directly (the phones and/or asterisk can't be behind a nat/firewall box). _______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
