Hi all! 1 for sip to sip call and h323 to h323 adn h323 to sip, can asterixk proxy call with signal only ( no rtp go through yate)?
if yes, how to configure for this? thanks in advance best regards! _______________________________________________ --Bandwidth and Colocation sponsored by Easynews.com -- Asterisk-Users mailing list [email protected] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
