Rob a écrit : > Hi all, > Hi > I'm using GIZMO with my asterisk (1.4.13) box ... I've had CALL IN for a > while and it works fine .... I just added CALL OUT ... I have no problem > with call setup ... the called party hears me ... but I can't hear them .... > again if the call comes INTO the server both sides work fine. > > Looks like a nat issue: do you have nat=yes and canreinvite=no in your sip.conf for Gizmo5?
-- Daniel _______________________________________________ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2009 - October 13 - 15 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