|
Hi Although canreinvite option is yes,
the asterix doesn't send reinvite and the media is going through the asterix
instead of between the two sip phones. Both sip phones (handytone 486) don't use NAT and are configure with canreinvite option yes and use the same codec G.729. And Dial() command don't contains t or T. Any suggestion on
what could be the problem ? Thanks, Ishay |
_______________________________________________ --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
