I have both t and T options in dial command. SIP phones configured with canreinvite=no and when I pres #1 (as I have in features.conf) during call there nothing to happened.
Thank for any suggestions. Bob. _______________________________________________ --Bandwidth and Colocation sponsored by Easynews.com -- 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
