Hello, we're trying to use the attended and blind transfer feature with Asterisk (version 1.0.7) but have failed every time we've tried. We have tested every possible configuration (i.e. we have the T and t options in the Dial() function, we have "canreinvite=no" in sip.conf, etc), and now have no clue about how to proceed. Can you help us? What may be happening?
Thanks, Ma. Eugenia.- _______________________________________________ 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
