Make sure you have canreinvite=no in your sip peers definition, and/or that you pass 't' or 'T', to the Dial statement.
Julian J. M. On 4/25/05, Tim Pushor <[EMAIL PROTECTED]> wrote: > Hi all, > > I am still unable to initiate a call transfer with the keypresses > defined in features.conf in a couple month old version of asterisk from > CVS HEAD. _______________________________________________ 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
