I am using canreinvite=no on all my sip.conf settings and also U use t on my Dials... No luck so far.
-----Original Message----- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Julian J. M. Sent: Domingo, 27 de Febrero de 2005 01:24 p.m. To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [Asterisk-Users] Limit the call & recording when pressing *1 You need to tell asterisk to stay in the media path. You must add 't' to the Dial options: exten => 21,1,Dial(${phone1},20,trwL(300000:240000:60000)) Or set canreinvite=no in your sip peer definition. Julian J. M. > but even adding it and commenting out "automon => *1" didn't work. > and of course I restart asterisk after modifying features.conf _______________________________________________ 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 _______________________________________________ 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
