Somebody correct me if I'm wrong here, but without reinvite being disabled, I don't think the * can inject audio on the middle of the call.
-----Original Message----- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Nabeel Jafferali Sent: Wednesday, May 04, 2005 11:41 PM To: 'Asterisk Users Mailing List - Non-Commercial Discussion' Subject: RE: [Asterisk-Users] MOH > First off, yes, canreinvite=no would be a good choice. Well, I am in a situation where my * server is hosted and it is quite pointless to have all media going through the * server when two SIP devices are talking. > Secondly, did you "make mpg123" from the asterisk source directory? Yes, tried that. It said "mpg123 is up to date". -- Nabeel Jafferali X2 Networks www.x2n.ca T: 1.647.722.6900 1.877.VOIP.X2N F: 1.866.655.6698 FWD: 46990 _______________________________________________ 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
