You'll need canreinvite=no to each sip section in sip.conf, if you want * to stay in the loop.
> -----Original Message----- > From: Adnan Ahmed [mailto:[EMAIL PROTECTED] > Sent: Wednesday, March 09, 2005 1:14 AM > To: [email protected] > Subject: [Asterisk-Users] i am missing something! > > > Hello ppl, > At initial level i configure asterisk woth only soft phones > ,in which one at windows machine and other is linux i am > using windows messenger and linphone respectively both phones > registered with asterisk respectively problem is that they > bypass asterisk on call when i send request from linphone to > messenger request shown on messenger but on asterisk console > nothing to and also if i send request from messenger to > linphone it doesn't recognized at all my config are: _______________________________________________ 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
