thanks for replying but no change at all any other tips,suggestions thanks in advance
On Wed, 9 Mar 2005 01:44:41 -0600, Jay Milk <[EMAIL PROTECTED]> wrote: > 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
