[Asterisk-Users] i am missing something!

[Adnan Ahmed]

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 <jay at skimmilk.net> 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:asteriskster at gmail.com]
> > Sent: Wednesday, March 09, 2005 1:14 AM
> > To: asterisk-users at lists.digium.com
> > 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:
> 
>