--- Senthil Murugan <[EMAIL PROTECTED]> wrote:
> Hello All,
>
> Currently my setup uses Xlite and Asterisk and i found that all the RTP
> voice packets are transfered via the asterisk server from one xlite to
> another. Is there any possibility that we can make all the RTP Packets to be
> transfered directly between the two clients once the connection is
> established?.
Do you have a dial statement with T, or t on it? If so, remove them. Also add
canreinvite=yes to
the peer configuration in your sip.conf.
Girish
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