[Asterisk-Users] How to make RTP Packets NOT passing thru Asterisk?

[Asterisk .]

--- Senthil Murugan <zenmurugan at HotPOP.com> 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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