Hello,
I need direct RTP stream between 2 extensions, now Freepbx is like a proxy, everything goes through it. I tried with putting directmedia=yes in sip_general, I also put canreinvite=yes and nat=no in all the extension, also in Advanced Settings I deleted Asterisk Dial Options and put SIP nat = no and SIP canrenivite (directmedia) =yes and finaly I went to Asterisk SIP Setting and put NAT=no and can reinvite(directmedia)=yes. And it is not working, all rtp packets go through Asteris, my extensions are on local LAN. Can anyone help please