Hi, I can not get this working: Asterisk on public IP. Two SIP phones behind NAT - in the same LAN.
I works perfectly (two way sound) when each peer (friend) can not reinvite - audio stream goes through Asterisk. The problem pops up when I define canreinvite=yes on each peer definision so I suppose to stream audio directly between phones (in the same local LAN). Right after called party answers, Asterisk sends new INVITE's to each phone pointing in DSP messages that audio should be sent to ip:port of each phone. So, phone A is sending RTP to gateways public ip ad port of phone B - all this fails, response from gateway is 'Destination unreachable (Port unreachable)? Why is it so? In 'no reinvite' scenario Asterisk communicates whit each phone without any problems so why phone can not send rtp to another port and Asterisk can? Is it possible to get this working at all? Lukasz. _______________________________________________ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
