I am trying to configure a direct rtp media stream between SIP devices using canreinvite=yes and it is working fine when both SIP devices are connected to the same Asterisk server.
The problem is that we have several Asterisk server in our branch offices and if I try make a call between SIP devices that are connected to different Asterisk servers then the rtp media stream is always via the Asterisk servers.
The Asterisk servers are connected via IAX2 trunks.
Is it possible to implement direct rtp media stream between SIP devices that are connected to different Asterisk servers ?
Gardar Georg Nielsen.