[Asterisk-Users] rtp channels not through asterisk

[Rich Adamson]

> In wiki pages it is stated that The audio channels (RTP) may go directly 
> from phone to phone or may go through Asterisk's media bridge.
> 
> Currently with my settings, I notice that all rtps are passing through
>  my asterisk. How could I achieve that they go directly from phone to
> phone?  I assume this way, my machine will have less load and therefore 
> could handle more calls.

As bkw pointed out, use canreinvite=yes for each sip phone definition.
But, that will only work if the phones can reach each other directly
(the phones and/or asterisk can't be behind a nat/firewall box).