Has anyone had any success in getting Gafachi to allow sip reinvites? Even though my setup is clean, with honest, internet routable addresses in all the phones, Gafachi still drops the line as soon as asterisk sends the second invite.
I really don't want asterisk in the media loop. Asterisk runs on a shared server here and I can't count on the audio rtp packet latency being low enough for good conversations.
-wolfgang