[Asterisk-Users] Asterisk + GrandStream SIP phones

[pesb]

Hi,
Thanks for the help. You were correct. There was some data missing in the 
extension.conf file
I was able to call one SIP phone from the other. I was even able to call an 
H323 IP phone registered to the gnugk GK (It has Asterisk registered to him 
as a GW).
But, I have another problem rigth now.
All the RTP Data Flow is passing through the Asterisk Proxy, which is a bad 
thing if I want to have many SIP phones in my system.
How can I configure the SIP phone in order to make all RTP data flow directly 
from one SIP phone to the other? 
And, how can I configure it in order to make all RTP data flow directly from 
one SIP phone to the H323 IP phone (the one registered to my gnugk GK)?
I would also like to be able to make calls from a SIP phone to the other SIP 
phone, but instead of having the ASTERISK PBX authorizing the calls, it would 
be the H323 GK the one that would authorize calls. How can I do this?

Thanks again