[Asterisk-Users] Autorizing SIP to SIP calls across oh323

[M. Willigs]

Hi everybody.

With te module oh323 SIP to SIP calls can be autorized in the same * server
in an h323 gatekeeper. I need to do it this way because all the original
system has been built under h323, so, it's easy to integrate the Asterisk
with the rest of the system this way.

My question is: is there a way to force a native bridge between both SIP
terminals in order to avoid the RTP trafic across the Asterisk?

Thanks in advance.
M. Willigs