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 _______________________________________________ Asterisk-Users mailing list [EMAIL PROTECTED] http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
