search for: digipoint

Hi all, i'm tring ro use sip with an external sip proxy as vocal or ser. My scenario is Vocal or SER ----> Asterisk with cnah_oh323 -----> Gatekeeper I would like that sip termial register themself to Vocal or ser and the h.323 terminal to gatekeeper. When i place a call from h323 side to sip side all work.... When a try to place a call form sip to h323 nothing happen Does

Hi all, i' using asterisk with chan_oh323 to test the SIP/H.323 interoperability. All working well... Now i would like to know if is it possible that * doesn't control the RTP traffic. I would tht the end point control the RTP flow... Any idea, is it possible??? Thanks Marco