Hi all,
i would like to know if it is possible to bridging the rtp traffic over Asterisk...
I would like that the RTP flow is not controlled by * but by the endpoint.
Is it possible??? Any suggestion to do this???
Thanks
Marco
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