On Jan 5, 2008 4:40 AM, ameel <[EMAIL PROTECTED]> wrote: > I am trying to setup asterisk as a registrar and sip server only. > Currently When I make calls all my rtp traffic is going through the > asterisk server as a B2BUA. > Is it possible to turn off this feature and have all my calls RTP traffic > going directly to the SIP UA? > > ______
Hi Try SER or OpenSEr ram
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