On Jan 5, 2008 4:40 AM, ameel <akamboh at connectageek.com> 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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