Hi, I am new to asterisk I want to configure my asterisk server such that it only establishes the call rest the audio must bypass the server and transmitted directly to the peer
In my config file I did changes which are below canreinvite=yes nat=force_rtp dirtectmedia=yes directsetup=yes I am using asterisk version 12.3 -- Regards Sameer Rathod 8109413462
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