Sameer Rathod wrote:
Hi,
Kia ora,
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
Remove the nat option. What does the console output show when making a call between two SIP devices?
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