Sameer Rathod wrote:
Hi,
Kia ora,Remove the nat option. What does the console output show when making a call between two SIP devices?
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
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Joshua Colp
Digium, Inc. | Senior Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - US
Check us out at: www.digium.com & www.asterisk.org
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