[asterisk-users] packet2packet bridging

Joshua Colp jcolp at digium.com
Wed Jul 2 08:00:59 CDT 2014

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?

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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