asterisk users - Jul 2014 - packet2packet bridging

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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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