Re: packet2packet bridging

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= Using SIP RTP CoS mark 5
    -- Executing [1061@sameer:1] Dial("SIP/1060-00000088", "SIP/1061") in new stack
  == Using SIP RTP CoS mark 5
    -- Called SIP/1061
    -- SIP/1061-00000089 is ringing
       > 0x7f67f90b43c0 -- Probation passed - setting RTP source address to 192.168.1.176:8000
    -- SIP/1061-00000089 answered SIP/1060-00000088
    -- Channel SIP/1061-00000089 joined 'simple_bridge' basic-bridge <1937a88f-5b6f-4a5a-ae3c-adc2d68c8bb4>
    -- Channel SIP/1060-00000088 joined 'simple_bridge' basic-bridge <1937a88f-5b6f-4a5a-ae3c-adc2d68c8bb4>
       > Bridge 1937a88f-5b6f-4a5a-ae3c-adc2d68c8bb4: switching from simple_bridge technology to native_rtp
       > 0x7f67f90b43c0 -- Probation passed - setting RTP source address to 192.168.1.176:8000
       > 0x7f6780047090 -- Probation passed - setting RTP source address to 192.168.1.191:8000
  == WebSocket connection from '192.168.1.191:54390' closed


It is giving me following output on asterisk console


On Wed, Jul 2, 2014 at 6:30 PM, Joshua Colp <jcolp@xxxxxxxxxx> wrote:
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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--
Regards
Sameer Rathod
8109413462 

-- 
_____________________________________________________________________
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