[asterisk-users] packet2packet bridging

[Sameer Rathod]

= Using SIP RTP CoS mark 5
    -- Executing [1061 at 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 at digium.com> 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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