<div dir="ltr"><div><div>= Using SIP RTP CoS mark 5<br> -- Executing [1061@sameer:1] Dial("SIP/1060-00000088", "SIP/1061") in new stack<br> == Using SIP RTP CoS mark 5<br> -- Called SIP/1061<br> -- SIP/1061-00000089 is ringing<br>
> 0x7f67f90b43c0 -- Probation passed - setting RTP source address to <a href="http://192.168.1.176:8000">192.168.1.176:8000</a><br> -- SIP/1061-00000089 answered SIP/1060-00000088<br> -- Channel SIP/1061-00000089 joined 'simple_bridge' basic-bridge <1937a88f-5b6f-4a5a-ae3c-adc2d68c8bb4><br>
-- Channel SIP/1060-00000088 joined 'simple_bridge' basic-bridge <1937a88f-5b6f-4a5a-ae3c-adc2d68c8bb4><br> > Bridge 1937a88f-5b6f-4a5a-ae3c-adc2d68c8bb4: switching from simple_bridge technology to native_rtp<br>
> 0x7f67f90b43c0 -- Probation passed - setting RTP source address to <a href="http://192.168.1.176:8000">192.168.1.176:8000</a><br> > 0x7f6780047090 -- Probation passed - setting RTP source address to <a href="http://192.168.1.191:8000">192.168.1.191:8000</a><br>
== WebSocket connection from '<a href="http://192.168.1.191:54390">192.168.1.191:54390</a>' closed<br><br><br></div>It is giving me following output on asterisk console<br></div></div><div class="gmail_extra"><br>
<br><div class="gmail_quote">On Wed, Jul 2, 2014 at 6:30 PM, Joshua Colp <span dir="ltr"><<a href="mailto:jcolp@digium.com" target="_blank">jcolp@digium.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
Sameer Rathod wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
Hi,<br>
</blockquote>
<br>
Kia ora,<div class=""><br>
<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
I am new to asterisk I want to configure my asterisk server such that it<br>
only establishes the call<br>
rest the audio must bypass the server and transmitted directly to the peer<br>
<br>
In my config file I did changes which are below<br>
<br>
canreinvite=yes<br>
nat=force_rtp<br>
dirtectmedia=yes<br>
directsetup=yes<br>
<br>
I am using asterisk version 12.3<br>
</blockquote>
<br></div>
Remove the nat option. What does the console output show when making a call between two SIP devices?<span class="HOEnZb"><font color="#888888"><br>
<br>
-- <br>
Joshua Colp<br>
Digium, Inc. | Senior Software Developer<br>
445 Jan Davis Drive NW - Huntsville, AL 35806 - US<br>
Check us out at: <a href="http://www.digium.com" target="_blank">www.digium.com</a> & <a href="http://www.asterisk.org" target="_blank">www.asterisk.org</a><br>
<br>
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</font></span></blockquote></div><br><br clear="all"><br>-- <br><div>Regards</div>Sameer Rathod<div>8109413462 <br><div><br></div></div>
</div>