sip-ip and rtp-ip should be the ip address not "domain"<br><br>-BDF<br><br><div class="gmail_quote">2012/4/18 Sharad Garg <span dir="ltr"><<a href="mailto:sharad@coraltele.com">sharad@coraltele.com</a>></span><br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><u></u>
<div bgcolor="#ffffff">
<div><u></u><font face="Verdana">Hi All </font></div>
<div><font face="Verdana"></font> </div>
<div><font face="Verdana">I am trying the FS with network bonding. My
configuration is as follows -</font></div>
<div><font face="Verdana"></font> </div>
<div><font face="Verdana">eth0 - No IP
assigned<br>eth1 - No IP
assigned</font></div><div class="im">
<div><font face="Verdana">bond0 - 192.168.4.65
This is the virtual IP which is to access the
freeswitch.<br></font></div>
</div><div><font face="Verdana">Earlier when I was throwing the call to FS on
192.168.4.65, call was not arriving at FS. But later on with the help of some
kind people of this forum, I got some suggestions & I defined the
"domain=192.168.4.65" & in internal profile, I have defined sip-ip=domain
& rtp-ip=domain. After this, FS is attending the call. </font></div>
<div><font face="Verdana"></font> </div>
<div><font face="Verdana">Now the problem is voice is one way. >From FS to
Network, voice is OK but Network to FS there is no voice. </font></div>
<div><font face="Verdana"></font> </div>
<div><font face="Verdana">I request the forum to help me on this issue
also.</font></div>
<div><font face="Verdana"></font> </div>
<div><font face="Verdana">Best Regards</font></div>
<div><font face="Verdana">Sharad</font></div></div>
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<br></blockquote></div><br><br clear="all"><br>-- <br>Brian D. Foster<br>Endigo Computer LLC<br>Email: <a href="mailto:bdfoster@endigotech.com" target="_blank">bdfoster@endigotech.com</a><br>Phone: 317-800-7876<br>Indianapolis, Indiana, USA<br>
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