<div dir="ltr">Yes, send it to the freeswitch server port 5080<div>Or if you don't need the internal profile, remove it and set external to use port 5060.<br clear="all"><div dir="ltr"><span style="font-family:Verdana,Arial,Helvetica,sans-serif"><span style="font-size:small">-Avi</span></span></div>
<br><br><div class="gmail_quote">On Sun, Dec 18, 2011 at 5:07 AM, Peter Spinato <span dir="ltr"><<a href="mailto:peter@spinato.ca">peter@spinato.ca</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<div lang="EN-US" link="blue" vlink="purple"><div><p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d">The calls go to internal profile – I guess case the OpenSIPs server is local lan – is there a way to force that calls to external profile as I find calls that hit that profile load the ext-rtp-ip ip and work.<u></u><u></u></span></p>
<p class="MsoNormal"><span style="font-size:11.0pt;font-family:"Calibri","sans-serif";color:#1f497d"><u></u> <u></u></span></p><div style="border:none;border-top:solid #b5c4df 1.0pt;padding:3.0pt 0in 0in 0in">
<p class="MsoNormal"><b><span style="font-size:10.0pt;font-family:"Tahoma","sans-serif"">From:</span></b><span style="font-size:10.0pt;font-family:"Tahoma","sans-serif""> <a href="mailto:freeswitch-users-bounces@lists.freeswitch.org" target="_blank">freeswitch-users-bounces@lists.freeswitch.org</a> [mailto:<a href="mailto:freeswitch-users-bounces@lists.freeswitch.org" target="_blank">freeswitch-users-bounces@lists.freeswitch.org</a>] <b>On Behalf Of </b>Avi Marcus<br>
<b>Sent:</b> Saturday, December 17, 2011 11:07 AM<br><b>To:</b> FreeSWITCH Users Help<br><b>Subject:</b> Re: [Freeswitch-users] OpenSIPS Load Balancer and FreeSwitch issue<u></u><u></u></span></p></div><div><div class="h5">
<p class="MsoNormal"><u></u> <u></u></p><div><p class="MsoNormal">Someone familiar with this might be able to answer right off the bat, but if you pastebin a SIP trace (and an fs_cli log for completeness) , the problem should become apparent.<u></u><u></u></p>
<div><p class="MsoNormal"><u></u> <u></u></p><div><p class="MsoNormal">Are you using the internal profile for all the calls? If you use the external.xml profile, you might need to set your ext-rtp-ip in that file, too. Seeing a trace will tell (almost) the whole story.<u></u><u></u></p>
</div><div><p class="MsoNormal"><br clear="all"><u></u><u></u></p><div><p class="MsoNormal"><span style="font-family:"Verdana","sans-serif"">-Avi</span><u></u><u></u></p></div><p class="MsoNormal" style="margin-bottom:12.0pt">
<u></u> <u></u></p><div><p class="MsoNormal">On Fri, Dec 16, 2011 at 10:46 PM, Peter Spinato <<a href="mailto:peter@spinato.ca" target="_blank">peter@spinato.ca</a>> wrote:<u></u><u></u></p><div><div><p class="MsoNormal">
All,<u></u><u></u></p><p class="MsoNormal"> Hopefully someone can assist me - I'll gladly give $50 to the person who helps me fix the issue - I have an OpenSIPs server configured as a load balancer (<a href="http://wiki.freeswitch.org/wiki/Enterprise_deployment_OpenSIPS" target="_blank">http://wiki.freeswitch.org/wiki/Enterprise_deployment_OpenSIPS</a>) that receives the call and forwards it to my Freeswitch Server for an IVR. When I had the call just routing to the Freeswitch server I got the audio working by setting the ext-rtp-ip to the public IP. Now that I route the SIP call through the OpenSIPs server there is no audio - I'm guessing it a NAT issue as always.<u></u><u></u></p>
<p class="MsoNormal"> <u></u><u></u></p><p class="MsoNormal">Both the OpenSIPS and FreeSwitch server have an internal private IP - OpenSIPS = 192.168.23.1 and FreeSwitch 192.168.23.2<u></u><u></u></p><p class="MsoNormal">
Both servers also have a public IP that routes to the internal IP OpensIPS = 47.1.1.1 and Freeswitch = 47.1.1.2<u></u><u></u></p><p class="MsoNormal"> <u></u><u></u></p><p class="MsoNormal">Call gets received by the OpenSIPs via the external IP - routes the call to the internal IP on the Freeswitch server which answers the call - but no audio - I think the FreeSwitch is trying to route the RTP Audio via its internal private IP instead of the public IP of 47.1.1.2. Not sure if this is the real issue or how to configure it route RTP properly ... all help is appreciated. $50 Paypal to whoever fixes this for me! Thanks<u></u><u></u></p>
<p class="MsoNormal"><span style="color:#888888"> <u></u><u></u></span></p><p class="MsoNormal"><span style="color:#888888">-Peter<u></u><u></u></span></p></div></div><p class="MsoNormal" style="margin-bottom:12.0pt"><br>
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<hr size="1" width="100%" noshade style="color:#a0a0a0" align="center"></div><p class="MsoNormal">No virus found in this message.<br>Checked by AVG - <a href="http://www.avg.com" target="_blank">www.avg.com</a><br>Version: 2012.0.1890 / Virus Database: 2108/4686 - Release Date: 12/17/11<u></u><u></u></p>
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