<div dir="ltr">Yeah, try kristian's suggestion first. At some point that variable changed. This happened to me last week and turned out my colleagues checked out the 1.5 branch instead of 1.2stable. </div><div class="gmail_extra">
<br><br><div class="gmail_quote">On Fri, Jan 10, 2014 at 7:14 PM, Kristian Kielhofner <span dir="ltr"><<a href="mailto:kris@kriskinc.com" target="_blank">kris@kriskinc.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
You need to export rtp_secure_media=true<div class="HOEnZb"><div class="h5"><span></span><br><br>On Friday, January 10, 2014, Iskren Hadzhinedev wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<u></u>
<div style="font-family:'Roboto';font-size:9pt;font-weight:200;font-style:normal">
<p style="margin-top:0px;margin-bottom:0px;margin-left:0px;margin-right:0px;text-indent:0px">Hello, everyone!</p>
<p style="margin-top:0px;margin-bottom:0px;margin-left:0px;margin-right:0px;text-indent:0px">I'm having trouble with SRTP calls between two internal extensions.</p>
<p style="margin-top:0px;margin-bottom:0px;margin-left:0px;margin-right:0px;text-indent:0px">Whenever I configure the phones to require SRTP calls fail and the freeswitch console shows:</p>
<p style="margin-top:0px;margin-bottom:0px;margin-left:0px;margin-right:0px;text-indent:0px"> [INFO] mod_dptools.c:3201 Originate Failed. Cause: INCOMPATIBLE_DESTINATION</p>
<p style="margin-top:0px;margin-bottom:0px;margin-left:0px;margin-right:0px;text-indent:0px">If I select "allow SRTP" so that encryption is not mandatory I get encryption on a single leg in a call.</p>
<p style="margin-top:0px;margin-bottom:0px;margin-left:0px;margin-right:0px;text-indent:0px">Here's an example of what's happening:</p>
<p style="margin-top:0px;margin-bottom:0px;margin-left:0px;margin-right:0px;text-indent:0px">Extension 100 -- SRTP --> FreeSWITCH -- RTP --> 102 (call success)</p>
<p style="margin-top:0px;margin-bottom:0px;margin-left:0px;margin-right:0px;text-indent:0px">Extension 100 -- RTP --> FreeSWITCH -- SRTP --> 102 (call success)</p>
<p style="margin-top:0px;margin-bottom:0px;margin-left:0px;margin-right:0px;text-indent:0px">Extension 100 -- SRTP --> FreeSWITCH -- SRTP --> 102 (call fail: INCOMPATIBLE_DESTINATION)</p>
<p style="margin-top:0px;margin-bottom:0px;margin-left:0px;margin-right:0px;text-indent:0px"> </p>
<p style="margin-top:0px;margin-bottom:0px;margin-left:0px;margin-right:0px;text-indent:0px">I've setup the sip_secure_media=true variable as instructed from </p>
<p style="margin-top:0px;margin-bottom:0px;margin-left:0px;margin-right:0px;text-indent:0px"><a href="https://wiki.freeswitch.org/wiki/Secure_RTP" target="_blank">https://wiki.freeswitch.org/wiki/Secure_RTP</a></p>
<p style="margin-top:0px;margin-bottom:0px;margin-left:0px;margin-right:0px;text-indent:0px">but it's still not working. Any help is greatly appreciated!</p>
<p style="margin-top:0px;margin-bottom:0px;margin-left:0px;margin-right:0px;text-indent:0px"> </p>
<p style="margin-top:0px;margin-bottom:0px;margin-left:0px;margin-right:0px;text-indent:0px">Kind regards,</p>
<p style="margin-top:0px;margin-bottom:0px;margin-left:0px;margin-right:0px;text-indent:0px">-- </p>
<p style="margin-top:0px;margin-bottom:0px;margin-left:0px;margin-right:0px;text-indent:0px"><span style="font-family:'liberation mono'">Iskren Hadzhinedev</span></p></div></blockquote><br><br></div></div><span class="HOEnZb"><font color="#888888">-- <br>
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