You need to export rtp_secure_media=true<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:&#39;Roboto&#39;;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&#39;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 &quot;allow SRTP&quot; 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&#39;s an example of what&#39;s happening:</p>
<p style="margin-top:0px;margin-bottom:0px;margin-left:0px;margin-right:0px;text-indent:0px">Extension 100 -- SRTP --&gt; FreeSWITCH -- RTP --&gt; 102 (call success)</p>
<p style="margin-top:0px;margin-bottom:0px;margin-left:0px;margin-right:0px;text-indent:0px">Extension 100 -- RTP --&gt; FreeSWITCH  -- SRTP --&gt; 102 (call success)</p>
<p style="margin-top:0px;margin-bottom:0px;margin-left:0px;margin-right:0px;text-indent:0px">Extension 100 -- SRTP --&gt; FreeSWITCH -- SRTP --&gt; 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&#39;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&#39;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:&#39;liberation mono&#39;">Iskren Hadzhinedev</span></p></div></blockquote><br><br>-- <br>Sent from mobile device<br>