I am building an ostn node. I followed a recipe, but got caught up with learning FS basics. I had this trouble that calls were being signalled, but no audio was actually put through.<br>I thought it was nat problems, but connectivity was going fine. <br>
This is my dialplan snippet:<br> &lt;extension name=&quot;Local_Extension&quot;&gt;<br>      &lt;condition field=&quot;destination_number&quot; expression=&quot;^(\d{3})$&quot;&gt;<br>        &lt;action application=&quot;set&quot; data=&quot;dialed_extension=$1&quot;/&gt;<br>
        &lt;action application=&quot;export&quot; data=&quot;dialed_extension=$1&quot;/&gt;<br>       &lt;!-- &lt;bind_meta_app can have these args &lt;key&gt; [a|b|ab] [a|b|o|s] &lt;app&gt; --&gt;<br>        &lt;action application=&quot;bind_meta_app&quot; data=&quot;1 b s execute_extension::dx XML features&quot;/&gt;<br>
&lt;!--        &lt;action application=&quot;bind_meta_app&quot; data=&quot;2 b s record_session::$${recordings_dir}/${caller_id_number}.${strftime(%Y-%m-%d-%H-%M-%S)}.wav&quot;/&gt; --&gt;<br>        &lt;action application=&quot;bind_meta_app&quot; data=&quot;3 b s execute_extension::cf XML features&quot;/&gt;<br>
        &lt;action application=&quot;set&quot; data=&quot;ringback=${us-ring}&quot;/&gt;<br>        &lt;action application=&quot;set&quot; data=&quot;transfer_ringback=$${hold_music}&quot;/&gt;<br>        &lt;action application=&quot;set&quot; data=&quot;call_timeout=30&quot;/&gt;<br>
        &lt;!-- &lt;action application=&quot;set&quot; data=&quot;sip_exclude_contact=${network_addr}&quot;/&gt; --&gt;<br>        &lt;action application=&quot;set&quot; data=&quot;hangup_after_bridge=true&quot;/&gt;<br>        &lt;action application=&quot;set&quot; data=&quot;continue_on_fail=NORMAL_TEMPORARY_FAILURE,USER_BUSY,NO_ANSWER,TIMEOUT,NO_ROUTE_DESTINATION&quot;/&gt;<br>
        &lt;action application=&quot;hash&quot; data=&quot;insert/${domain_name}-call_return/${dialed_extension}/${caller_id_number}&quot;/&gt;<br>        &lt;action application=&quot;hash&quot; data=&quot;insert/${domain_name}-last_dial_ext/${dialed_extension}/${uuid}&quot;/&gt;<br>
        &lt;action application=&quot;set&quot; data=&quot;called_party_callgroup=${user_data(${dialed_extension}@${domain_name} var callgroup)}&quot;/&gt;<br>        &lt;action application=&quot;hash&quot; data=&quot;insert/${domain_name}-last_dial/${called_party_callgroup}/${uuid}&quot;/&gt;<br>
&lt;!--       &lt;action application=&quot;answer&quot;/&gt;  --&gt;<br>        &lt;action application=&quot;info&quot;/&gt;<br>         &lt;action application=&quot;answer&quot;/&gt;<br>        &lt;action application=&quot;bridge&quot; data=&quot;user/${dialed_extension}@${domain_name}&quot;/&gt;<br>
    &lt;!--    &lt;action application=&quot;answer&quot;/&gt; --&gt;<br>        &lt;action application=&quot;sleep&quot; data=&quot;1000&quot;/&gt;<br>        &lt;action application=&quot;voicemail&quot; data=&quot;default ${domain_name} ${dialed_extension}&quot;/&gt;<br>
      &lt;/condition&gt;<br>    &lt;/extension&gt;<br><br>This works, but you can see i have an answer app right before the bridge. If i don&#39;t answer the call, i don&#39;t get to the point where i get Activating SRTP RECV/SRTP SEND. Yet both channels get answered and RTP goes both ways, but i have a feeling it&#39;s not SRTP and thus is not being decrypted.<br>
I have this in the global context:<br> &lt;condition field=&quot;${sip_has_crypto}&quot; expression=&quot;^(AES_CM_128_HMAC_SHA1_80)$&quot; break=&quot;never&quot;&gt;<br>        &lt;action application=&quot;set&quot; data=&quot;sip_secure_media=true&quot;/&gt;<br>
        &lt;!-- Offer SRTP on outbound legs if we have it on inbound. --&gt;<br>        &lt;action application=&quot;export&quot; data=&quot;sip_secure_media=true&quot;/&gt;<br>      &lt;/condition&gt;<br><br> which should activate SRTP on both legs.<br>
<br>I have pasted in to the bin a debug log with and without the answer and media sending here :<br><a href="http://pastebin.freeswitch.org/19937">http://pastebin.freeswitch.org/19937</a><br>