Hello!<br><br>I got a problem with one way audio, symptoms are:<br>firstly play audio file to channel A (A is hears sound)<br>secondly bridge channel B with A (A doesn&#39;t hear B).<br><br>Environment:<br>- no NAT<br>- User Agents being used X-Lite, EyeBeam, SJphone - same result for all of them- no audio, Wireshark shows that there is no RTP-flow to A from FreeSwitch<br>
- dialplan:<br>&lt;extension name=&quot;playback_media_file&quot;&gt;<br>    &lt;condition field=&quot;destination_number&quot; expression=&quot;playmedia&quot;&gt;<br>      &lt;action application=&quot;answer&quot;/&gt;<br>
      &lt;action application=&quot;playback&quot; data=&quot;test.wav&quot;/&gt;<br>    &lt;/condition&gt;<br>  &lt;/extension&gt;<br><br>&lt;extension name=&quot;Local_Extension_from_SP&quot;&gt;<br>      &lt;condition field=&quot;destination_number&quot; expression=&quot;^([0-9]{2,9})$&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=$1&quot;/&gt;<br>      &lt;/condition&gt;<br>      &lt;condition field=&quot;${sip_to_host}&quot; expression=&quot;^([^.]*)\..*$&quot;&gt;<br>
        &lt;action application=&quot;set&quot; data=&quot;orgname=$1&quot;/&gt;<br>      &lt;/condition&gt;<br>      &lt;condition field=&quot;destination_number&quot; expression=&quot;^${caller_id_number}$&quot;&gt;<br>        &lt;anti-action application=&quot;set&quot; data=&quot;ringback=${us-ring}&quot;/&gt;<br>
        &lt;anti-action application=&quot;set&quot; data=&quot;transfer_ringback=${us-ring}&quot;/&gt;<br>        &lt;anti-action application=&quot;set&quot; data=&quot;call_timeout=10&quot;/&gt;<br>        &lt;anti-action application=&quot;set&quot; data=&quot;hangup_after_bridge=true&quot;/&gt;<br>
       
&lt;anti-action application=&quot;set&quot;
data=&quot;continue_on_fail=NORMAL_TEMPORARY_FAILURE,USER_BUSY,NO_ANSWER,TIMEOUT,NO_ROUTE_DESTINATION,USER_NOT_REGISTERED&quot;/&gt;
<br>        &lt;anti-action application=&quot;set&quot; data=&quot;continue_on_fail=true&quot;/&gt;<br>        &lt;anti-action application=&quot;db&quot; data=&quot;insert/${domain_name}-call_return/${dialed_extension}/${caller_id_number}&quot;/&gt;<br>
        &lt;anti-action application=&quot;db&quot; data=&quot;insert/${domain_name}-last_dial_ext/${dialed_extension}/${uuid}&quot;/&gt;<br>       
&lt;anti-action application=&quot;set&quot;
data=&quot;called_party_callgroup=${user_data(${dialed_extension}@${domain_name}
var callgroup)}&quot;/&gt;<br>        &lt;anti-action application=&quot;db&quot; data=&quot;insert/${domain_name}-last_dial/${called_party_callgroup}/${uuid}&quot;/&gt;<br>        &lt;anti-action application=&quot;bridge&quot; data=&quot;user/${dialed_extension}@${domain_name}&quot;/&gt;<br>
        &lt;anti-action application=&quot;answer&quot;/&gt;<br>       
&lt;anti-action application=&quot;export&quot;
data=&quot;sip_h_X-SPFrom=&amp;quote;${sip_from_user}&amp;quote;&amp;lt;${sip_from_uri}&amp;gt;&quot;/&gt;<br>        &lt;anti-action application=&quot;export&quot; data=&quot;sip_h_X-SPTo=&amp;lt;${sip_to_uri}&amp;gt;&quot;/&gt;<br>
        &lt;anti-action application=&quot;export&quot; data=&quot;sip_h_X-SPCallId=${sip_call_id}&quot;/&gt;<br>        &lt;anti-action application=&quot;bridge&quot; data=&quot;sofia/external/${orgname}send2voicemail@$${starpound_sip_app_server}&quot;/&gt;<br>
      &lt;/condition&gt;<br>    &lt;/extension&gt; <br>- Call routing scheme:<br>user calls to FS, FS calls to my B2BUA which manage call with SIP 3pcc<br>Exact description what&#39;s going on is:<br>user A -&gt; FS -(bridge)-&gt; my B2BUA<br>
Then my B2BUA transfers (using re-INVITE&#39;s), on behalf of user, call to extension &quot;playback_media_file&quot; . After a while B2BUA transfer (re-Inviting) user to extension &quot;Local_Extension_from_SP&quot;. This should create a new call to user B. As a result - A doesn&#39;t hear B, but B- is OK.<br>
On the contrary, if a call is routed (by B2BUA) to the &quot;Local_Extension_from_SP&quot; extension (ommiting &quot;playback_media_file&quot; ext) - everything is OK.<br><br><br>What I&#39;ve tried:<br>- set parameter &quot;inbound-proxy-media&quot; to &quot;true&quot; in Sofia profile<br>
- set parameter &quot;disable_rtp_auto_adjust to &quot;true&quot; in Sofia profile<br>Nothing helps.<br><br><br>Any help or thoughts would be MUCH appreciated!<br>Artem<br><br>