This could be a dialplan issue. You will need to supply a console debug log. Use <a href="http://pastebin.freeswitch.org">pastebin.freeswitch.org</a><br><br>-MC<br><br><div class="gmail_quote">On Tue, Feb 7, 2012 at 1:47 PM, Sherif Omran <span dir="ltr">&lt;<a href="mailto:sherifomran2000@yahoo.com">sherifomran2000@yahoo.com</a>&gt;</span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><table border="0" cellpadding="0" cellspacing="0"><tbody><tr><td style="font:inherit" valign="top">Ok, now i get the following <br>
<br> [NO_ROUTE_DESTINATION]<br><br><br>could you please help?<div><div class="h5"><br><br>thank you<br><br><br><br>--- On <b>Tue, 2/7/12, <a href="mailto:rsaavedra@ecogizmos.com" target="_blank">rsaavedra@ecogizmos.com</a> <i>&lt;<a href="mailto:rsaavedra@ecogizmos.com" target="_blank">rsaavedra@ecogizmos.com</a>&gt;</i></b> wrote:<br>
<blockquote style="border-left:2px solid rgb(16,16,255);margin-left:5px;padding-left:5px"><br>From: <a href="mailto:rsaavedra@ecogizmos.com" target="_blank">rsaavedra@ecogizmos.com</a> &lt;<a href="mailto:rsaavedra@ecogizmos.com" target="_blank">rsaavedra@ecogizmos.com</a>&gt;<br>
Subject: Re: [Freeswitch-users] rtmp help<br>To: &quot;FreeSWITCH Users Help&quot; &lt;<a href="mailto:freeswitch-users@lists.freeswitch.org" target="_blank">freeswitch-users@lists.freeswitch.org</a>&gt;<br>Date: Tuesday, February 7, 2012, 10:18 PM<br>
<br><div>Hello,<br>you have to permit anonymous calls on the external profile.<br><br>Bye,<br><br>Ricardo Saavedra<br><br><br>&gt; Hello all,<br>&gt;<br>&gt; Any body knows how to adjust the rtmp?<br>&gt;<br>&gt; When i try to place a call and give 1000@serverip, I can see the call
 in<br>&gt; fs_cli. However, it says<br>&gt;<br>&gt; freeswitch@internal&gt; 2012-02-07 19:01:32.972040 [DEBUG] rtmp.c:869<br>&gt; [chunk_stream=3 type=0x14 ts=12247591 stream_id=0x0] len=58<br>&gt; 2012-02-07 19:01:32.972040 [DEBUG] rtmp.c:170 [amfnumber=3] Got INVOKE for<br>
&gt; makeCall<br>&gt; 2012-02-07 19:01:32.972040 [WARNING] rtmp_sig.c:296 Unauthorized call to<br>&gt; sip:<a href="http://mc/compose?to=1000@serverip.com" target="_blank">1000@serverip.com</a>, client is not logged in.<br>
&gt;<br>&gt; How can i make the client log in?<br>&gt; I have the following in the autoconfig/rtmp.conf.xml.<br>&gt;<br>&gt;<br>&gt; &lt;configuration name=&quot;rtmp.conf&quot; description=&quot;RTMP Endpoint&quot;&gt;<br>
&gt;   &lt;profiles&gt;<br>&gt;     &lt;profile name=&quot;default&quot;&gt;<br>&gt;         &lt;settings&gt;<br>&gt;             &lt;param name=&quot;bind-address&quot;
 value=&quot;<a href="http://0.0.0.0:1935" target="_blank">0.0.0.0:1935</a>&quot; /&gt;<br>&gt;             &lt;param name=&quot;context&quot; value=&quot;sipinterface_1&quot; /&gt;<br>&gt;             &lt;param name=&quot;dialplan&quot; value=&quot;XML&quot; /&gt;<br>
&gt;             &lt;!-- If this is set to true, no unauthenticated inbound calls<br>&gt; will be allowed --&gt;<br>&gt;             &lt;param name=&quot;auth-calls&quot; value=&quot;false&quot; /&gt;<br>&gt;             &lt;!-- How much time should the clients buffer the media stream<br>
&gt; (miliseconds) --&gt;<br>&gt;             &lt;param name=&quot;buffer-len&quot; value=&quot;50&quot; /&gt;<br>&gt;             &lt;!-- Sets the maximum size of outbound RTMP chunks --&gt;<br>&gt;    
         &lt;param name=&quot;chunksize&quot; value=&quot;512&quot; /&gt;<br>&gt;         &lt;/settings&gt;<br>&gt;     &lt;/profile&gt;<br>&gt;   &lt;/profiles&gt;<br>&gt; &lt;/configuration&gt;<br>&gt;<br>&gt; any help is appreciated<br>
&gt;<br>&gt; thank you<br>&gt;<br>&gt; regards,<br>&gt; Sherif Omran<br>&gt;<br>&gt;<br>&gt; _________________________________________________________________________<br>&gt; Professional FreeSWITCH Consulting Services:<br>
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<br></blockquote></div><br>