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    Thanks Jack,<br>
    <br>
    but as mentioned we are at an earlier stage. So far the Flex client
    has not yet conneted to Freeswitch.<br>
    We still have a connect problem.<br>
    <br>
    Best regards<br>
    Peter<br>
    <br>
    On 08/07/12 17:51, Jack wrote:
    <blockquote cite="mid:5021399D.2030501@livecall.com" type="cite">
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      Peter,<br>
      &nbsp; Here is my rtmp.conf.xml:<br>
      <br>
      &lt;configuration name="rtmp.conf" description="RTMP Endpoint"&gt;<br>
      &nbsp; &lt;profiles&gt;<br>
      &nbsp;&nbsp;&nbsp; &lt;profile name="default"&gt;<br>
      &nbsp;&nbsp;&nbsp; &nbsp;&nbsp;&nbsp; &lt;settings&gt;<br>
      &nbsp;&nbsp;&nbsp; &nbsp;&nbsp;&nbsp; &nbsp;&nbsp;&nbsp; &lt;param name="bind-address" value="0.0.0.0:1935"
      /&gt;<br>
      &nbsp;&nbsp;&nbsp; &nbsp;&nbsp;&nbsp; &nbsp;&nbsp;<big>&nbsp; &lt;param name="context" value="public" /&gt;<br>
      </big>&nbsp;&nbsp;&nbsp; &nbsp;&nbsp;&nbsp; &nbsp;&nbsp;&nbsp; &lt;param name="dialplan" value="XML" /&gt;<br>
      &nbsp;&nbsp;&nbsp; &nbsp;&nbsp;&nbsp; &nbsp;&nbsp;&nbsp; &lt;!-- If this is set to true, no unauthenticated
      inbound calls will be allowed --&gt;<br>
      &nbsp;&nbsp;&nbsp; &nbsp;&nbsp;&nbsp; &nbsp;&nbsp;&nbsp; &lt;param name="auth-calls" value="true" /&gt;<br>
      &nbsp;&nbsp;&nbsp; &nbsp;&nbsp;&nbsp; &nbsp;&nbsp;&nbsp; &lt;!-- How much time should the clients buffer the
      media stream (miliseconds) --&gt;<br>
      &nbsp;&nbsp;&nbsp; &nbsp;&nbsp;&nbsp; &nbsp;&nbsp;&nbsp; &lt;param name="buffer-len" value="50" /&gt;<br>
      &nbsp;&nbsp;&nbsp; &nbsp;&nbsp;&nbsp; &nbsp;&nbsp;&nbsp; &lt;!-- Sets the maximum size of outbound RTMP chunks
      --&gt;<br>
      &nbsp;&nbsp;&nbsp; &nbsp;&nbsp;&nbsp; &nbsp;&nbsp;&nbsp; &lt;param name="chunksize" value="512" /&gt;<br>
      &nbsp;&nbsp;&nbsp; &nbsp;&nbsp;&nbsp; &lt;/settings&gt;<br>
      &nbsp;&nbsp;&nbsp; &lt;/profile&gt;<br>
      &nbsp; &lt;/profiles&gt;<br>
      &lt;/configuration&gt;<br>
      <br>
      It is important to specify the context and then make sure you have
      a condition in your corresponding dial plan that will catch your
      call.<br>
      <br>
      Jack<br>
      <br>
      <br>
      <br>
      <div class="moz-cite-prefix">On 8/7/2012 1:11 AM, Peter Steinbach
        wrote:<br>
      </div>
      <blockquote cite="mid:5020CDB9.3060102@telefaks.de" type="cite">
        <pre wrap="">Hello Jack,

here are some answers to your your questions:

</pre>
        <blockquote type="cite">
          <pre wrap="">Make sure you use a fully qualified username not just the extension
<a moz-do-not-send="true" class="moz-txt-link-abbreviated" href="mailto:1001@xxx.xxx.xxx.xxx">1001@xxx.xxx.xxx.xxx</a> &gt;the XXX would be the IP of your FreeSwitch Server.
</pre>
        </blockquote>
        <pre wrap="">I am still in the "connect" state, I am not logging in yet. The webpage
tells me "Connecting... " and waits forever

</pre>
        <blockquote type="cite">
          <pre wrap="">did you configure your rtmp.conf.xml and have a matching context in
</pre>
        </blockquote>
        <pre wrap="">your dial plan?
I am still in the "connect" state

In the web page make sure these two vars are set to YOUR Freeswitch
server ip: var rtmpIPURL =
</pre>
        <blockquote type="cite">
          <pre wrap="">"rtmp://xxx.xxx.xxx.xxx/phone"; var rtmpIP="xxx.xxx.xxx.xxx";
</pre>
        </blockquote>
        <pre wrap="">I did not have a rtmpIP var, so I added it to the code, but no change.

In fact as I can see while grepping the network traffic, the Flex app
and Freeswicth are negociating their protocol and freeswitch logs a
"Sent connect reply", but this is very diferent from what is sent from
the conference.freeswitch.org server.

Best regards
Peter


On 08/07/12 06:26, Jack wrote:
</pre>
        <blockquote type="cite">
          <pre wrap="">Hi Peter,
Make sure you use a fully qualified username not just the extension 
<a moz-do-not-send="true" class="moz-txt-link-abbreviated" href="mailto:1001@xxx.xxx.xxx.xxx">1001@xxx.xxx.xxx.xxx</a>
the XXX would be the IP of your FreeSwitch Server.

did you configure your rtmp.conf.xml and have a matching context in your 
dial plan?

In the web page make sure these two vars are set to YOUR Freeswitch 
server ip:
var rtmpIPURL = "rtmp://xxx.xxx.xxx.xxx/phone";
  var rtmpIP="xxx.xxx.xxx.xxx";

jack

On 8/6/2012 5:44 PM, Peter Steinbach wrote:
</pre>
          <blockquote type="cite">
            <pre wrap="">Hello,

today I tried the flex client with the mod_rtmp implementation.
On a brandnew freeswitch I installed mod_rtmp and copied the flex
direrectory to a web server and loaded the web page.

However the flex client does not connect:
Here's my freeswitch log
2012-08-07 02:32:03.585671 [NOTICE] mod_rtmp.c:744 New RTMP session
[f35a192c-0d02-4320-89e9-775a3573ee25]
2012-08-07 02:32:03.685670 [DEBUG] rtmp.c:702 Sent handshake response
2012-08-07 02:32:03.685670 [DEBUG] rtmp.c:727 Done with handshake
2012-08-07 02:32:03.685670 [DEBUG] rtmp.c:892 [chunk_stream=3 type=0x14
ts=0 stream_id=0x0] len=342
2012-08-07 02:32:03.685670 [DEBUG] rtmp.c:170 [amfnumber=3] Got INVOKE
for connect
2012-08-07 02:32:03.685670 [DEBUG] rtmp.c:566 [amfnumber=2 type=0x1
stream_id=0x0] len=4
2012-08-07 02:32:03.685670 [DEBUG] rtmp.c:566 [amfnumber=2 type=0x5
stream_id=0x0] len=4
2012-08-07 02:32:03.685670 [DEBUG] rtmp.c:566 [amfnumber=2 type=0x6
stream_id=0x0] len=5
2012-08-07 02:32:03.685670 [DEBUG] rtmp.c:566 [amfnumber=2 type=0x4
stream_id=0x0] len=6
2012-08-07 02:32:03.685670 [DEBUG] rtmp.c:566 [amfnumber=3 type=0x14
stream_id=0x0] len=201
2012-08-07 02:32:03.685670 [DEBUG] rtmp.c:566 [amfnumber=3 type=0x14
stream_id=0x0] len=61
2012-08-07 02:32:03.685670 [NOTICE] rtmp_sig.c:121 Sent connect reply
2012-08-07 02:32:03.685670 [DEBUG] rtmp.c:892 [chunk_stream=2 type=0x5
ts=12247364 stream_id=0x0] len=4
2012-08-07 02:32:03.685670 [DEBUG] rtmp.c:945 Set window size: 131072 bytes

If I change the rtmp_url to 'rtmp://conference.freeswitch.org/phone',
flex does connect.

But in my freeswitch this fails. I have traced the network traffic and I
can see that there is information flow between freeswitch and the client
on connect request.

Somebody has an idea where to look further?

</pre>
          </blockquote>
          <pre wrap="">_________________________________________________________________________
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</pre>
        </blockquote>
        <pre wrap="">
_________________________________________________________________________
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</pre>
      </blockquote>
      <br>
      <br>
      <fieldset class="mimeAttachmentHeader"></fieldset>
      <br>
      <pre wrap="">_________________________________________________________________________
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</pre>
    </blockquote>
    <br>
    <br>
    <pre class="moz-signature" cols="72">-- 
With kind regards
Peter Steinbach 

Telefaks Services GmbH
<a class="moz-txt-link-freetext" href="mailto:lists">mailto:lists</a> (att) telefaks.de
Internet: <a class="moz-txt-link-abbreviated" href="http://www.telefaks.de">www.telefaks.de</a>

</pre>
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