<div dir="ltr">I ran the netstat command, and it doesn&#39;t appear to be listening. It doesn&#39;t appear to be listening on anything.  I do have this error in the console when starting:<div><br></div><div><div>2013-09-11 18:16:14.445921 [ERR] switch_nat.c:201 Error checking for PMP [general error]</div>

<div><br></div><div>AND</div><div><br></div><div><div>2013-09-11 18:09:30.233568 [CRIT] switch_loadable_module.c:1383 Error Loading module /opt/freeswitch-1.4b/mod/mod_cluechoo.so</div><div>**/opt/freeswitch-1.4b/mod/mod_cluechoo.so: undefined symbol: waddch**</div>

</div><div><br></div><div>I  believe I overlooked these earlier. But it&#39;s possible I&#39;m missing a dependency.  I&#39;m going to install the libncurses packages as described here <a href="http://jira.freeswitch.org/browse/FS-3689">http://jira.freeswitch.org/browse/FS-3689</a> and then rebuild to see if that helps.  I&#39;m running Ubuntu 12.10.</div>

</div><div><br></div><div>James</div><div><br></div></div><div class="gmail_extra"><br clear="all"><div><div dir="ltr"><div><br></div><div>James<br></div><div><br><div><i><b>&quot;Every Call, Every Time!&quot;</b></i></div>

<div><br></div><div><h4 style="font-family:Helvetica,Arial,sans-serif;font-size:14px;color:#111;padding:0px;margin:0 0 8px">How did I do?</h4><p style="padding:0px;margin:0 0 8px"><a href="https://secure.teamhively.com/welcome/rate/735/3" target="_blank"><img width="41" height="45" style="border:none;font-size:9px;text-decoration:none;margin-right:15px;font-weight:bold;color:#5b723f" alt="Happy" src="https://secure.teamhively.com/images/icon-happy_45.png"></a><a href="https://secure.teamhively.com/welcome/rate/735/2" target="_blank"><img width="41" height="45" style="border:none;font-size:9px;text-decoration:none;margin-right:15px;font-weight:bold;color:#0e88b6" alt="Satisfied" src="https://secure.teamhively.com/images/icon-satisfied_45.png"></a><a href="https://secure.teamhively.com/welcome/rate/735/1" target="_blank"><img width="41" height="45" style="border:none;font-size:9px;text-decoration:none;font-weight:bold;color:#e2644f" alt="Unhappy" src="https://secure.teamhively.com/images/icon-unhappy_45.png"></a></p>

<p style="padding:0px;margin:0 0 8px"><a href="https://secure.teamhively.com/welcome/rate/735" style="font-family:Helvetica,Arial,sans-serif;font-size:12px;font-weight:bold;color:rgb(88,169,210)" target="_blank">Click on a face to provide feedback on my performance!</a></p>

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<br><br><div class="gmail_quote">On Wed, Sep 11, 2013 at 11:07 AM, Anthony Minessale <span dir="ltr">&lt;<a href="mailto:anthony.minessale@gmail.com" target="_blank">anthony.minessale@gmail.com</a>&gt;</span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">

<div dir="ltr">Did you open all the necessary firewall ports?<div>Playing around on amazon as your first try complicates thing a bit for you.</div><div><br></div><div>You should be able to verify its listening on the port with netstat -an | grep 5066</div>


<div><br></div></div><div class="HOEnZb"><div class="h5"><div class="gmail_extra"><br><br><div class="gmail_quote">On Wed, Sep 11, 2013 at 12:55 PM, James Mortensen <span dir="ltr">&lt;<a href="mailto:james.mortensen@synclio.com" target="_blank">james.mortensen@synclio.com</a>&gt;</span> wrote:<br>


<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir="ltr">Here&#39;s another update to my adventures in Freeswitch WebRTC.  I assume from the getting started documentation that there are users created by default with the password 1234, so I&#39;m trying to create the ws 5066 connection from the TryIt JsSIP demo:  <a href="http://tryit.jssip.net" target="_blank">http://tryit.jssip.net</a><div>




<br></div><div>Name: James</div><div>SIP URI: 1000@Y.Y.Y.Y   &lt;--- Local IP of EC2 server<br><div class="gmail_extra">SIP password: 1234</div><div class="gmail_extra">WS URI:  ws://X.X.X.X:5066   &lt;--- Public IP of EC2 server<br clear="all">




<div><div dir="ltr"><br></div></div><div>Hope this helps!</div><span><font color="#888888"><div>James</div></font></span><div><div>
<br><br><div class="gmail_quote">On Wed, Sep 11, 2013 at 10:31 AM, James Mortensen <span dir="ltr">&lt;<a href="mailto:james.mortensen@synclio.com" target="_blank">james.mortensen@synclio.com</a>&gt;</span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">




<div dir="ltr">Hi Anthony,<div><br></div><div>I have a Chrome client behind a NAT&#39;d network, and I have a Freeswitch 1.4b instance sitting behind a NAT&#39;d EC2 server.  I enabled ws-binding in internal.xml. It&#39;s the only area where I see anything related to Websockets.  </div>





<div><br></div><div>I also verified my firewall is opened on port 5066. I can see requests coming in from Chrome when I try to register, but Freeswitch&#39;s logs don&#39;t even twitch.  I followed the instructions here to enable DEBUG logging: <a href="http://wiki.freeswitch.org/wiki/Troubleshooting_Freeswitch" target="_blank">http://wiki.freeswitch.org/wiki/Troubleshooting_Freeswitch</a></div>





<div><br></div><div>I&#39;ve also grepped the configuration files and internal.xml is the only location of &quot;websocket&quot;, other than the htdocs.... which seem to relate to getting a GUI running.</div><div><br></div>





<div>Perhaps there&#39;s something I&#39;m misunderstanding about logging in Freeswitch?  I verified from the console that debug logging is enabled.</div><div><br></div><div>Hope this helps and thank you!</div><span><font color="#888888"><div>




James</div></font></span><div><div>
<div class="gmail_extra"><br clear="all"><div><div dir="ltr"><div><br></div></div></div><br><div class="gmail_quote">On Wed, Sep 11, 2013 at 9:15 AM, Anthony Minessale <span dir="ltr">&lt;<a href="mailto:anthony.minessale@gmail.com" target="_blank">anthony.minessale@gmail.com</a>&gt;</span> wrote:<br>





<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div dir="ltr">Are you testing by connecting to internal or external IP?<div>Uncommenting the param is really the only thing you have to do.</div>





<div><br></div><div><br></div></div><div class="gmail_extra"><div><div><br><br><div class="gmail_quote">
On Wed, Sep 11, 2013 at 10:28 AM, James Mortensen <span dir="ltr">&lt;<a href="mailto:james.mortensen@synclio.com" target="_blank">james.mortensen@synclio.com</a>&gt;</span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">






<div dir="ltr">Hi Brian,<div><br></div><div>I didn&#39;t stray too far from the default configuration.  I just uncommented the param and left the default port in place, 5066.  Is there a way to tell if the WS/HTTP server is running from within Freeswitch?  I didn&#39;t see a command for that.</div>









<div><br></div><div>I also replaced the ext-sip-ip and ext-rtp-ip with my stun server and my server&#39;s public IP in internal.xml:</div><div><br></div><div><div>    &lt;param name=&quot;ext-rtp-ip&quot; value=&quot;stun:<a href="http://stun.l.google.com:19302" target="_blank">stun.l.google.com:19302</a>&quot;/&gt;</div>








<div>    &lt;param name=&quot;ext-sip-ip&quot; value=&quot;MY_PUBLIC_IP&quot; /&gt;</div></div><div><br></div><div>The server starts up, but doesn&#39;t accept websocket connections.  What information can I get you so this is more helpful?</div>








<div><br></div><div>Thank you!</div><div>James </div></div><div class="gmail_extra"><br clear="all"><div><div dir="ltr"><div><br></div><div>James<br></div><div><br><div><i><b>&quot;Every Call, Every Time!&quot;</b></i></div>








<div><br></div><div><h4 style="font-family:Helvetica,Arial,sans-serif;font-size:14px;color:#111;padding:0px;margin:0 0 8px">How did I do?</h4><p style="padding:0px;margin:0 0 8px"><a href="https://secure.teamhively.com/welcome/rate/735/3" target="_blank"><img width="41" height="45" style="border:none;font-size:9px;text-decoration:none;margin-right:15px;font-weight:bold;color:#5b723f" alt="Happy"></a><a href="https://secure.teamhively.com/welcome/rate/735/2" target="_blank"><img width="41" height="45" style="border:none;font-size:9px;text-decoration:none;margin-right:15px;font-weight:bold;color:#0e88b6" alt="Satisfied"></a><a href="https://secure.teamhively.com/welcome/rate/735/1" target="_blank"><img width="41" height="45" style="border:none;font-size:9px;text-decoration:none;font-weight:bold;color:#e2644f" alt="Unhappy"></a></p>






<div>

<p style="padding:0px;margin:0 0 8px"><a href="https://secure.teamhively.com/welcome/rate/735" style="font-family:Helvetica,Arial,sans-serif;font-size:12px;font-weight:bold;color:rgb(88,169,210)" target="_blank">Click on a face to provide feedback on my performance!</a></p>








</div></div><div><div><br></div><div>Synclio Extension:<i>  110627</i></div><div>Phone: <i> <a href="tel:866-707-4590" value="+18667074590" target="_blank">866-707-4590</a></i></div><div>Skype: <i>No!</i> <i>Call me on Synclio! :)</i></div>






</div></div></div></div>
<br><br><div class="gmail_quote"><div><div>On Wed, Sep 11, 2013 at 6:13 AM, Brian West <span dir="ltr">&lt;<a href="mailto:brian@freeswitch.org" target="_blank">brian@freeswitch.org</a>&gt;</span> wrote:<br></div>
</div><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex"><div><div>

What exactly did you put into the ws-binding param?<br>
<span><font color="#888888"><br>
/b<br>
</font></span><div><div><br>
<br>
On Sep 10, 2013, at 8:48 PM, James Mortensen &lt;<a href="mailto:james.mortensen@synclio.com" target="_blank">james.mortensen@synclio.com</a>&gt; wrote:<br>
<br>
&gt; Hello,<br>
&gt;<br>
&gt; I&#39;m working on a one-way audio bug that may or may not be either Asterisk or Chrome WebRTC related. Details are here: <a href="https://code.google.com/p/webrtc/issues/detail?id=2347" target="_blank">https://code.google.com/p/webrtc/issues/detail?id=2347</a>, but basically the problem is that I get about 1 in 10 calls returning one way audio where audio flows from the PSTN to Chrome but not the other way.<br>









&gt;<br>
&gt; I believe the issue may be with Asterisk 11, based on Wireshark traces and other troubleshooting, and if that&#39;s the case, I&#39;m happy to try Freeswitch as an alternative.<br>
&gt; I tried the demo here:  <a href="https://webrtc.freeswitch.org/webrtc/portal.html" target="_blank">https://webrtc.freeswitch.org/webrtc/portal.html</a>, and the 20 test calls to the PSTN worked perfect with two way audio.<br>









&gt;<br>
&gt; So, now, onto the Freeswitch question:  Where is all the documentation for configuring the Websocket server on Freeswitch 1.4b?  Where is the documentation that explains how to get ICE enabled?  Why, when I enabled the ws-bind port in internal.xml does the client fail to establish a WS connection?<br>









&gt;<br>
&gt; Basically, I&#39;d love to get my hands on the docs for setting this up, but the Internet is largely silent on this matter?  Is it possible for the Freeswitch team to share the configuration they used to get that JsSIP/Freeswitch demo setup?<br>









&gt;<br>
&gt; If not the configuration, where would I find the docs for Freeswitch 1.4b?<br>
&gt;<br>
&gt; Thank you!!<br>
&gt;<br>
&gt; James<br>
&gt;<br>
&gt; &quot;Every Call, Every Time!&quot;<br>
&gt;<br>
&gt; How did I do?<br>
&gt;<br>
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<br></blockquote></div><br><br clear="all"><div><br></div></div></div>-- <br>Anthony Minessale II<br><br>FreeSWITCH <a href="http://www.freeswitch.org/" target="_blank">http://www.freeswitch.org/</a><br>ClueCon <a href="http://www.cluecon.com/" target="_blank">http://www.cluecon.com/</a><br>






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pstn:<a href="tel:%2B19193869900" value="+19193869900" target="_blank">+19193869900</a>
</div>
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</blockquote></div><br></div></div></div></div></div>
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<br></blockquote></div><br><br clear="all"><div><br></div>-- <br>Anthony Minessale II<br><br>FreeSWITCH <a href="http://www.freeswitch.org/" target="_blank">http://www.freeswitch.org/</a><br>ClueCon <a href="http://www.cluecon.com/" target="_blank">http://www.cluecon.com/</a><br>


Twitter: <a href="http://twitter.com/FreeSWITCH_wire" target="_blank">http://twitter.com/FreeSWITCH_wire</a><br><br>AIM: anthm<br><a href="mailto:MSN%3Aanthony_minessale@hotmail.com" target="_blank">MSN:anthony_minessale@hotmail.com</a><br>

GTALK/JABBER/<a href="mailto:PAYPAL%3Aanthony.minessale@gmail.com" target="_blank">PAYPAL:anthony.minessale@gmail.com</a><br>
IRC: <a href="http://irc.freenode.net" target="_blank">irc.freenode.net</a> #freeswitch<br><br>FreeSWITCH Developer Conference<br><a href="mailto:sip%3A888@conference.freeswitch.org" target="_blank">sip:888@conference.freeswitch.org</a><br>

<a href="mailto:googletalk%3Aconf%2B888@conference.freeswitch.org" target="_blank">googletalk:conf+888@conference.freeswitch.org</a><br>
pstn:<a href="tel:%2B19193869900" value="+19193869900" target="_blank">+19193869900</a>
</div>
</div></div><br>_________________________________________________________________________<br>
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<br>
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<br></blockquote></div><br></div>