<div dir="ltr">did you do all the things in <a href="https://wiki.freeswitch.org/wiki/Amazon_EC2">https://wiki.freeswitch.org/wiki/Amazon_EC2</a><div><br></div><div>Also are you using the external or internal profile (port 5065 is the default unauthenticated port)</div>
<div>if you want to use internal you have to comment out the auth-calls and the inbound acl lines. </div><div><br></div><div>Most of this should be available in our various books and wiki articles.</div><div><br></div></div>
<div class="gmail_extra"><br><br><div class="gmail_quote">On Fri, Sep 13, 2013 at 3:31 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"><div>Hi Anothony,<br></div><div><br></div><div>Inbound calling is proving to be more challenging.  Here&#39;s what I&#39;ve tried from <a href="https://wiki.freeswitch.org/wiki/Auto_NAT" target="_blank">https://wiki.freeswitch.org/wiki/Auto_NAT</a>:</div>


<div><br></div><div>1.  Verified the following params were set in my sip profiles (these were already set by default):</div><div><br></div><div><pre style="line-height:1.3em;font-size:13px;background-color:rgb(249,249,249);font-family:monospace,Courier;border:1px dashed rgb(47,111,171);padding:1em">

  &lt;param name=&quot;ext-rtp-ip&quot; value=&quot;auto-nat&quot;/&gt;
    &lt;param name=&quot;ext-sip-ip&quot; value=&quot;auto-nat&quot;/&gt;</pre></div><div class="gmail_extra"><br></div><div class="gmail_extra">2.  Ran sofia status, which shows this:</div><div class="gmail_extra"><br></div>


<div class="gmail_extra"><div class="gmail_extra"><br></div><div class="gmail_extra">                     Name<span style="white-space:pre-wrap">        </span>   Type<span style="white-space:pre-wrap">        </span>                                      Data<span style="white-space:pre-wrap">        </span>State</div>


<div class="gmail_extra">=================================================================================================</div><div class="gmail_extra">           10.166.245.111<span style="white-space:pre-wrap">        </span>  alias<span style="white-space:pre-wrap">        </span>                                  internal<span style="white-space:pre-wrap">        </span>ALIASED</div>


<div class="gmail_extra">            internal-ipv6<span style="white-space:pre-wrap">        </span>profile<span style="white-space:pre-wrap">        </span>                  sip:mod_sofia@[::1]:5060<span style="white-space:pre-wrap">        </span>RUNNING (0)</div>


<div class="gmail_extra">                 external<span style="white-space:pre-wrap">        </span>profile<span style="white-space:pre-wrap">        </span>         <a href="http://sip:mod_sofia@10.166.245.111:5080" target="_blank">sip:mod_sofia@10.166.245.111:5080</a><span style="white-space:pre-wrap">        </span>RUNNING (0)</div>


<div class="gmail_extra">    external::<a href="http://example.com" target="_blank">example.com</a><span style="white-space:pre-wrap">        </span>gateway<span style="white-space:pre-wrap">        </span>                   <a href="mailto:sip%3Ajoeuser@example.com" target="_blank">sip:joeuser@example.com</a><span style="white-space:pre-wrap">        </span>NOREG</div>


<div class="gmail_extra">                 internal<span style="white-space:pre-wrap">        </span>profile<span style="white-space:pre-wrap">        </span>         <a href="http://sip:mod_sofia@10.166.245.111:5060" target="_blank">sip:mod_sofia@10.166.245.111:5060</a><span style="white-space:pre-wrap">        </span>RUNNING (0)</div>


<div class="gmail_extra">  internal::<a href="http://bandwidth.com" target="_blank">bandwidth.com</a><span style="white-space:pre-wrap">        </span>gateway<span style="white-space:pre-wrap">        </span>           sip:your user <a href="mailto:name@67.231.8.195" target="_blank">name@67.231.8.195</a><span style="white-space:pre-wrap">        </span>NOREG</div>


<div class="gmail_extra">=================================================================================================</div><div class="gmail_extra">3 profiles 1 alias</div><div><br></div></div><div class="gmail_extra">


<br></div><div class="gmail_extra">3. Ran <b style="line-height:19.1875px;font-size:13px;font-family:sans-serif">sofia status profile internal</b><span style="line-height:19.1875px;font-size:13px;font-family:sans-serif"> expecting to see ext-rtp-ip and ext-sip-ip set, but they&#39;re missing:</span></div>


<div class="gmail_extra"><span style="line-height:19.1875px;font-size:13px;font-family:sans-serif"><br></span></div><div class="gmail_extra"><div class="gmail_extra">Name             <span style="white-space:pre-wrap">        </span>internal</div>


<div class="gmail_extra">Domain Name      <span style="white-space:pre-wrap">        </span>N/A</div><div class="gmail_extra">Auto-NAT         <span style="white-space:pre-wrap">        </span>false</div><div class="gmail_extra">

DBName           <span style="white-space:pre-wrap">        </span>sofia_reg_internal</div><div class="gmail_extra">Pres Hosts       <span style="white-space:pre-wrap">        </span>10.166.245.111,10.166.245.111</div><div class="gmail_extra">


Dialplan         <span style="white-space:pre-wrap">        </span>XML</div><div class="gmail_extra">Context          <span style="white-space:pre-wrap">        </span>public</div><div class="gmail_extra">Challenge Realm  <span style="white-space:pre-wrap">        </span>auto_from</div>


<div class="gmail_extra">RTP-IP           <span style="white-space:pre-wrap">        </span>10.166.245.111</div><div class="gmail_extra">SIP-IP           <span style="white-space:pre-wrap">        </span>10.166.245.111</div><div class="gmail_extra">


URL              <span style="white-space:pre-wrap">        </span><a href="http://sip:mod_sofia@10.166.245.111:5060" target="_blank">sip:mod_sofia@10.166.245.111:5060</a></div><div class="gmail_extra">BIND-URL         <span style="white-space:pre-wrap">        </span>sip:mod_sofia@10.166.245.111:5060;transport=udp,tcp</div>


<div class="gmail_extra">HOLD-MUSIC       <span style="white-space:pre-wrap">        </span>local_stream://moh</div><div class="gmail_extra">OUTBOUND-PROXY   <span style="white-space:pre-wrap">        </span>N/A</div><div class="gmail_extra">


CODECS IN        <span style="white-space:pre-wrap">        </span>G722,PCMU,PCMA,GSM</div><div class="gmail_extra">CODECS OUT       <span style="white-space:pre-wrap">        </span>G722,PCMU,PCMA,GSM</div><div class="gmail_extra">

TEL-EVENT        <span style="white-space:pre-wrap">        </span>101</div><div class="gmail_extra">DTMF-MODE        <span style="white-space:pre-wrap">        </span>rfc2833</div><div class="gmail_extra">CNG              <span style="white-space:pre-wrap">        </span>13</div>


<div class="gmail_extra">SESSION-TO       <span style="white-space:pre-wrap">        </span>0</div><div class="gmail_extra">MAX-DIALOG       <span style="white-space:pre-wrap">        </span>0</div><div class="gmail_extra">NOMEDIA          <span style="white-space:pre-wrap">        </span>false</div>


<div class="gmail_extra">LATE-NEG         <span style="white-space:pre-wrap">        </span>true</div><div class="gmail_extra">PROXY-MEDIA      <span style="white-space:pre-wrap">        </span>false</div><div class="gmail_extra">

ZRTP-PASSTHRU    <span style="white-space:pre-wrap">        </span>true</div><div class="gmail_extra">AGGRESSIVENAT    <span style="white-space:pre-wrap">        </span>false</div><div class="gmail_extra">CALLS-IN         <span style="white-space:pre-wrap">        </span>86</div>


<div class="gmail_extra">FAILED-CALLS-IN  <span style="white-space:pre-wrap">        </span>86</div><div class="gmail_extra">CALLS-OUT        <span style="white-space:pre-wrap">        </span>0</div><div class="gmail_extra">FAILED-CALLS-OUT <span style="white-space:pre-wrap">        </span>0</div>


<div class="gmail_extra">REGISTRATIONS    <span style="white-space:pre-wrap">        </span>1</div><div><br></div><div><br></div><div>So regarding NAT issues, it seems Freeswitch isn&#39;t able to get the server&#39;s public IP perhaps, despite auto-nat being the correct setting?  This is just an EC2 server behind Amazon NAT.</div>


</div><div class="gmail_extra"><span style="line-height:19.1875px;font-size:13px;font-family:sans-serif"><br></span></div><div class="gmail_extra"><span style="line-height:19.1875px;font-size:13px;font-family:sans-serif">Also, I ran:</span></div>


<div class="gmail_extra"><div class="gmail_extra"><font color="#000000" face="sans-serif"><span style="line-height:19.1875px">nat_map status</span></font></div><div class="gmail_extra"><font color="#000000" face="sans-serif"><span style="line-height:19.1875px"><br>


</span></font></div><div class="gmail_extra"><font color="#000000" face="sans-serif"><span style="line-height:19.1875px">Nat Type: UNKNOWN, ExtIP: </span></font></div><div class="gmail_extra"><font color="#000000" face="sans-serif"><span style="line-height:19.1875px">NAT port mapping enabled.</span></font></div>


<div class="gmail_extra"><font color="#000000" face="sans-serif"><span style="line-height:19.1875px"><br></span></font></div><div class="gmail_extra"><font color="#000000" face="sans-serif"><span style="line-height:19.1875px">0 total.</span></font></div>


<div style="line-height:19.1875px;font-size:13px;font-family:sans-serif"><br></div></div><div class="gmail_extra"><br></div><div class="gmail_extra">I did perform an echo test from user 1000 to 9196 from Chrome to Freeswitch, which works as expected, but getting Bandwidth incoming calls to be processed correctly is a challenge. It just loops over and over again with this in the DEBUG console:</div>


<div class="gmail_extra"><br></div><div class="gmail_extra"><div class="gmail_extra">2013-09-13 20:27:31.255055 [NOTICE] switch_channel.c:1030 New Channel sofia/internal/+<a href="mailto:19712383780@192.168.37.68" target="_blank">19712383780@192.168.37.68</a> [80196f26-febf-4363-b48c-ee88022c6b9c]</div>


<div class="gmail_extra">2013-09-13 20:27:31.255055 [DEBUG] switch_core_session.c:1006 Send signal sofia/internal/+<a href="mailto:19712383780@192.168.37.68" target="_blank">19712383780@192.168.37.68</a> [BREAK]</div><div class="gmail_extra">


2013-09-13 20:27:31.255055 [DEBUG] switch_core_session.c:1006 Send signal sofia/internal/+<a href="mailto:19712383780@192.168.37.68" target="_blank">19712383780@192.168.37.68</a> [BREAK]</div><div class="gmail_extra">2013-09-13 20:27:31.255055 [DEBUG] switch_core_state_machine.c:418 (sofia/internal/+<a href="mailto:19712383780@192.168.37.68" target="_blank">19712383780@192.168.37.68</a>) Running State Change CS_NEW</div>


<div class="gmail_extra">2013-09-13 20:27:31.255055 [DEBUG] switch_core_state_machine.c:436 (sofia/internal/+<a href="mailto:19712383780@192.168.37.68" target="_blank">19712383780@192.168.37.68</a>) State NEW</div><div class="gmail_extra">


2013-09-13 20:27:31.275057 [DEBUG] sofia.c:8003 IP 67.231.4.195 Rejected by acl &quot;domains&quot;. Falling back to Digest auth.</div><div class="gmail_extra">2013-09-13 20:27:31.275057 [DEBUG] switch_core_session.c:1006 Send signal sofia/internal/+<a href="mailto:19712383780@192.168.37.68" target="_blank">19712383780@192.168.37.68</a> [BREAK]</div>


<div class="gmail_extra">2013-09-13 20:27:31.275057 [DEBUG] sofia.c:1787 detaching session 80196f26-febf-4363-b48c-ee88022c6b9c</div><div><br></div></div><div class="gmail_extra"><br></div><div class="gmail_extra">So in summary, the WebRTC seems good but not the PSTN. I can do Chrome to FS to Chrome, Chrome to FS Echo, I can call from Chrome to FS to Bandwidth (NO AUDIO) but calling inbound loops.  I&#39;m assuming all of the routing is done using the Bandwidth DID I&#39;ve pointed to the server, not the 971xxxxxxx number that I&#39;m calling from on Verizon/Google Voice.</div>


<div class="gmail_extra"><br></div><div class="gmail_extra">Thank you for your help!</div><span class="HOEnZb"><font color="#888888"><div class="gmail_extra">James</div></font></span><div><div class="h5"><div class="gmail_extra">
<br></div><div class="gmail_extra"><br><div class="gmail_quote">On Fri, Sep 13, 2013 at 11:51 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:0px 0px 0px 0.8ex;border-left-width:1px;border-left-color:rgb(204,204,204);border-left-style:solid;padding-left:1ex"><div dir="ltr">That should not matter.  It will be taken care of.<div>


<br></div><div>Verify you can just use bandwidth to call the server and run an echo test or something.</div><div>Then try a sip phone registered.  Then on to the webrtc instance.</div>
<div><br></div><div>You probably have some nat problems to bandwidth regardless of WebRTC.</div></div><div class="gmail_extra"><br><br><div class="gmail_quote"><div><div>On Fri, Sep 13, 2013 at 1:42 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>



</div></div><blockquote class="gmail_quote" style="margin:0px 0px 0px 0.8ex;border-left-width:1px;border-left-color:rgb(204,204,204);border-left-style:solid;padding-left:1ex"><div><div><div dir="ltr">Here is my follow up to this issue.  It seems something happened to the server where it stopped sending Binding responses to Chrome.  This is a known issue that I&#39;ve seen before that Google says was an Asterisk issue.  However, the same thing happens to Freeswitch as well, indicating the problem is the network/server, not the software.<div>





<br></div><div>I booted up another server from a snapshot, verified two way audio with Asterisk, then reinstalled Freeswitch, uncommented the ws-binding parameter to enable WS on port 5066, then registered user 1000 and 1002 from the <a href="http://tryit.jssip.net" target="_blank">tryit.jssip.net</a> demo using the following configuration:</div>





<div><br></div><div>Name: James</div><div>SIP URI:  sip:1000@54.X.X.X   &lt;-- public IP of server</div><div>password: 1234</div><div>WS URI: ws://54.X.X.X:5066  &lt;--- same public IP</div><div><br></div><div>and I substituted 1002 in place of 1000 for another user on another network.  I verified audio flows both ways with audio flowing through Freeswitch.</div>





<div><br></div><div>I made no other changes to the configuration.  This was a lot easier to get started with than Asterisk WebRTC.  I just made it out to be a lot harder than it was by assuming I needed to manually add in my server&#39;s IP address in the configuration files.  This happens automatically, even on a NAT&#39;d server.  Amazing! :)</div>





<div><br></div><div><br></div><div>Now, for the PSTN part, I configured my BANDWIDTH.com provider as an internal context and as an outbound and inbound dialplan.  I can connect a call between my cellphone and Chrome, but there&#39;s no audio flowing to/from Bandwidth.  I suspect the problem has something to do with bridging AVP and SAVPF, since the carriers don&#39;t support SRTP.</div>





<div><br></div><div>I&#39;ve grepped the configuration files, and I don&#39;t see how I would configure the system to bridge AVP and SAVPF and do transcoding.  Any ideas?</div><div><br></div><div>Thank you!!</div><span><font color="#888888"><div>



<br>

</div><div>James</div></font></span><div><div><div class="gmail_extra"><div><div dir="ltr"><div><br></div><div><br></div></div></div><br><div class="gmail_quote">On Wed, Sep 11, 2013 at 12:01 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:0px 0px 0px 0.8ex;border-left-width:1px;border-left-color:rgb(204,204,204);border-left-style:solid;padding-left:1ex"><div dir="ltr"><div class="gmail_extra"><div><div dir="ltr">


<div>Installing <span style="line-height:17px;font-size:13px;background-color:rgb(240,240,240);font-family:Arial,FreeSans,Helvetica,sans-serif">ibncursesw5 and </span><span style="line-height:17px;font-size:13px;background-color:rgb(240,240,240);font-family:Arial,FreeSans,Helvetica,sans-serif">libncursesw5-dev</span> did resolve the issue.  I can now connect to the WebSocket server.  Seems the cluechoo module is just something that was added in to separate the help vampires from the people with legitimate issues. :D</div>







<div><br></div><div>I got 405 Method Not Allowed errors, and I resolved them by making sure the IP address in the SIP URI matches the IP address in the WS field.  </div><div><br></div><div>At this time, I&#39;ve successfully registered and am getting back 200 OK&#39;s and have now moved onto tweaking the settings so I can get audio going both ways. The candidates appear to be only showing the public IP, so I&#39;ll have to figure that out. I&#39;ll provide more updates, or questions, as I move forward.</div>





<span><font color="#888888">

<div><br></div><div>James</div></font></span></div></div>
<br><br><div class="gmail_quote"><div>On Wed, Sep 11, 2013 at 11:32 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>





</div><blockquote class="gmail_quote" style="margin:0px 0px 0px 0.8ex;border-left-width:1px;border-left-color:rgb(204,204,204);border-left-style:solid;padding-left:1ex">

<div dir="ltr"><div>Okay, I added in the dependencies and still get those errors.  Do I need to file a bug in JIRA for this or am I just missing something?  <div><br></div><div>I&#39;m happy to try a different OS if there&#39;s one that&#39;s been tried and tested.</div>








<div><br></div><div>Of course, more googling reveals that mod_cluechoo is just a joke?  <a href="http://wiki.freeswitch.org/wiki/Mod_cluechoo" target="_blank">http://wiki.freeswitch.org/wiki/Mod_cluechoo</a></div><div><span style="font-size:medium;font-family:Times"><br>








</span></div><div><span style="font-size:medium;font-family:Times">&gt; SL (Steam Locomotive) runs across your terminal when you type &quot;sl&quot; as you meant to type &quot;ls&quot;. It&#39;s just a joke command, and not usefull at all. Put the binary to /usr/local/bin.</span></div>








<div><br></div><div>I hope I&#39;m not chasing a problem that has absolutely no impact on my ability to get WebRTC working here. :D</div><div><br></div><div><br></div><div>Thanks for any additional help you can provide,</div>








<div>James</div></div><div><div><div><br></div><div class="gmail_extra"><div><div dir="ltr"><div><br></div></div></div><br><div class="gmail_quote"><div>On Wed, Sep 11, 2013 at 11:19 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>








</div><blockquote class="gmail_quote" style="margin:0px 0px 0px 0.8ex;border-left-width:1px;border-left-color:rgb(204,204,204);border-left-style:solid;padding-left:1ex"><div><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" target="_blank">http://jira.freeswitch.org/browse/FS-3689</a> and then rebuild to see if that helps.  I&#39;m running Ubuntu 12.10.</div>








<span><font color="#888888">
</font></span></div><span><font color="#888888"><div><br></div><div>James</div><div><br></div></font></span></div></div><div class="gmail_extra"><div><div><div dir="ltr"><div>
</div></div></div>
<br><br></div><div><div><div><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:0px 0px 0px 0.8ex;border-left-width:1px;border-left-color:rgb(204,204,204);border-left-style: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><div><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:0px 0px 0px 0.8ex;border-left-width:1px;border-left-color:rgb(204,204,204);border-left-style: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"><br></div></div></div></div></div></div></blockquote></div></div></div></div></blockquote></div><br></div></div></div></div>
</blockquote></div><br></div></div></div></div>
</blockquote></div><br></div></div>
</blockquote></div><br></div></div></div></div>
<br></div></div><div><div>_________________________________________________________________________<br>
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<br></div></div></blockquote></div><div><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>

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</div></div></div>
<br>_________________________________________________________________________<br>
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<br>
FreeSWITCH-powered IP PBX: The CudaTel Communication Server<br>
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<br></blockquote></div><br></div></div></div></div>
<br>_________________________________________________________________________<br>
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<br>
FreeSWITCH-powered IP PBX: The CudaTel Communication Server<br>
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<br>
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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/">http://www.freeswitch.org/</a><br>ClueCon <a href="http://www.cluecon.com/">http://www.cluecon.com/</a><br>
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pstn:+19193869900
</div>