Hi Michael,<div><br></div><div>Thanks for getting back to me, this is why I on the whole love the Linux/FOSS community (most of the time!)</div><div><br></div><div>I&#39;ve actually had a fun afternoon studying more and more about OpenSwitch and ended up changing my dial logic totally. I now have a PHP script staying resident, that queries fs_cli every 5 seconds for the number of active calls. If that number is less than the target number of calls per agent (in total) I know I need to add some more calls. So the script grabs a load out the database (based on how many I need to get back to the target # of calls) and chucks them into FreeSwitch using fs_cli -x and just an originate. I originate the call and lob it to an extension for that campaign. </div>

<div><br></div><div>That &#39;extension&#39; then launches an LUA script for each call which starts a playback and runs AVMD at the same time. So we start playing the OGM, but if I hear a beep tone I just hang up as we&#39;re wasting money. If a DTMF has been detected I process that and handle the call accordingly (If they press 9, I go back to the database and add the number to the DNC list, if they press 5 I look for an agent that has been waiting the longest and bridge the call to them).</div>

<div><br></div><div>If neither of those conditions is reached I mark the call in the database for a retry and count how many times we&#39;ve tried it (unless it&#39;s a duff number, then I mark it as such and the call centre get a report at the end of the day on dead lines so they can get a refund on the dead numbers in any particular data set).</div>

<div><br></div><div>It&#39;s almost running now, few bugs to stamp on tomorrow, and then I&#39;ll run it with 20 or so &#39;test&#39; agents tomorrow or Wednesday and see how hard I can push it. I&#39;m on an 8 core Xeon dedicated server at the moment, so I&#39;ll see how hard I can thrash that and then see about migrating it to AWS maybe. </div>

<div><br></div><div>Thanks to all the FreeSwitch developers, my job is suddenly fun again! </div><div><br></div><div>Bon</div><div><br><br><div class="gmail_quote">On 16 July 2012 20:32, Michael Collins <span dir="ltr">&lt;<a href="mailto:msc@freeswitch.org" target="_blank">msc@freeswitch.org</a>&gt;</span> wrote:<br>

<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">Hi Ian,<br><br>Just curious - at what point during the call are you currently launching AVMD? In any case, this sounds like a perfect job for &quot;execute_on_answer&quot;:<br>

<a href="http://wiki.freeswitch.org/wiki/Channel_Variables#execute_on_answer" target="_blank">http://wiki.freeswitch.org/wiki/Channel_Variables#execute_on_answer</a><br>
<br>Just be nice and don&#39;t abuse your new found powers! :)<span class="HOEnZb"><font color="#888888"><br><br>-MC<br></font></span><br>P.S. - Welcome to the FreeSWITCH community.<div class="HOEnZb"><div class="h5"><br>

<br><div class="gmail_quote">On Mon, Jul 16, 2012 at 1:44 AM, Ian Bonham <span dir="ltr">&lt;<a href="mailto:ian.bonham@gmail.com" target="_blank">ian.bonham@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">Hi Everyone,<div><br></div><div>New member, so nice to &#39;meet&#39; you all! </div><div><br></div><div>I&#39;m experimenting with FreeSwitch to see how it could replace Asterisk in a mass dialler system that I run in Europe, and I have been playing with the AVMD module. It seems to work perfectly, and does detect voicemail very well. What I am trying to do through is have the dialler generate calls and run as much as possible through FreeSwitch&#39;s core, rather than external scripts (I&#39;m using PHP for most of my external functionality as thats what I use to write the agent interfaces too). So I generate individual calls and launch them in FreeSwitch, and it then does most of the call logic from XML dialplan commands, jumping to PHP shell scripts for quick MySQL functions. </div>




<div><br></div><div>I&#39;d like very much to be able to run AVMD from the second the call is answered, which is simple enough, but at the same time have my outgoing message playing. So we are assuming the call is a human, but if during the OGM playout AVMD hears a beep, it just terminates the call. Can anyone tell me if this is possible please, either using XML dialplan (I&#39;m guessing not as the XML is sequential), or using a programming script like LUA.</div>




<div><br></div><div>Many thanks for any advise anyone can offer,</div><div><br></div><div>Ian</div><div><br></div></blockquote><br></div><br>
</div></div><br>_________________________________________________________________________<br>
Professional FreeSWITCH Consulting Services:<br>
<a href="mailto:consulting@freeswitch.org">consulting@freeswitch.org</a><br>
<a href="http://www.freeswitchsolutions.com" target="_blank">http://www.freeswitchsolutions.com</a><br>
<br>
FreeSWITCH-powered IP PBX: The CudaTel Communication Server<br>
<a href="http://www.cudatel.com" target="_blank">http://www.cudatel.com</a><br>
<br>
Official FreeSWITCH Sites<br>
<a href="http://www.freeswitch.org" target="_blank">http://www.freeswitch.org</a><br>
<a href="http://wiki.freeswitch.org" target="_blank">http://wiki.freeswitch.org</a><br>
<a href="http://www.cluecon.com" target="_blank">http://www.cluecon.com</a><br>
<br>
Join Us At ClueCon - Aug 7-9, 2012<br>
<br>
FreeSWITCH-users mailing list<br>
<a href="mailto:FreeSWITCH-users@lists.freeswitch.org">FreeSWITCH-users@lists.freeswitch.org</a><br>
<a href="http://lists.freeswitch.org/mailman/listinfo/freeswitch-users" target="_blank">http://lists.freeswitch.org/mailman/listinfo/freeswitch-users</a><br>
UNSUBSCRIBE:<a href="http://lists.freeswitch.org/mailman/options/freeswitch-users" target="_blank">http://lists.freeswitch.org/mailman/options/freeswitch-users</a><br>
<a href="http://www.freeswitch.org" target="_blank">http://www.freeswitch.org</a><br>
<br></blockquote></div><br></div>