Thanks, I suspect this is what I really needed:<br><br>&gt; if you are tying to do billing inside some js script you are 
using, stop now, and go learn about cdrs.<br><br>I guess there&#39;s more for me to un-learn from Asterisk than I&#39;d thought :-(<br><br clear="all">Cheers,<br>Fraser<br><br><br>
<br><br><div class="gmail_quote">On Wed, Mar 17, 2010 at 7:41 PM, Anthony Minessale <span dir="ltr">&lt;<a href="mailto:anthony.minessale@gmail.com">anthony.minessale@gmail.com</a>&gt;</span> wrote:<br><blockquote class="gmail_quote" style="margin: 0pt 0pt 0pt 0.8ex; border-left: 1px solid rgb(204, 204, 204); padding-left: 1ex;">

i&#39;ll try again<br><br>progress_media and progress are both the same for your purposes<br>you will only have exactly one of the 2 set.  its the exact time it started ringing.<br><br>for our sake, if we have either of them we will take the one that is not zero and call it the progress time.<br>


<br>so you have the created time, the progress time , the answered time and the hangup time<br><br><br>you always have created time and hangup time but not always progress and answered<br><br>so talk time is (hangup_time - (answered_time unless its 0)) or if answered_time is 0 there is no talk time<br>


ring time is: <br>         (answered time unless its zero otherwise hangup_time) - created_time<br><br><br>edit conf/autoload_configs/cdr_csv.conf.xml<br><br>uncomment:<br><br>&lt;param name=&quot;debug&quot; value=&quot;true&quot;/&gt;<br>


<br>and restart FS<br><br><br>now when you hangup all your calls you will see the values you have to work with in the cdr<br><br>if you are tying to do billing inside some js script you are using, stop now, and go learn about cdrs.<div>

<div></div><div class="h5"><br>
<br><br><br><br><br><div class="gmail_quote">On Wed, Mar 17, 2010 at 2:18 PM, Fraser Redmond <span dir="ltr">&lt;<a href="mailto:fraserredmond@gmail.com" target="_blank">fraserredmond@gmail.com</a>&gt;</span> wrote:<br><blockquote class="gmail_quote" style="margin: 0pt 0pt 0pt 0.8ex; border-left: 1px solid rgb(204, 204, 204); padding-left: 1ex;">


Sorry - newbie question: what is the 180 or 183? Can I manipulate that any way?<br><br>I&#39;ve realised since my last email that even having the progress_time doesn&#39;t help without the progress_media_time or some way to find when the ringing stopped - getting the server timestamp wouldn&#39;t happen until the call ended, at which point I can&#39;t split the ring time from the talk time.<br>




<br>Surely there can&#39;t be something that Asterisk handles well that FreeSwitch doesn&#39;t? ;-)<br><br>I&#39;m looking into the CDR now - might have to parse the log files to get the end time from there :-(<br><br clear="all">




Cheers,<br><font color="#888888">Fraser</font><div><div></div><div><br><br><br>
<br><br><div class="gmail_quote">On Wed, Mar 17, 2010 at 6:52 PM, 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: 0pt 0pt 0pt 0.8ex; border-left: 1px solid rgb(204, 204, 204); padding-left: 1ex;">




progress_media_time is only set if you get a 183<br>if you get a 180 with no media it uses progress_time instead <br><br>as long as you have one or the other you have the timestamp of when it started ringing.<div><div></div>




<div><br><br><br><br>
<br><br><div class="gmail_quote">On Wed, Mar 17, 2010 at 12:17 PM, Fraser Redmond <span dir="ltr">&lt;<a href="mailto:fraserredmond@gmail.com" target="_blank">fraserredmond@gmail.com</a>&gt;</span> wrote:<br><blockquote class="gmail_quote" style="margin: 0pt 0pt 0pt 0.8ex; border-left: 1px solid rgb(204, 204, 204); padding-left: 1ex;">





Yeah, that&#39;s what I had expected, and was why I was confused.<br><br>I&#39;ve done some more tests, and tracked 
the results a little closer in a spreadsheet, and it seems that when I call the extension
 directly from another sipphone the  Caller-Channel-Answered-Time minus 
Caller-Channel-Created-Time  matches the ring time, so that scenario is fine.<br>
<br>
When I call into an IVR with the javascript dialplan and then create the
 new session and bridge them that way, the Answered-Time and Created-Time, that are reported after the call ends are reported on the A-leg&#39;s Created &amp; Answered. <br><br>The good news is that the Progress-Time is reported on when the B-leg started ringing, so I can know when the call started ringing.<br>







The bad news is that the Progress-Media-Time is always blank<br>
<br>I can take the Progress-Time and compare it to the system-clock, and that should generally be accurate to a second or so, which is better than nothing, but it&#39;d be nice to do it properly.<br><br>Any other ideas? Is there any channel-variables I can set that would be worth playing with? I&#39;m currently only setting:<br>







ignore_early_media=true,hangup_after_bridge=false,continue_on_fail=true<br><br clear="all">Cheers,<br><font color="#888888">Fraser</font><div><div></div><div><br><br><br>
<br><br><div class="gmail_quote">On Wed, Mar 17, 2010 at 4:57 PM, 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: 0pt 0pt 0pt 0.8ex; border-left: 1px solid rgb(204, 204, 204); padding-left: 1ex;">







even better you have the progress and progress_media timestamps too<br>so you can measure from the instance you got the first ringing indication<br>so then you can also measure how long it took to start ringing too (PDD)<br>








<br><br><div class="gmail_quote">On Wed, Mar 17, 2010 at 10:46 AM, Brian West <span dir="ltr">&lt;<a href="mailto:brian@freeswitch.org" target="_blank">brian@freeswitch.org</a>&gt;</span> wrote:<br><blockquote class="gmail_quote" style="margin: 0pt 0pt 0pt 0.8ex; border-left: 1px solid rgb(204, 204, 204); padding-left: 1ex;">








Call start answer time minus call start time = ring time in the CDR<br>
<font color="#888888"><br>
/b<br>
</font><div><div><br>
On Mar 16, 2010, at 5:21 PM, Fraser Redmond wrote:<br>
<br>
&gt; I&#39;m converting a call-center app from Asterisk to FreeSwitch (using xml and javascript dialplans) and I think I&#39;ve worked out how to do nearly everything, except for tracking one important metric: How long the phone rang before an agent picked it up.<br>









<br>
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