i'll try again progress_media and progress are both the same for your purposes you will only have exactly one of the 2 set. its the exact time it started ringing. for our sake, if we have either of them we will take the one that is not zero and call it the progress time. so you have the created time, the progress time , the answered time and the hangup time you always have created time and hangup time but not always progress and answered so talk time is (hangup_time - (answered_time unless its 0)) or if answered_time is 0 there is no talk time ring time is: (answered time unless its zero otherwise hangup_time) - created_time edit conf/autoload_configs/cdr_csv.conf.xml uncomment: <param name="debug" value="true"/> and restart FS now when you hangup all your calls you will see the values you have to work with in the cdr if you are tying to do billing inside some js script you are using, stop now, and go learn about cdrs. On Wed, Mar 17, 2010 at 2:18 PM, Fraser Redmond <fraserredmond@gmail.com>wrote:
Sorry - newbie question: what is the 180 or 183? Can I manipulate that any way?
I've realised since my last email that even having the progress_time doesn't help without the progress_media_time or some way to find when the ringing stopped - getting the server timestamp wouldn't happen until the call ended, at which point I can't split the ring time from the talk time.
Surely there can't be something that Asterisk handles well that FreeSwitch doesn't? ;-)
I'm looking into the CDR now - might have to parse the log files to get the end time from there :-(
Cheers, Fraser
On Wed, Mar 17, 2010 at 6:52 PM, Anthony Minessale < anthony.minessale@gmail.com> wrote:
progress_media_time is only set if you get a 183 if you get a 180 with no media it uses progress_time instead
as long as you have one or the other you have the timestamp of when it started ringing.
On Wed, Mar 17, 2010 at 12:17 PM, Fraser Redmond <fraserredmond@gmail.com
wrote:
Yeah, that's what I had expected, and was why I was confused.
I'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.
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's Created & Answered.
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. The bad news is that the Progress-Media-Time is always blank
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'd be nice to do it properly.
Any other ideas? Is there any channel-variables I can set that would be worth playing with? I'm currently only setting: ignore_early_media=true,hangup_after_bridge=false,continue_on_fail=true
Cheers, Fraser
On Wed, Mar 17, 2010 at 4:57 PM, Anthony Minessale < anthony.minessale@gmail.com> wrote:
even better you have the progress and progress_media timestamps too so you can measure from the instance you got the first ringing indication so then you can also measure how long it took to start ringing too (PDD)
On Wed, Mar 17, 2010 at 10:46 AM, Brian West <brian@freeswitch.org>wrote:
Call start answer time minus call start time = ring time in the CDR
/b
On Mar 16, 2010, at 5:21 PM, Fraser Redmond wrote:
I'm converting a call-center app from Asterisk to FreeSwitch (using xml and javascript dialplans) and I think I'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.
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