[Freeswitch-users] How long did the phone ring?

Wed Mar 17 13:10:33 PDT 2010

Thanks, I suspect this is what I really needed:

> if you are tying to do billing inside some js script you are using, stop
now, and go learn about cdrs.

I guess there's more for me to un-learn from Asterisk than I'd thought :-(

Cheers,
Fraser

On Wed, Mar 17, 2010 at 7:41 PM, Anthony Minessale <
anthony.minessale at gmail.com> wrote:

> 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 at 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 at 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 at 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 at 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 at 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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>>>>>
>>>>>
>>>>>
>>>>> --
>>>>> Anthony Minessale II
>>>>>
>>>>> FreeSWITCH http://www.freeswitch.org/
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>>>>>
>>>>> AIM: anthm
>>>>> MSN:anthony_minessale at hotmail.com<MSN%3Aanthony_minessale at hotmail.com>
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>>>>
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>>>
>>>
>>> --
>>> Anthony Minessale II
>>>
>>> FreeSWITCH http://www.freeswitch.org/
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>>>
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>>
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>
>
> --
> Anthony Minessale II
>
> FreeSWITCH http://www.freeswitch.org/
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>
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>
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