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

Wed Mar 17 12:41:36 PDT 2010

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/
>>>> ClueCon http://www.cluecon.com/
>>>> Twitter: http://twitter.com/FreeSWITCH_wire
>>>>
>>>> AIM: anthm
>>>> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
>>>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>>>> IRC: irc.freenode.net #freeswitch
>>>>
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>>>>
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>>>
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>>>
>>
>>
>> --
>> Anthony Minessale II
>>
>> FreeSWITCH http://www.freeswitch.org/
>> ClueCon http://www.cluecon.com/
>> Twitter: http://twitter.com/FreeSWITCH_wire
>>
>> AIM: anthm
>> MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
>> GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
>> IRC: irc.freenode.net #freeswitch
>>
>> FreeSWITCH Developer Conference
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>> iax:guest at conference.freeswitch.org/888
>> googletalk:conf+888 at conference.freeswitch.org<googletalk%3Aconf%2B888 at conference.freeswitch.org>
>> pstn:+19193869900
>>
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>>
>
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>

-- 
Anthony Minessale II

FreeSWITCH http://www.freeswitch.org/
ClueCon http://www.cluecon.com/
Twitter: http://twitter.com/FreeSWITCH_wire

AIM: anthm
MSN:anthony_minessale at hotmail.com <MSN%3Aanthony_minessale at hotmail.com>
GTALK/JABBER/PAYPAL:anthony.minessale at gmail.com<PAYPAL%3Aanthony.minessale at gmail.com>
IRC: irc.freenode.net #freeswitch

FreeSWITCH Developer Conference
sip:888 at conference.freeswitch.org <sip%3A888 at conference.freeswitch.org>
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