[asterisk-users] Need help understanding CDR

Asghar Mohammad asghar144 at gmail.com
Mon Mar 18 06:03:47 CDT 2013


hi,

00:00 -- Call Connected to asterisk -----> duration start here
00:01 -- welcome greeting starts --------> billisec start here
00:11 -- welcome greeting ends (10 sec wav file)
00:12 -- Call enters queue and at the same time rings on first available
extension
00:15 -- Call is answered by an agent
01:15 -- Conversation over, Call disconnected -- agents spoke for 60 sec
-------> both end here

duration = 01:15
bilsec = 01:14

duration start as soon as call arrived in asterisk.
bilsec start as soon as call answered.

exten s,1,Answer() --------> duration and bilsec start at same time because
you answered the call immidataly
exten s,n,Plaback(something)
exten s,n,Dial(agent)
exten s,n,Hangup --------> duration and billsec are same

exten s,1,Ringing(10) ------> duration start here
exten s,n,Answer() --------> bilsec start here
exten s,n,Plaback(something)
exten s,n,Dial(agent)
exten s,n,Hangup --------> duration and billsec end here

so billsec is 10 seconds less then duration

hope this will help you.

On Mon, Mar 18, 2013 at 6:29 AM, RSCL Mumbai <rscl.mumbai at gmail.com> wrote:

> I am using SIP.
>
> I am still a bit confused about "answered" & billed time.
>
> For example:
> 00:00 -- Call Connected to asterisk
> 00:01 -- welcome greeting starts
> 00:11 -- welcome greeting ends (10 sec wav file)
> 00:12 -- Call enters queue and at the same time rings on first available
> extension
> 00:15 -- Call is answered by an agent
> 01:15 -- Conversation over, Call disconnected -- agents spoke for 60 sec.
>
> In the given schematic what will be the "Answered" time and "billed" time.
>
> Thank you for the help in advance!!
>
>
>
>
>
>
>
>
>
> On Sun, Mar 17, 2013 at 10:06 PM, Asghar Mohammad <asghar144 at gmail.com>wrote:
>
>> "If you have analog FXO ports then the call is considered answered as
>> soon as dialing is completed" not always true if FXO configured properly it
>> should not send back answered as soon as dialed.
>>
>>
>> On Sun, Mar 17, 2013 at 5:29 PM, Eric Wieling <EWieling at nyigc.com> wrote:
>>
>>> If you have analog FXO ports then the call is considered answered as
>>> soon as dialing is completed.   This does not apply to SIP, PRI, or other
>>> technologies which support far end answer detection.
>>>
>>> -----Original Message-----
>>> From: asterisk-users-bounces at lists.digium.com [mailto:
>>> asterisk-users-bounces at lists.digium.com] On Behalf Of RSCL Mumbai
>>> Sent: Sunday, March 17, 2013 12:15 PM
>>> To: Asterisk Users Mailing List - Non-Commercial Discussion
>>> Subject: [asterisk-users] Need help understanding CDR
>>>
>>> Hi,
>>>
>>> Attached is a sample CDR.
>>>
>>> I need some help to understand the "billsec" column.
>>> PS: the time value in billsec & duration is same.
>>>
>>> With reference to the attached log, what does the 10 sec / 6 sec / 2 sec
>>> correspond to:
>>>
>>> (a) Time between call connection to asterisk and disconnection from
>>> asterisk?
>>> (b) Time after welcome greeting and before hangup -- the time the call
>>> rang on the extension?
>>> (c) Or any other scenario
>>>
>>> Thank you in advance.
>>>
>>> Best regards,
>>> Sans
>>>
>>> --
>>> _____________________________________________________________________
>>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
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>>
>>
>> --
>> _____________________________________________________________________
>> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>> New to Asterisk? Join us for a live introductory webinar every Thurs:
>>                http://www.asterisk.org/hello
>>
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>> To UNSUBSCRIBE or update options visit:
>>    http://lists.digium.com/mailman/listinfo/asterisk-users
>>
>
>
> --
> _____________________________________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> New to Asterisk? Join us for a live introductory webinar every Thurs:
>                http://www.asterisk.org/hello
>
> asterisk-users mailing list
> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-users
>
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