[asterisk-users] Need help understanding CDR

[RSCL Mumbai]

Thank you every one.
Now I understand why I was confused.
I have always been using Asterisk in an Inbound environment.
Hence my thought were misaligned wrt "answered" & "billed".
Now I understand. Thank you all!!

Is there anyway to capture the time for conversation, IVR, hold etc etc.
If not inbuilt into AsteriskCDR, by way of any patch or Dialplan or any 3rd
party application, more suitable for an Inbound environment.

It would help a lot if I could capture fragmented distribution of time per
call -- time in IVR, Queue, Call etc.

Regards,
Sans









On Mon, Mar 18, 2013 at 4:33 PM, Asghar Mohammad <asghar144 at gmail.com>wrote:

> 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
>>>>
>>>> --
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>>>>
>>>
>>>
>>> --
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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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