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.mum...@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 <asghar...@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 <ewiel...@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-boun...@lists.digium.com [mailto: >>> asterisk-users-boun...@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 -- >>> 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 >>> >> >> >> -- >> _____________________________________________________________________ >> -- 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 >> > > > -- > _____________________________________________________________________ > -- 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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