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 -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20130317/56440ae6/attachment.htm> -------------- next part -------------- +---------------------+------------------------------+--------------+-------------+--------------------+---------------------------------------------------------------------+-----------+-------------------------------------------------------------------+---------+-------------+-------------+ | calldate | clid | src | dst | dcontext | dstchannel | lastapp | lastdata | billsec | disposition | dnid | +---------------------+------------------------------+--------------+-------------+--------------------+---------------------------------------------------------------------+-----------+-------------------------------------------------------------------+---------+-------------+-------------+ | 2013-03-15 17:52:53 | "19170002018" <8130006555> | 8130006555 | s | app-announcement-4 | | Playback | custom/Welcome,noanswer | 10 | ANSWERED | 19170002018 | | 2013-03-12 16:32:05 | "19170002018" <2810007178> | 2810007178 | s | app-announcement-4 | | Playback | custom/Welcome,noanswer | 6 | ANSWERED | 19170002018 | | 2013-03-12 16:31:55 | "19170002018" <2810007178> | 2810007178 | s | app-announcement-4 | | Playback | custom/Welcome,noanswer | 2 | ANSWERED | 19170002018 | +---------------------+------------------------------+--------------+-------------+--------------------+---------------------------------------------------------------------+-----------+-------------------------------------------------------------------+---------+-------------+-------------+
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
hi, billsec is time in seconds after call has answered, duration is total time in seconds of call. as your calls answered imidiatly your billsec and duration is almost same. On Sun, Mar 17, 2013 at 5:14 PM, RSCL Mumbai <rscl.mumbai at gmail.com> wrote:> 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 >-------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20130317/6e42db3b/attachment.htm>
"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 -- > 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 >-------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20130317/f4f9b3d5/attachment.htm>
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 -- >> 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 >-------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20130318/6176cba1/attachment.htm>
Top replying ... In the CDR you have two fields, "duration" and "billed". "Duration" is the total time from "Dial" command to end of calls. It is the time the "Dial" command is running. "Billed" is the time from when the other party answered and the end of the call. In your example, duration and billsec will differ for just a second, the time from the "Call Connected to asterisk" and the "Welcome greeting starts". Leandro 2013/3/18 RSCL Mumbai <rscl.mumbai at gmail.com>:> 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 -- >>> 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
Answered means your call answered by answer application or by ivr or moh kind of dialplan. Here, call connected to asterisk means your calls starts ringing and its start duration field counter. As soon as you answer the call, i,e, start playing file or moh its start counter of billsec field. In below scenario, answer time and billed time both is same i.e 10 (wav) + 60 (agent). Also duration field in CDR is same i.e 60 or 61 sec as you directly start playing greetings. What i understand from below scenario and if i do not misunderstood, you want time of call answered by Agent i.e. 60 sec time in you billsec. But you are getting confused on answered word, it does not mean answer by agent. If you want agent talk time with your customer, then i think CDR is not provided the same. Regards, Bharat Lalcheta On Mon, Mar 18, 2013 at 10:59 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 -- >>> 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 >-- Bharat Lalcheta -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20130318/17590954/attachment.htm>
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 -- >>> 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 >-------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20130318/0458ee37/attachment.htm>
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 >>>> >>>> -- >>>> _____________________________________________________________________ >>>> -- 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 >> > > > -- > _____________________________________________________________________ > -- 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 >-------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20130318/a6b63c9f/attachment-0001.htm>
You can add custom fields in the CDR, so your dialplan can store start time, end time and duration whenever you like. Just use something like the Set(CDR(customfield)=100); Leandro 2013/3/18 RSCL Mumbai <rscl.mumbai at gmail.com>:> 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 >>>>> >>>>> -- >>>>> _____________________________________________________________________ >>>>> -- 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 >> >> >> >> -- >> _____________________________________________________________________ >> -- 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
hi, try Asterisk manager or AGI. On Mon, Mar 18, 2013 at 12:36 PM, RSCL Mumbai <rscl.mumbai at gmail.com> wrote:> 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 >>>>> >>>>> -- >>>>> _____________________________________________________________________ >>>>> -- 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 >>> >> >> >> -- >> _____________________________________________________________________ >> -- 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 >-------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20130318/22e55fa9/attachment.htm>
If you use application Queue to pass the calls to the agents you will have the advantage of having the queue log available which will give you lots of detailed information. Regards Ish On Mon, 2013-03-18 at 17:06 +0530, RSCL Mumbai wrote:> 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 > > > -- > _____________________________________________________________________ > -- 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 > > > > -- > _____________________________________________________________________ > -- 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-- Ishfaq Malik <ish at pack-net.co.uk> Department: VOIP Support Company: Packnet Limited t: +44 (0)845 004 4994 f: +44 (0)161 660 9825 e: ish at pack-net.co.uk w: http://www.pack-net.co.uk Registered Address: PACKNET LIMITED, 2A ENTERPRISE HOUSE, LLOYD STREET NORTH, MANCHESTER SCIENCE PARK, MANCHESTER, M156SE COMPANY REG NO. 04920552