Re: [asterisk-users] Switched from Asterisk 1.8 to 13 - CDR ringtime now always zero (Joel)
>Hi, >On the other side.. There is a specific note regarding CDR behavior changes >from asterisk 12 onwards. So going from 1.8 to 13 means it applies to you. >Have a look at: >https://wiki.asterisk.org/wiki/display/AST/Upgrading+to+Asterisk+12 >And >https://wiki.asterisk.org/wiki/display/AST/Asterisk+12+CDR+Specification Hi Joel I've read the above in detail, and studied them, but I cannot find anything that explains the CDR behaviour I'm seeing in Asterisk 13. The only thing relevant is the following in the Upgrade Notes: --- The duration, billsec, start, answer, and end times now reflect the times associated with the current CDR for the channel, as opposed to a cumulative measurement of all CDRs for that channel. --- That is exactly where the problem lies, e. g. the above behaviour is exactly what is NOT happening... E. g. instead of the origination channel created by the AMI call we make as previously described, having its CDR written and done with, its CDR data for the originating call leg apparently gets propagated into the resulting outgoing call leg of the call to the AMI originate command. E. g. my origination channel's CDR is Start: 2019-01-11 08:22:07 Answer: 2019-01-11 08:22:12 End: 2019-01-11 08:24:09 and the associated outgoin channel's CDR is Start: 2019-01-11 08:22:12 Answer: 2019-01-11 08:22:12 End: 2019-01-11 08:24:09 e.g. it appears that the origination channel's pickup time of Answer: 2019-01-11 08:22:12 gets propagated into the outgoing channel's answer time of Answer: 2019-01-11 08:22:12. E. g. the above in the release notes is 180 degrees the opposite of how Asterisk 13 actually behaves in practice? Despite the above release note, it DOES appear in fact that CDR measurement is cumulative... very definitely not distinct Thanks for the reply! Regards Stefan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Switched from Asterisk 1.8 to 13 - CDR ringtime now always zero (Tony)
In article <009d01d4a7e0$af2e0a50$0d8a1ef0$@verishare.co.za>, Stefan Viljoen wrote: >> Regarding this I've read the specs linked to in detail, but I can find >> no mention anywhere of any change that implies or states that no ring >> time will be recorded anymore in Asterisk 13 and that all times in start and >> answer columns will now be equal for all calls. >> >> Can this be because I nowhere use the Answer() application in my dialplan >> when dialing out? >You shouldn't Answer() the originating channel before calling Dial() on it. >You should allow Dial() to propagate the answer, busy or other failure from >the destination channel back to the originating channel. Hi Tony Yes, that is exactly what I'm doing... no Answer() calls anywhere in the dialout parts of my dialplan, as detailed in my previous posts. >Is it possible that the setup part of the call (between initiation and answer) >is recorded in a separate CDR? Yes, that is correct, I get a separate CDR for the originate of the call to the agent's extension, and then a separate CDR for the call that then goes out to the client. The separate CDR for an originate event DOES appear to be correct, there is usually a second or two seconds worth of ringtime indicated, e. g. start and answer will vary (as outgoing calls did on 1.8) under 13: Start | Answer | End 2019-01-11 08:01:20 | 2019-01-11 08:01:22 | 2019-01-11 08:01:25 E. g. the agent originated by calling the AMI via our app at 2019-01-11 08:01:20, he picked up the phone at 2019-01-11 08:01:22 (e. g. he let it ring for 2 seconds) etc. This is exactly what I need to have, but not on the origination calls, the outgoing calls under Asterisk 13 - e. g. that there is a difference between the Start and Answer times, which = ringtime. For the origination calls, duration and billsec is correct as well, just as it was under 1.8 too. It is just with 13 that the origination calls are STILL correct, as they were under 1.8, but with 13 the outgoing calls now indicate no ringtime. Thanks for the reply! Stefan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Switched from Asterisk 1.8 to 13 - CDR ringtime now always zero (Joshua C. Colp)
On Wed, Jan 9, 2019, at 6:08 AM, Tony Mountifield wrote: > In article <009d01d4a7e0$af2e0a50$0d8a1ef0$@verishare.co.za>, > Stefan Viljoen wrote: > > Regarding this I've read the specs linked to in detail, but I can find no > > mention anywhere of any change that implies or > > states that no ring time will be recorded anymore in Asterisk 13 and that > > all times in start and answer columns will now > > be equal for all calls. > > > > Can this be because I nowhere use the Answer() application in my dialplan > > when dialing out? > > You shouldn't Answer() the originating channel before calling Dial() on it. > You should allow Dial() to propagate the answer, busy or other failure from > the destination channel back to the originating channel. > > Is it possible that the setup part of the call (between initiation and answer) > is recorded in a separate CDR? An excellent question. Unlike in the past versions calls can actually generate multiple CDRs because CDRs now represent the flow of communication between things. Providing the actual CDR records that were generated as well as console output would allow better understanding of what is going on. -- Joshua C. Colp Digium - A Sangoma Company | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com & www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Switched from Asterisk 1.8 to 13 - CDR ringtime now always zero (Joshua C. Colp)
In article <009d01d4a7e0$af2e0a50$0d8a1ef0$@verishare.co.za>, Stefan Viljoen wrote: > Regarding this I've read the specs linked to in detail, but I can find no > mention anywhere of any change that implies or > states that no ring time will be recorded anymore in Asterisk 13 and that all > times in start and answer columns will now > be equal for all calls. > > Can this be because I nowhere use the Answer() application in my dialplan > when dialing out? You shouldn't Answer() the originating channel before calling Dial() on it. You should allow Dial() to propagate the answer, busy or other failure from the destination channel back to the originating channel. Is it possible that the setup part of the call (between initiation and answer) is recorded in a separate CDR? Cheers Tony -- Tony Mountifield Work: t...@softins.co.uk - http://www.softins.co.uk Play: t...@mountifield.org - http://tony.mountifield.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Switched from Asterisk 1.8 to 13 - CDR ringtime now always zero (Joshua C. Colp)
Message: 2
Date: Mon, 07 Jan 2019 06:07:54 -0500
From: "Joshua C. Colp"
>> On Mon, Jan 7, 2019, at 3:04 AM, Stefan Viljoen wrote:
>> Hi guys
.
.
.
>> E. g. on 13, I see this (zero ringtime) for a call that I make to my
>> cellphone to test, with my cellphone ringing for at least 10 seconds
>> and ringing heard on the Yealink connected to the asterisk - e. g.
>> completely wrong:
>This is the way it is supposed to work[1], but it ultimately depends on your
>dialplan. Are you using Local channels? Are you doing Answer in the dialplan?
>What is the complete flow?
Hi Joshua
Thank you for the reply. I'll go read the spec in detail (which is probably
what I should have done in the first place anyway.)
Yes, I am using local channels and generating the calls via AJAM by calling the
Originate AMI / AJAM application. The "local" extension that is calling out is
defined as
local/@local
where is the extension number, e. g. 3509, 3175, or whatever.
So my AJAM Originate command is
ActionID=201901080814t4qn82v
Action=Originate
Channel=local/3916@local
Exten=0825588996
Context=local
Priority=1
CallerID=3916_ctd
Account=201901080814t4qn82v
ChannelID=201901080814t4qn82v
OtherChannelID=201901080814t4qn82vB
Variable=__CallLimit=3600
Async=true
I'm not calling the Answer application in the dialplan when dialing out.
Effectively the flow is the user clicks dial in our external application. This
sends the above AJAM command to Asterisk, with parameters as specified.
Asterisk originates the call on his extension as passed in the AJAM command, he
picks up and the call then goes into the local context and gets routed. The
STDOUT macro is called which does some prep (lots of 1.8 legacy stuff still in
here) and this then calls VCCALLOUT, which contains the call to the dial()
application.
VCCALLOUT calls
exten=>s,n(dialNoLimit),Dial(${chantouse}/${numtodial},60,TL(390:6:3))
${chantouse} will be something like SIP/sip-trunk-name
${numtodial} will be the target number, e. g. 27825588996
to dial out.
---
Detail:
My local context (simplified):
[local]
exten=>_082xxx,1,Macro(STDOUT,SIP/centra-out,27${EXTEN:1})
and the STDOUT macro, edited for brevity:
[macro-STDOUT]
;${ARG1} = channel
;${ARG2} = number
exten=>s,1,Macro(WAITCHANNEL)
exten=>s,n,Macro(WAITCDR)
exten=>s,n,Macro(VCRECORD,${MACRO_CONTEXT}X${CALLERID(num)}ACC${CHANNEL(accountcode)},${ARG2})
exten=>s,n(dodial),Macro(VCCALLOUT,${ARG1},${ARG2},${ARG3},${ARG4})
exten=>s,n,NoOp(Setting Userfield after call completion)
exten=>s,n,Set(CDR(userfield)=${MIXMONITOR_FILENAME})
exten=>s,dodial+101,Busy()
exten=>s,n,Hangup()
exten=>s,n,MacroExit
exten=>h,1,NoOp(Call hangup MACROSTDOUT)
exten=>h,n,UserEvent(RecordingToFile,Uniqueid: ${UNIQUEID},Channel:
${Channel},Filename: ${MIXMONITOR_FILENAME})
exten=>h,n,NoOp(Recorded to ${MIXMONITOR_FILENAME})
exten=>h,n,Set(CDR(userfield)=${MIXMONITOR_FILENAME})
exten=>h,n,NoOp(Account code is ${CHANNEL(accountcode)})
exten=>h,n,NoOp(call link var is ${call_Link})
exten=>h,n,GotoIf($["${CHANNEL(accountcode)}" != ""]?done)
exten=>h,n(setacc),Set(CHANNEL(accountcode)=${call_Link})
exten=>h,n(done),noOp(Call Completed)
and the VCCALLOUT macro:
[macro-VCCALLOUT] ;macro to dial numbers
; ${ARG1} Channel To Use
; ${ARG2} Number To Dial
; ${ARG3} FailOver Channel
; ${ARG4} FailOverNumber
exten=>s,n(setchan),Set(chantouse=${ARG1})
exten=>s,n,Set(numtodial=${ARG2})
exten=>s,n(makecall),GotoIf($["${timeLimit}" = ""]?dialNoLimit:dialLimit)
exten=>s,n(dialNoLimit),Dial(${chantouse}/${numtodial},60,TL(390:6:3))
exten=>s,n,NoOp(Dial Status: ${DIALSTATUS})
exten=>s,n,GoTo(s-${DIALSTATUS},1)
exten=>s,n(dialLimit),Dial(${chantouse}/${numtodial},60,TL(390:6:3))
exten=>s,n,NoOp(Dial Status: ${DIALSTATUS})
exten=>s,n,GoTo(s-${DIALSTATUS},1)
exten=>s,dialNoLimit+101,Goto(s-${DIALSTATUS},1)
exten=>s,dialLimit+101,Goto(s-${DIALSTATUS},1)
exten=>s,n(endcall),busy()
exten=>s,n,NoOp(Call Completed - setting userfield to recording)
exten=>s,n,Set(CDR(userfield)=${MIXMONITOR_FILENAME})
exten=>s,n,Hangup()
exten=>s,n,MacroExit
exten=>s-NOANSWER,1,goto(s,endcall)
exten=>s-CANCEL,1,goto(s,endcall)
exten=>s-BUSY,1,goto(s,endcall)
exten=>h,1,NoOp(Call Hungup)
exten=>h,n,UserEvent(RecordingToFile,Uniqueid: ${UNIQUEID},Channel:
${Channel},Filename: ${MIXMONITOR_FILENAME})
exten=>h,n,NoOp(Recorded to ${MIXMONITOR_FILENAME})
exten=>h,n,Set(CDR(userfield)=${MIXMONITOR_FILENAME})
exten=>h,n,NoOp(Account code is ${CHANNEL(accountcode)})
exten=>h,n,NoOp(call link var is ${call_Link})
exten=>h,n,GotoIf($["${CHANNEL(accountcode)}" != ""]?done)
exten=>h,n(setacc),Set(CHANNEL(accountcode)=${call_Link})
exten=>h,n(done),noOp(Call Completed)
---
Thank you very much for the reply!
Regards
Stefan
--
_
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
Check out the new Asterisk community forum at:
Re: [asterisk-users] Switched from Asterisk 1.8 to 13 - CDR ringtime now always zero (Joshua C. Colp)
Regarding this I've read the specs linked to in detail, but I can find no
mention anywhere of any change that implies or states that no ring time will be
recorded anymore in Asterisk 13 and that all times in start and answer columns
will now be equal for all calls.
Can this be because I nowhere use the Answer() application in my dialplan when
dialing out?
-Original Message-
From: Stefan Viljoen
Sent: Tuesday, 08 January 2019 08:49
To: 'asterisk-users@lists.digium.com'
Subject: RE: Re: Switched from Asterisk 1.8 to 13 - CDR ringtime now always
zero (Joshua C. Colp)
Message: 2
Date: Mon, 07 Jan 2019 06:07:54 -0500
From: "Joshua C. Colp"
>> On Mon, Jan 7, 2019, at 3:04 AM, Stefan Viljoen wrote:
>> Hi guys
.
.
.
>> E. g. on 13, I see this (zero ringtime) for a call that I make to my
>> cellphone to test, with my cellphone ringing for at least 10 seconds
>> and ringing heard on the Yealink connected to the asterisk - e. g.
>> completely wrong:
>This is the way it is supposed to work[1], but it ultimately depends on your
>dialplan. Are you using Local channels? Are you doing Answer in the dialplan?
>What is the complete flow?
Hi Joshua
Thank you for the reply. I'll go read the spec in detail (which is probably
what I should have done in the first place anyway.)
Yes, I am using local channels and generating the calls via AJAM by calling the
Originate AMI / AJAM application. The "local" extension that is calling out is
defined as
local/@local
where is the extension number, e. g. 3509, 3175, or whatever.
So my AJAM Originate command is
ActionID=201901080814t4qn82v
Action=Originate
Channel=local/3916@local
Exten=0825588996
Context=local
Priority=1
CallerID=3916_ctd
Account=201901080814t4qn82v
ChannelID=201901080814t4qn82v
OtherChannelID=201901080814t4qn82vB
Variable=__CallLimit=3600
Async=true
I'm not calling the Answer application in the dialplan when dialing out.
Effectively the flow is the user clicks dial in our external application. This
sends the above AJAM command to Asterisk, with parameters as specified.
Asterisk originates the call on his extension as passed in the AJAM command, he
picks up and the call then goes into the local context and gets routed. The
STDOUT macro is called which does some prep (lots of 1.8 legacy stuff still in
here) and this then calls VCCALLOUT, which contains the call to the dial()
application.
VCCALLOUT calls
exten=>s,n(dialNoLimit),Dial(${chantouse}/${numtodial},60,TL(390:6:3))
${chantouse} will be something like SIP/sip-trunk-name ${numtodial} will be the
target number, e. g. 27825588996
to dial out.
---
Detail:
My local context (simplified):
[local]
exten=>_082xxx,1,Macro(STDOUT,SIP/centra-out,27${EXTEN:1})
and the STDOUT macro, edited for brevity:
[macro-STDOUT]
;${ARG1} = channel
;${ARG2} = number
exten=>s,1,Macro(WAITCHANNEL)
exten=>s,n,Macro(WAITCDR)
exten=>s,n,Macro(VCRECORD,${MACRO_CONTEXT}X${CALLERID(num)}ACC${CHANNEL(accountcode)},${ARG2})
exten=>s,n(dodial),Macro(VCCALLOUT,${ARG1},${ARG2},${ARG3},${ARG4})
exten=>s,n,NoOp(Setting Userfield after call completion)
exten=>s,n,Set(CDR(userfield)=${MIXMONITOR_FILENAME})
exten=>s,dodial+101,Busy()
exten=>s,n,Hangup()
exten=>s,n,MacroExit
exten=>h,1,NoOp(Call hangup MACROSTDOUT)
exten=>h,n,UserEvent(RecordingToFile,Uniqueid: ${UNIQUEID},Channel:
${Channel},Filename: ${MIXMONITOR_FILENAME}) exten=>h,n,NoOp(Recorded to
${MIXMONITOR_FILENAME})
exten=>h,n,Set(CDR(userfield)=${MIXMONITOR_FILENAME})
exten=>h,n,NoOp(Account code is ${CHANNEL(accountcode)}) exten=>h,n,NoOp(call
link var is ${call_Link}) exten=>h,n,GotoIf($["${CHANNEL(accountcode)}" !=
""]?done)
exten=>h,n(setacc),Set(CHANNEL(accountcode)=${call_Link})
exten=>h,n(done),noOp(Call Completed)
and the VCCALLOUT macro:
[macro-VCCALLOUT] ;macro to dial numbers ; ${ARG1} Channel To Use ; ${ARG2}
Number To Dial ; ${ARG3} FailOver Channel ; ${ARG4} FailOverNumber
exten=>s,n(setchan),Set(chantouse=${ARG1})
exten=>s,n,Set(numtodial=${ARG2})
exten=>s,n(makecall),GotoIf($["${timeLimit}" = ""]?dialNoLimit:dialLimit)
exten=>s,n(dialNoLimit),Dial(${chantouse}/${numtodial},60,TL(390:6:3))
exten=>s,n,NoOp(Dial Status: ${DIALSTATUS})
exten=>s,n,GoTo(s-${DIALSTATUS},1)
exten=>s,n(dialLimit),Dial(${chantouse}/${numtodial},60,TL(390:6:3))
exten=>s,n,NoOp(Dial Status: ${DIALSTATUS})
exten=>s,n,GoTo(s-${DIALSTATUS},1)
exten=>s,dialNoLimit+101,Goto(s-${DIALSTATUS},1)
exten=>s,dialLimit+101,Goto(s-${DIALSTATUS},1)
exten=>s,n(endcall),busy()
exten=>s,n,NoOp(Call Completed - setting userfield to recording)
exten=>s,n,Set(CDR(userfield)=${MIXMONITOR_FILENAME})
exten=>s,n,Hangup()
exten=>s,n,MacroExit
exten=>s-NOANSWER,1,goto(s,endcall)
exten=>s-CANCEL,1,goto(s,endcall)
exten=>s-BUSY,1,goto(s,endcall)
exten=>h,1,NoOp(Call Hungup)
exten=>h,n,UserEvent(RecordingToFile,Uniqueid: ${UNIQUEID},Channel:
${Channel},Filename: ${MIXMONITOR_FILENAME}) exten=>h,n,NoOp(Recorded to
Re: [asterisk-users] Switched from Asterisk 1.8 to 13 - CDR ringtime now always zero
On Mon, Jan 7, 2019, at 3:04 AM, Stefan Viljoen wrote: > Hi guys > > A few months ago I upgraded most of my Asterisk servers to 13 from 1.8. > I've still got about 25% of my servers on 1.8. > > I've since noticed that ringtime on Asterisk 13 - the time difference > between "start" and "answer" in the CDR record for any call, and > between "duration" and "billsec" - has completely disappeared. E. g. > the two times and two durations are now the same for all outgoing calls > made on Asterisk 13. > > On 1.8 the time difference between "start" and "answer" and "duration" > and "billsec" was always my ring time - e. g. if I phone out to my > cellphone from one of my 1.8 servers, the amount of seconds the call > rings on my cell in my 1.8 instances is the difference between "start" > and "answer" in the 1.8-generated CDR record, and the difference > between "duration" and "billsec". > > E. g. on 13, I see this (zero ringtime) for a call that I make to my > cellphone to test, with my cellphone ringing for at least 10 seconds > and ringing heard on the Yealink connected to the asterisk - e. g. > completely wrong: This is the way it is supposed to work[1], but it ultimately depends on your dialplan. Are you using Local channels? Are you doing Answer in the dialplan? What is the complete flow? [1] https://wiki.asterisk.org/wiki/display/AST/Asterisk+12+CDR+Specification -- Joshua C. Colp Digium - A Sangoma Company | Senior Software Developer 445 Jan Davis Drive NW - Huntsville, AL 35806 - US Check us out at: www.digium.com & www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Switched from Asterisk 1.8 to 13 - CDR ringtime now always zero
Hi guys A few months ago I upgraded most of my Asterisk servers to 13 from 1.8. I've still got about 25% of my servers on 1.8. I've since noticed that ringtime on Asterisk 13 - the time difference between "start" and "answer" in the CDR record for any call, and between "duration" and "billsec" - has completely disappeared. E. g. the two times and two durations are now the same for all outgoing calls made on Asterisk 13. On 1.8 the time difference between "start" and "answer" and "duration" and "billsec" was always my ring time - e. g. if I phone out to my cellphone from one of my 1.8 servers, the amount of seconds the call rings on my cell in my 1.8 instances is the difference between "start" and "answer" in the 1.8-generated CDR record, and the difference between "duration" and "billsec". E. g. on 13, I see this (zero ringtime) for a call that I make to my cellphone to test, with my cellphone ringing for at least 10 seconds and ringing heard on the Yealink connected to the asterisk - e. g. completely wrong: Start | Answer | End | dur.| billsec 2019-01-07 08:13:49 | 2019-01-07 08:13:49 | 2019-01-07 08:14:35 | 45 | 45 when it should have shown duration 45 billsec 30, 'cause I let it ring 10 seconds before answering my cell / mobile. On 1.8, I see this for a call that I make to my cellphone to test, with the phone ringing for 10 seconds and ringing heard on the Yealink connected to the asterisk - e. g. correct: Start | Answer | End | dur. | billsec 2019-01-07 08:26:43 | 2019-01-07 08:26:53 | 2019-01-07 08:29:25 | 162 | 152 e. g. it shows clearly that the call was ringing for 10 seconds before I answered my cell / mobile. My billing logic can thus tell that on Asterisk 1.8 there was 10 seconds of ringtime, but on Asterisk 13 it says that there was 0 seconds of ringtime - which in reality was also 10 seconds, but is no longer written as such into the CDR on Asterisk 13. So on all my Asterisk 13 instances all my outgoing calls are "suddenly" immediately answered by whomever I'm calling, and clients' landlines / mobiles never ring at all, they just instantly answer... which is obviously incorrect. How can I get Asterisk 13 to behave like Asterisk 1.8 as regards CDRs, e. g. still recording ring time and not just make all call start and call answer times, and all durations and billsec counts, exactly the same? I'm using the same SIP trunk provider with 13 as with 1.8, and in all cases the same operating system (Centos 7, upgraded to date) with both my 1.8 and 13 instances, and almost always the same physical hardware, same network, etc. The CDRs are logging locally on the Asterisk itself in both my 1.8 and 13.22.0 instances, unixODBC connections to Percona 5.6 running locally on the same box. So why is there no longer any ringing recorded in CDRs Asterisk 13 generates? It seems some element of logic changed in the CDR engine, but I cannot find any references to such... anybody got any help or pointers? The holy grail would be to have Asterisk 13 record ringing time on its CDR records, e. g. have a variation between the time a call is shown as having started and the time it is shown as having answered, e. g. the time spent in ringing waiting for the callee to pick up. So 1.8 CDR ringtime behaviour - but in 13. Any ideas? Thanks, Stefan -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- Check out the new Asterisk community forum at: https://community.asterisk.org/ New to Asterisk? Start here: https://wiki.asterisk.org/wiki/display/AST/Getting+Started asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
