Asterisk Call Log not registering Call Corectly

Can anyone help me or have some idea. The Call Log menu is not working correctly. If I call with my SoftPhone the Sipura 3102 on extension and if I wait few seconds and after that answer the call, the call is regisered that is Answered but also calculate the seconds for ringing while the phone station is not pick up. If you don't answered then is OK because it whos that there is no Answer. If I try the billing solution I don't want to get this seconds for ringing calculated.
I also tried with adding Sipura 3102 like trunk and from the first time when the line is pickup I get seconds in call log.
I now that this is freePBX software but I hope that can be used for Voip provider or can anyone sugest me what software to use who is registering the Call Log correctly.

If someone have solution please respond.

Hi
Welcome to Voipuser, Hopefuly we will be ale to clar this up quickly.
In the CDR logs there are two fields for the call, CallDuation and Billiable seconds. The first includes ringing time, The second is the time of the call excluding ringing.

You need to bill on the second.

Quote:

I now that this is freePBX software but I hope that can be used for Voip provider or can anyone sugest me what software to use who is registering the Call Log correctly.

Im not sure what you mean here ? But freepbx is not a billing solution, Astbill or any of the others are.

Ian

I now that I can get data from CDR for the billing if the call is answered or not. But if you have setup Sipura 3102 like sip trunk and you make +389 to be prefix for geting active the sip trunk and after that you dial some number. In the same moment when the line is pick up to dial it's calculating like the call was picked up and is registered like ANSWERED and if the caled party not answer the call the call is billed. Is there a way to setup the asterisk to bill if the called party pick up in some way with tones or I' not sure how the PBX's are working. I think that only way to do this is with ISDN HFC modems or special digital cards because I think all analog FXO cards will register in the CDR that this trunk is active event if the called party didn't pick up the phone. How actualy the VOIP providers configure the billing or what type of hardware did they use.
Thanks.

Hi

Ok I see what you are saying now.

The reason for this is that 99% of Alog PSTN lines do not provide answer supervision in their inital configured state. That is a signal or reversal of the line when the far end has answered, Combined with the fact that much of the Alog equipment also doesnt support it, Means it will show answered as soon as it goes off hook.

The only way to get reliable billing is to use Digital links either BRI or PRI or use Voip trunks such as IAX or SIP. Using Alog lines for ANY commercial venture is not a good idea.

Ian

Thanks for the solution it was very good idea but I did sucesfuly installed HCF-S ISDN card and I'm able to dial via this card but also the Call Log registration is not correct because when I type the number and click call the call is registered ANSWERED. I'm not sure that this is very good solution for VOIP provider, doe's anyone else have some idea what type of hardware to use for the TRUNKs.

Thanks very much.

Hi

Could you post the raw output from the Master.csv for this call. What you are saying doesnt sound correct and is not what I have experianced.

As to what trunks to use, How many trunks to you need ? For a gateway you need at least a PRI connection. or use a voip gateway to terminate your traffic.

Ian

This is my MAster.csv you are right the call is correctly registred for billing.

"","350","9070707535","from-internal","350","SIP/350-a66d","Zap/1-1","Busy","","2007-01-26 21:19:02",,"2007-01-26 21:19:15",13,0,"BUSY","DOCUMENTATION"
"","350","93095874","from-internal","350","SIP/350-44e3","Zap/1-1","Dial","ZAP/g0/3095874|120|W","2007-01-26 21:23:04","2007-01-26 21:23:10","2007-01-26 21:23:24",20,14,"ANSWERED","DOCUMENTATION"
"","350","93095874","from-internal","350","SIP/350-6a32","Zap/1-1","Dial","ZAP/g0/3095874|120|W","2007-01-26 21:23:31","2007-01-26 21:23:44","2007-01-26 21:24:06",35,22,"ANSWERED","DOCUMENTATION"
"","350","93095874","from-internal","350","SIP/350-1711","Zap/1-1","Dial","ZAP/g0/3095874|120|W","2007-01-26 21:24:12","2007-01-26 21:24:19","2007-01-26 21:24:37",25,18,"ANSWERED","DOCUMENTATION"

"","350","93095874","from-internal","350","SIP/350-9fe1","Zap/1-1","ResetCDR","w","2007-01-27 14:54:52",,"2007-01-27 14:55:00",8,0,"NO ANSWER","DOCUMENTATION"
"","350","93095874","from-internal","350","SIP/350-b502","Zap/1-1","ResetCDR","w","2007-01-27 14:56:13",,"2007-01-27 14:56:33",20,0,"NO ANSWER","DOCUMENTATION"
"","350","93095874","from-internal","350","SIP/350-3059","Zap/1-1","ResetCDR","w","2007-01-27 14:56:56","2007-01-27 14:57:21","2007-01-27 14:57:26",30,5,"ANSWERED","DOCUMENTATION"

From 30 seconds call the 25sec-ringing and 5 seconds answer. This is good for billing.
Thank you very much.

Hi

Im glad we got there in the end.

This info should also be in the mysql database as well.

HTH Ian

You should use asterisk CSV-file, or use other billing sysytem (like a2b).

Chang PHP-files, so they now are showing only talk time (on BUSY or NOT ANSWERED there is now 0 sec):

In folder "/var/www/html/admin/cdr" change 4 PHP-files:
'call-comp.php'
'call-daily-load.php'
'call-last-month.php'
'call-log.php'

Changed word 'duration' (CASE SENSITIVE) on word 'billsec'

'billsec'--in asterisk CDR is talk time

what should i edit and from where?

I have Asterisk@Home version 2.2

thank you.

m_scheirer :

You should use asterisk CSV-file, or use other billing sysytem (like a2b). Chang PHP-files, so they now are showing only talk time (on BUSY or NOT ANSWERED there is now 0 sec): In folder "/var/www/html/admin/cdr" change 4 PHP-files: 'call-comp.php' 'call-daily-load.php' 'call-last-month.php' 'call-log.php' Changed word 'duration' (CASE SENSITIVE) on word 'billsec' 'billsec'--in asterisk CDR is talk time

As Ask by the original poster of this topic.

I am asking again,I am using Asterisk@Home version 2.2.

Smae the Call Details shows false billing Ringing time also counted.

origingal voip provider counts 5 sec then Asterisk@home count 9 seconds (ringing time also counts.

I am new to asterisk world.

I hope i can get better answer and also solution.

mani.

Hi Mani

You will have to post an example and line types etc.

Please post the raw csv file of a call.

Ian

ianplain :

Hi Mani You will have to post an example and line types etc. Please post the raw csv file of a call. Ian

Sorry to ask but as i am truely new to asterisk, where should i find the rw CSV

i can see pdf and CSV file bottom of CAll history in Asterisk @home

but it show false bill only (mean Ring or silent time also counted in it)

Asterisk CDR

calldate channel src clid dst disposition 2007-05-30 17:16:29 SIP/6000-098db780 6000 6000 004420x xxx xxxx ANSWERED 6

i am using asterisk@home set Sip trunk to it, and then set extesion to Linksys or x-lite .


Where i should see in details ,i tried to change the Call details php's as posted in this post ,i think am doing something wrong,pls guide me, do tell me the commands as i am new student to asterisk

mani

You need to look at the raw data that is held in /var/log/asterisk/cdr-csv/Master.csv

Ian

ianplain :

You need to look at the raw data that is held in /var/log/asterisk/cdr-csv/Master.csv Ian

I can see 2 Different seconds there, Rining and Billed ,but each time i have to open the Master.csv fiel?

Why it is not showing same in call history?

Mani.