[asterisk-users] CDR accuracy

[clive.chan(Alpha Trilogies Networks)]

Hi all users,

I has been joining this user list for about 1 year, and always has seen the
successful story about the Asterisk act as IP PBX and even communication
appliances solutions. And thank for this list to help each other and make
everyone success. I also being inspired by this user-list and wish to start
my implementation of Asterisk as IP PBX. 

However, billing is one of the main concern in the real life production
server, I has been trying with my testing server and it show like the
non-pro path of Asterisk csv file. 

 

For example;

(Polycom phone) and using polycom build in function blind Transfer function.

Exten SIP 200 call outsider (Mr.X)  through Zap Channel, Talk .and then SIP
200 transfer the call (Mr.X) to SIP 300. The CSV billing  shows, 

 

SIP 200 call Mr X and started and end as below;

""WSang"
<200>","200","90124086376","200","SIP/WSang-08b52148","Zap/1-1","Dial",zap/1
/0124086376||WTt","2007-06-09 10:32:52","2007-06-09 10:32:56","2007-06-09
10:33:18","26","22","ANSWERED",""

 

Mr.X has spoken to SIP 300 for about 12sec

"90124086376","90124086376","300","300","Zap/1-1","SIP/chan-08b57688","Hangu
p",0.5","2007-06-09 10:33:33","","2007-06-09 10:33:33","0","0","NO
ANSWER",""

 

Now, when come to billing, first I can bill SIP 200 for the period of
conversation. However, how can I bill SIP 300 for the period of 12sec
conversation? And where to prove that this call is being transfer by SIP 200
to SIP300. 

 

Since, we have so many experiences expert around the list, can some one help
on this issues????? Or do you all have such issues after implemented to your
customer or your own use???

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