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??? -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20070608/b2ce6dff/attachment.htm