Dear ALL: I want to implement a prepaid (for PSTN) ser sip solution. But there are serveral questions in my mind: 1. Is it possible to force to tear-down the call properly? Can I use asterisk or b2bua only if I don't want to do it on trunk? 2. In mediaproxy+rtpproxy mode, I know there is a sub-project under "ser/sip_router" named "rtpproxy". Does it must be start when I want to implement mediaproxy+rtpproxy? Or does the mediaproxy Python server have its own rtpproxy? 3. Do I must modify the acc module or a new module to write the cdr and billing for this goal? Because acc module only writes INVITE and BYE to acc table, and it will not calculate the call during time. Is there an another method to implement it? Best Regard Charles