Hi,

Please see the inline comments.

--- Charles Wang lazy.charles@gmail.com wrote:

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?

yes you can use b2bua for this.

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?

ser+rtpproxy is one solution and ser+media proxy is another solution to NAT traversal. So you can use either of the combination. You don't need rtpproxy when you are using mediaproxy.

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?

If you look at the accounting.php file under serweb, you can see how the accounting is done. It actually gives the call duration. It's only a matter of manipulatin the data in the mysql database.

Best Regard Charles

---

Serusers mailing list serusers@lists.iptel.org http://lists.iptel.org/mailman/listinfo/serusers

__________________________________ Do you Yahoo!? Yahoo! Mail - 250MB free storage. Do more. Manage less. http://info.mail.yahoo.com/mail_250