22 Feb
2005
22 Feb
'05
5:28 a.m.
Hi,
Please see the inline comments.
--- Charles Wang <lazy.charles(a)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
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