Hi Paul
As marian mentioned there are a few creative methods to get this to work "kind of".
Theone I think is the best is session_timer, however this puts a load on the server, since re-invites need to be sent (this is in the archive somewhere).
Also if you have different billing rates per day/per hour, and per location and even customer type (like some morons such as moi have decided to do :-)) then session timers become very complicated, since each timer needs to be different.
EG customer has credit of 100p, but calls some remote country which charges 70p per minute, in theory he should only get approx 90 secs of call, hence if session timer is fixed for him and everyone, at say 120 seconds (or even longer), he will overrun. This may not be a big problem, since if you have alook at your blended costs/rev model you may be able to work that into the overall scheme of things.
hence why I am having a few painful weeks :-), also sipsak can be used to send a BYE message to the gateway (although I am still in the process of trying to test this...but I think a few on this list have done it successfully).
Easiest is to use a B2Bua, eg asterisk, or asterisk-b2bua but then you are really handling all the media stream, which then adds to cost in terms of bandwidth server load etc etc, so you will have to work this out based upon what business model you really wish to go for.
Iqbal
On 4/23/2005, "Marian Dumitru" marian.dumitru@voice-sistem.ro wrote:
Hi Paul,
There were several discussions on the list about these topics. Just to summarize:
- to tear down hanged calls to PSTN, the best way to go is by using
Session Timer on the GW (if it supports); if not supported you have to use a B2BUA probably.
- to limit the call duration you need a B2BUA. Another alternative is
to play with dynamic values for Session Timer and force the GW to hang the call for you.
In both cases you relay only on signalling support.
Best regards, Marian
Paul van Schagen wrote:
Hello,
I wonder if anyone has a solution for having a call being terminated by SER in case a certain credit limit has been reached or
in case the connection was lost to either end of a call ? I am testing with mediaproxy and can have it timeout, it will tear down the
RTP relaying nevertheless it wont teminate the actual call by means of a BYE request that was setup to the PSTN gateway.
So this connection stays open.
Is there a way to interface with SER and to have it generate a BYE to the PSTN gateway ?
Can I use RADIUS to tear down the call in case of a prepaid credit limit ?
Best regards
Paul van Schagen
-- Voice System http://www.voice-system.ro
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