Hello there!
I'm working on some modifications on Asterisk to adapt it to our needs
considering some particular demandings of the infraestructure we want to
provide.
Two of these modifications are:
1- A proprietary configuration driver that will communicate with a
server that will be the source of information for the entire
infraestructure; and,
2- A call control application that will be responsible for call timing
control and pre-paid support;
Here we are prioritizing internal modifications and loadable modules
(like modules, applications, etc) against external AGI components to
acchieve the best performance possible for the entire solution.
One problem we have here is to find out the best option (even one that
results on some internal Asterisk files changing) that allow us to
propagate the SIP header "Call-ID" to both modules described above.
The best shot we have until now is to use the "callid" field from the
"sip_pvt" structure of SIP channel, what will lead us to two
considerable code changes: 1- Propagate the channel to method
"realtime_var_get" of our proprietary ARA driver; and 2- Duplication of
necessary structs to a header (".h") file so the modules can "navigate"
on private structure "sip_pvt".
The first change isn't big deal. But the need of validation of the
second modification, every time we make a merge with updated codes is
concerning me a lot.
Does anyone have a better approach to get this done ?
Thanks and best regards,
--
__At.,
_
*Technology and Quality on Information*
Mauro Sérgio Ferreira Brasil
Coordenador de Projetos e Analista de Sistemas
+ [email protected] <mailto:@tqi.com.br>
: www.tqi.com.br <http://www.tqi.com.br>
( + 55 (34)3291-1700
( + 55 (34)9971-2572
_______________________________________________
-- Bandwidth and Colocation Provided by http://www.api-digital.com --
AstriCon 2009 - October 13 - 15 Phoenix, Arizona
Register Now: http://www.astricon.net
asterisk-users mailing list
To UNSUBSCRIBE or update options visit:
http://lists.digium.com/mailman/listinfo/asterisk-users