[asterisk-dev] Integration with legacy systems ...

[Mauro Sergio Ferreira Brasil]

Hi Kevin!

Thanks a lot for you patience and sincere efforts to help and clear my 
doubts. I'll try to detail the questions we still have to answer before 
implementing our solution here.
For the sake of simplicity I'll call "middle-man" the application that 
will be the platform information driver and the responsible to provide 
the interface to legacy systems on our solution.

1- First demand we will have here is an "element" (probably a module), 
that will retrieve some configurations from middle-man (timeouts, SIP 
and RTP defaults, and retry values) on startup to be used by the 
Asterisk instance.
As long as I could see, is there no way to do that with AMI, CLI or 
other interface provided by Asterisk. Am I right ?
Maybe we can create a config engine (registrable through 
"ast_config_engine_register") that instead of accessing a database, 
consults middle-man and use this config engine through ARA. Is that 
possible ?

2- Another demand we have is regarding autentication of SIP users, that 
will be performed during SIP REGISTER. Until now, and considering what 
you've already told me, I think this will need to be done by another 
platform node, given that I haven't found a way to make it using any of 
Asterisk API, interfaces, etc. Am I missing something ?
(I would like to not consider the "programmable proxy" as suggested by 
you on this case).

3- We will have to update dialplan and allow SIP devices creation 
dynamically.
In fact we will need a way to operate with middle-man on each call to 
decide if it can be done (based on "autorization" rules, and credits for 
outgoing calls), and in case the user requested an outgoing call to PSTN 
maybe we will need to provide the better outgoing gateway too.
Besides sip users that will be created on legacy systems and that must 
be available on demand.
Again, the idea is to use the config engine pointed on item 1 so we can 
retrive extensions and sip_users configuration from middle-man. Do you 
see any problem with this approach ?

Thanks and best regards,
Mauro.



Kevin P. Fleming escreveu:
> You can't 'intercept' call setup events, because Asterisk is not a
> proxy. You can either just *handle* the events normally by letting them
> arrive at the dialplan, use whatever logic you wish, and create a new
> outbound call leg, *OR* you can use a programmable proxy. Which you
> choose will depend almost entirely on how 'transparent' you want the
> signaling changes to be, with Asterisk in the middle, the outbound
> INVITE is not really going to look very much like the incoming one at all.
>
>   

-- 
__At.,                                                                                                                             
   _
 
*Technology and Quality on Information*
Mauro Sérgio Ferreira Brasil
Coordenador de Projetos e Analista de Sistemas
+ mauro.brasil at tqi.com.br <mailto:@tqi.com.br>
: www.tqi.com.br <http://www.tqi.com.br>
( + 55 (34)3291-1700
( + 55 (34)9971-2572

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