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Hi Kevin!<br>
<br>
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.<br>
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.<br>
<br>
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.<br>
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 ?<br>
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 ?<br>
<br>
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 ?<br>
(I would like to not consider the "programmable proxy" as suggested by
you on this case).<br>
<br>
3- We will have to update dialplan and allow SIP devices creation
dynamically.<br>
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.<br>
Besides sip users that will be created on legacy systems and that must
be available on demand.<br>
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 ?<br>
<br>
Thanks and best regards,<br>
Mauro.<br>
<br>
<br>
<br>
Kevin P. Fleming escreveu:
<blockquote cite="mid:4A7700C6.1020009@digium.com" type="cite">
<pre wrap="">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.
</pre>
</blockquote>
<br>
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