So, a few questions:
- If the call received by asterisk from the PRI is sent to a number
not in the dialplan, what will asterisk do? Will the call be
cancelled, or will asterisk signal something back to the switch to
indicate "dunno about this, try another"?
Asterisk will do whatever you tell it to do. Here is a hint of what
you can do before you reject a number:
http://www.voip-info.org/wiki/index.php?page=Asterisk+variable
+PRI_CAUSE
Sure, but I can't find a cause code that means 'try next link'. do
you know if there is one, and what this is?
- If the call is received by Asterisk from a wildchar extensions
like exten => _X.,1,......., will it be possible to signal the switch
"dunno about this, try another" later in the chain? Since Asterisk's
What do you mean by later in the chaim? Again the above link will sure
help you if you use some GotoIf($[${EXTEN}) logic.
Using several boxes that serve separate functions and separate blocks
of DIDs etc, or just random numbers spread out after porting them,
it'll be good to let asterisk do the filtering instead of relying on
the telco.
current dialplan implementation does not scale wery well, we're doing
sip user/callerid lookups in mysql from an AGI script and the MYSQL
How does it not scale very well for the mentioned problem?
with thousands of customers and separating callerid and sip username,
you'll need some logic that interfaces with a dbms, and not simply
storing it all in extensions.(conf|ael). IIRC the current realtime
extensions support makes some three or four queries per lookup and
isn't really easy to normalise.
but then, I might as well be wrong.
app from -addons. Anyway, this means we'll accept all incoming calls
before handling them (not answering, though, just accepting them to
the dialplan).
Thats how it should be, it takes all of some 2 ms (give or take some,
I might be off) to reject a call in the Dialplan which shuldn't be a
problem.
I just wondered what PRI_CAUSE to use, really.....
roy
--
Roy Sigurd Karlsbakk
[EMAIL PROTECTED]
(+47) 98013356
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