[asterisk-dev] Fwd: Asterisk Sip Registration Hooks

Mike Myhre digium at aeisecure.com
Fri Jul 29 16:50:09 CDT 2011


> In spite of that, I have some good news to report to you: Asterisk 
> 1.8, 10, and trunk already contain a 'security events' framework, with 
> a simple module that can published the generated events to a log file. 
> The AMI code has the ability to generate security events (failed 
> logins and such) in Asterisk 1.8 and later. Russell added this 
> facility quite some time ago, but so far nobody has stepped up to 
> start adding event generation in other modules.

That is great news! That is more than I could have hoped for in a hook 
into asterisk. How risky is it to move from 1.6 to 1.8 though? From the 
digium website, I see that 1.6 is listed as the current stable version. 
The downloads page shows both as being stable. There are many other 
features in 1.8 that I would like to take advantage of, but haven't 
wanted to 'open a can of worms' in 'new bugs I don't understand'.

There is one other issue with asterisk that I have had problems working 
around. It is the CDR being lost when a redirect is done from the AMI. 
Looking at the bug reports, it seems to be acknowledged but not fixed. 
Is there a chance that it is fixed in 1.8 and I just haven't found it 
documented? Are there other options that can give me the same 
information (like CEL) that could handle the more complex situations 
that may occur in AMI call manipulation? A single CDR for each initial 
call is a lot to ask when in reality it may go from being answered by a 
queue, transferred to an operator, sent to a conference bridge and then 
transferred to another 3 way call. I see the ultimate solution to be a 
master-detail call record where there is one initial CDR with multiple 
records (queue.log or CEL?) that explain the details of each subsequent 
leg of the call. With 1.6, if I do an AMI redirect, I don't get any CDR 
and that is a big problem to work around.

Thanks for your help!

     Mike




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