VoIP Mailing List Archives

[noah at allresearch.com]

Hi,

As some of you are probably aware, I've had a really hard time getting
asterisk to work with FS.

The effective_caller_id_number and the accountcode were not getting
logged or passed through on outgoing calls.

I finally solved the problem, but attribute it to some unexpected
behavior of the acl setting. I'm curious as to whether this is the
intended behavior, or a bug.

In my sip profile, I had apply-inbound-acl set to a list that
contained the IP of my asterisk box. Apparently, when this happens,
FS was allowing ANY call from my asterisk box without registration.
Subsequently, since FS was just blindly accepting the call with no
registration, it didn't match anything in the directory and no
accountcode or caller-id was set.

By simply removing the inbound-acl setting, I was able to have
everything working perfectly.

My understanding was that the goal of the inbound-acl was to limit
which IP FS would accept SIP requests from. I was not aware that it
would then accept any call blindly without any associated user.
Interesting....

If any of the FS developers are interested in the exact configuration
file that I'm using, please contact me directly and I'll forward the
file to you.

-Noah

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[krice at suspicious.org]

The ACL is a way to specify a group of trusted machines and the system will
bypass auth on those calls...

If you need something from the directory don't use the ACL...

If you don't want FS to respond to SIP from unknown IP Addresses that's a
more appropriate job for your firewall software (iptables?)

[noah at allresearch.com]

That makes sense.

However it might make sense for me to add something to the wiki about
this. It isn't documented anywhere that an "allow" in the acl will
bypass the directory and registration.

On a separate topic, I was just reading a post of yours from February
where you describe your LCR platform. Would you be willing to share
some of your setup and/or software for this?

Thanks,

-N


On Sep 30, 2008, at 12:47 AM, Ken Rice wrote:

[krice at suspicious.org]

The software in question is a custom dialplan replacement module for
freeswitch that is available for licensing...

As far as the setup goes you set up the sip profiles as normal, ratedecks
and other items are loaded into a DB and routing happens automagically...