<div dir="ltr">You need this:<div> <param name="apply-inbound-acl" value="domains"/><br></div><div><br></div></div><div class="gmail_extra"><br><br><div class="gmail_quote">On 5 May 2014 17:13, Victor Chukalovskiy <span dir="ltr"><<a href="mailto:victor.chukalovskiy@gmail.com" target="_blank">victor.chukalovskiy@gmail.com</a>></span> wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">Hello,<br>
<br>
Coming from wholesale background, my FS's run without any registrations.<br>
So far everything was ACL-based using "apply-inbound-acl" and I did not<br>
use any directory entries.<br>
<br>
The only problem with this is that once I have all IPs together in one<br>
big ALC, I can't identify which customer the call came from. E.g. need<br>
to set my_channel_variable=customer1 if a call came from particular IPs<br>
and my_channel_variable=customer2 if a call came from other IPs.<br>
<br>
So I'm trying to move ACL logic into directory by means of defining a<br>
user with cidr attribute. So far, no matter what I do FS challenges<br>
INVITE with "407" even-though the INVITE comes from the IP that is<br>
included in CIDR attribute for a user. I suppose for whatever reason<br>
switch does not match INVITEs against CIDR's in the directory. Please<br>
help me with that. WiKi is written from a somewhat different logic /<br>
perspective, so it's hard to apply.<br>
<br>
My SIP profile is:<br>
<br>
<profile name="test"><br>
<gateways><br>
</gateways><br>
<domains><br>
</domains><br>
<settings><br>
<param name="parse-invite-tel-params" value="true"/><br>
<param name="user-agent-string" value="test"/><br>
<param name="debug" value="0"/><br>
<param name="sip-trace" value="no"/><br>
<param name="log-auth-failures" value="true"/><br>
<param name="rfc2833-pt" value="101"/><br>
<param name="sip-port" value="5060"/><br>
<param name="dialplan" value="XML"/><br>
<param name="context" value="test"/><br>
<param name="country" value="e164"/><br>
<param name="dtmf-duration" value="2000"/><br>
<param name="inbound-codec-prefs" value="$${default_codec_prefs}"/><br>
<param name="outbound-codec-prefs" value="$${default_codec_prefs}"/><br>
<param name="caller-id-type" value="none"/><br>
<param name="rtp-timer-name" value="soft"/><br>
<param name="rtp-ip" value="192.168.1.2"/><br>
<param name="sip-ip" value="192.168.1.2"/><br>
<param name="manage-presence" value="false"/><br>
<param name="manage-shared-appearance" value="false"/><br>
<param name="inbound-codec-negotiation" value="greedy"/><br>
<param name="disable-transcoding" value="true"/><br>
<param name="manual-redirect" value="false"/><br>
<param name="disable-transfer" value="true"/><br>
<param name="disable-register" value="false"/><br>
<param name="auth-calls" value="true"/><br>
<param name="rtp-timeout-sec" value="300"/><br>
<param name="rtp-hold-timeout-sec" value="1800"/><br>
<param name="pass-callee-id" value="false"/><br>
</settings><br>
</profile><br>
<br>
<br>
Thanks!<br>
-Victor<br>
<br>
<br>
<br>
<br>
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</blockquote></div><br></div>