<div dir="ltr">Hi Paul<div><br></div><div>I think we have our nat stuff working on the kamailio server. The problem now for me seems to be that I have a restrictive nat going on my home network which is preventing me from receiving calls.</div><div><br></div><div>I am going to do some research on changing up my network here for testing.</div><div><br></div><div>Thanks again for all the great help.</div><div><br></div><div>I hope you have a wonderful weekend.</div><div><br></div><div>Will</div></div><div class="gmail_extra"><br><div class="gmail_quote">On Fri, Oct 3, 2014 at 12:31 AM, Paul Smith <span dir="ltr"><<a href="mailto:paul.smith@claritytele.com" target="_blank">paul.smith@claritytele.com</a>></span> wrote:<br><blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<div bgcolor="#FFFFFF" text="#000000">
Great stuff.<br>
<br>
NAT is a whole other basket of pain. Again the example configs in
the kamailio distribution are a good place to start... in particular
the NATDETECT and NATMANAGE routines, and the nathelper and rtpproxy
module usage.<br>
<br>
good luck<div><div class="h5"><br>
<br>
<blockquote type="cite">
<div dir="ltr">Hi Paul
<div><br>
</div>
<div>Just wanted to give you an update.</div>
<div><br>
</div>
<div>This looks like it has worked. Now I am dealing with my own
natting issues on my home network to get the call but the
invites are being sent right now.</div>
<div><br>
</div>
<div>Thanks again for the assistance.</div>
<div><br>
All the best.</div>
<div><br>
</div>
<div>Will Ferrer</div>
</div>
<div class="gmail_extra"><br>
<div class="gmail_quote">On Wed, Oct 1, 2014 at 11:53 PM, Paul
Smith <span dir="ltr"><<a href="mailto:paul.smith@claritytele.com" target="_blank">paul.smith@claritytele.com</a>></span>
wrote:<br>
<blockquote class="gmail_quote" style="margin:0 0 0 .8ex;border-left:1px #ccc solid;padding-left:1ex">
<div bgcolor="#FFFFFF" text="#000000">
<div>Hi Will,<br>
It sounds like your kamailio.cfg is not looking up the
user location database before trying to relay the
INVITE. There is a relevant section in the
kamailio-basic.cfg example configuration file:<br>
<br>
<br>
<blockquote>
<pre>request_route {
...
# user location service
route(LOCATION);
}
...
# USER location service
route[LOCATION] {
if (!lookup("location")) {
$var(rc) = $rc;
t_newtran();
switch ($var(rc)) {
case -1:
case -3:
send_reply("404", "Not Found");
exit;
case -2:
send_reply("405", "Method Not Allowed");
exit;
}
}
# when routing via usrloc, log the missed calls also
if (is_method("INVITE")) {
setflag(FLT_ACCMISSED);
}
route(RELAY);
exit;
}
</pre>
</blockquote>
The logic is that if the call is for a local registered
user whose location is available in the "kamctl ul"
then request_route() should pass the request to the
route(LOCATION) routine. The function call
lookup("location") then does the magic if matching the
address of record ([subscriber_name]@[our_domain_name])
and returning the $ruri of the registered phone
([realid]@[realip]). route(RELAY) is then able to send
the call on to the phone's actual IP address.<br>
<br>
Hope that helps.<br>
<br>
Paul Smith
<div>
<div><br>
On 02/10/14 03:33, Will Ferrer wrote:<br>
</div>
</div>
</div>
<blockquote type="cite">
<div>
<div>
<div dir="ltr">Hi
<div><br>
</div>
<div>I was wondering if any one had any advice or
examples for me of how to get a call to be
routed to a subscribed softphone.</div>
<div><br>
</div>
<div>We have 2 boxes in our testing deployment, a
load balancer / sbc and a call processing box.</div>
<div><br>
</div>
<div>Calls come in to the sbc, and then are passed
to the call processing box. The call is analyzed
and the branch uri is rewritten to a destination
address when applicable for the call (this is
how we handle routing of calls to certain
numbers -- we do this utilizing custom code and
a custom db).</div>
<div><br>
</div>
<div>This works just fine when the destination sip
uri is phone number (in which case we do lcr) or
if the destination goes to a remote address.</div>
<div><br>
</div>
<div>However when the destination is a subscriber:
sip:[subscriber_name]@[our_domain_name], the
call is passed back to the sbc, which passes it
to the callprocessing box, back and forth until
a too many hops error occurs.</div>
<div><br>
</div>
<div>The subscriber I am trying to send the call
too does show up under "kamctl ul show".</div>
<div><br>
</div>
<div>I feel like there is something basic I must
be missing here.</div>
<div><br>
</div>
<div>Does any one have any advice for me?</div>
<div><br>
</div>
<div>Thank you very much in advance.</div>
<div><br>
</div>
<div>All the best.</div>
<div><br>
</div>
<div>Will Ferrer</div>
<div><br>
</div>
</div>
<br>
<fieldset></fieldset>
<br>
</div>
</div>
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