3 Oct
2014
3 Oct
'14
10:31 a.m.
Great stuff.
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.
good luck
Hi Paul
Just wanted to give you an update.
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.
Thanks again for the assistance.
All the best.
Will Ferrer
On Wed, Oct 1, 2014 at 11:53 PM, Paul Smith
<paul.smith(a)claritytele.com <mailto:paul.smith@claritytele.com>> wrote:
Hi Will,
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:
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;
}
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.
Hope that helps.
Paul Smith
On 02/10/14 03:33, Will Ferrer wrote:
Hi
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.
We have 2 boxes in our testing deployment, a load balancer / sbc
and a call processing box.
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).
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.
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.
The subscriber I am trying to send the call too does show up
under "kamctl ul show".
I feel like there is something basic I must be missing here.
Does any one have any advice for me?
Thank you very much in advance.
All the best.
Will Ferrer
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