Hi Paul
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.
I am going to do some research on changing up my network here for testing.
Thanks again for
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.
Thi
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 S
Hi Paul
Thank you very much for the quick response and the probably spot on
information. I really bet that is my issue.
I will be testing this out tonight and I will post back with what I find.
Thanks again and all the best
Will Ferrer
> On Oct 1, 2014, at 11:53 PM, Paul Smith wrote:
>
> H
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(LO
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 c
A little progress on this; the double-AOR problem was fixed; I had
callbackextension field in the realtime table, which caused extra REGISTER
messages to be sent from Asterisk to Kamailio and that messed the AORs for
my clients.
Calling between the clients is a problem though, the INVITE gets rout
Hello,
I've started playing with an idea to add multiple asterisk servers and
using dispatcher to balance the sip load between them. I added the code
according to dispatcher module documentation (
http://www.kamailio.org/docs/modules/4.2.x/modules/dispatcher.html), but I
think there's something of
Hi,
I'm looking for pointers on the following scenario.
I have two sites (A and B) and sip traffic is sent from A to B. At each site
there are a number of SIP (Asterisk) servers and a Kamailio server.
The two sites have various network routes (X, Y & Z) between them. I have been
asked to loa
