hould be over http(s).
>
> Should you be still stuck on something related to this topic, let us know if
> you already used a config like the one linked above, to follow up more on
> this.
>
> Cheers,
> Daniel
>
> On 23/03/2017 14:04, Paul Smith wrote:
>> Hi
>
Hi
I am struggling to figure out how to build and test an integrated presence
server with Resource-List (rls.so) and XCAP (xcap_server.so).
I am confused about what is meant to happen and how to debug it. Not sure if I
am suffering from incompatibilities, configuration errors, or bugs…
I have
Longer version of this got blocked by the mail list earlier today… and I have
dug a bit deeper since then…
I am trying to SUBSCRIBE to a resource-list (rls.so) but the
rls_handle_subscribe() is returning with a “list not found"
Mar 23 16:36:11 presence kamailio[12536]: DEBUG: rls [subscribe.c:2
te a new branch. It may work to
> just use append_branch(), otherwise reassigning $ru to itself should do it.
>
> Cheers,
> Daniel
>
> On 14/09/16 16:11, Paul Smith wrote:
>> Hi,
>> I am struggling to figure out variable scope in branches and
>> branch_failure_route
Hi,
I am struggling to figure out variable scope in branches and
branch_failure_routes. Is there a way to store a variable in a branch route so
that I can then read it from the branch failure route if that branch fails?
Also does writing $ru in branch_failure_route append a branch?
For exampl
Hi Jon,
The normal way would be to use multiple modparam lines, one for each of your
databases. Then in your route config you can select and use any of those
connections:
http://kamailio.org/docs/modules/4.4.x/modules/sqlops.html#idm20488
modparam("sqlops","sqlcon","cb=>mysql://kamailio:abc@10
Hi Zaka
You have misunderstood this line : if (!t_relay()) {
That line explicitly calls t_relay() and the packet is relayed at that time.
If the packet cannot be relayed then the t_relay() function returns an error
and the conditional block is executed.
Paul Smith
On 13 Nov 2014, at 09:10
place to maintain. The
t_relay() function is a good example of a block of functionality that is
reached from loads of places … so generally a route[relay] block is a good
idea.
Paul Smith
On 13 Nov 2014, at 07:52, Zaka Ul Isam wrote:
> Hello Friends and Gurus:
>
> I am struggling wit
Hi Daniel,
I’ve just done a few quick tests after “git pull origin” upgrade.
It works when the 2 snom endpoints are registered over UDP transport with stun
enabled. Which is great thank you very much.
But I have come across a couple of cases that are not working for me yet:
FAIL CASE 1 : It
> Kindly clarify.
>
> Thanks
> Kamal
>
> On Mon, Oct 27, 2014 at 11:29 PM, Paul Smith
> wrote:
> Hi Kamal
>
> dispatcher module needs to be loaded in order to call ds_is_from_list(), or
> ds_select_dst().
>
> It looks like you have added the line
Hi Kamal
dispatcher module needs to be loaded in order to call ds_is_from_list(), or
ds_select_dst().
It looks like you have added the line 'loadmodule “dispatcher.so”’inside the
conditional WITH_ANTIFLOOD block, so it never gets called after you have
disabled that block.
In short make sure t
I added a log line to the top of kamailio.cfg request_route block to grab the
message buffer of the REFER. I also put a condition around the sanity_check to
skip it for method=REFER …
I got the following output for $mb at the start of request_route for the REFER
packet (I have substituted MYP
Thank you for the reply Daniel. I have enabled debug=3 and put in a few more
xlog lines. I can see the REFER coming in on local interface 127.0.0.1. I am
now trying to narrow down the issue in the kamailio.cfg.
My conclusions so far are:
1) The REFER packet has a problem which causes it to fa
I seem to be going round in circles… I am trying to use dlg_bridge() from the
dialog module to establish a call between two SIP endpoints. I have tested
with Snom phones and linphone soft phone with the same result.
I get an outbound call to the first (from) end point, I answer the phone… and
le is used only in kamalio 4.2.x -- development
version. where to get its rpm packet?
B.R.
andrew
At 2014-10-13 16:38:47, "Paul Smith" wrote:
Hi Andrew
There is a condition in the NATMANAGE route which tests whether or
not to apply rtpproxy_manage() :
if (
Thanks for the feedback.
On 14/10/14 02:13, Alberto Llamas wrote:
Hello Paul,
That was amazing. Thanks a lot !
On Mon, Oct 13, 2014 at 3:42 AM, Paul Smith
mailto:paul.sm...@claritytele.com>> wrote:
Hi Alberto
The magic you are looking for is in the "domain" module.
Hi Andrew
There is a condition in the NATMANAGE route which tests whether or not
to apply rtpproxy_manage() :
if (!(isflagset(FLT_NATS) || isbflagset(FLB_NATB)))
return;
One way to force this is to make sure that FLT_NATS is always set. That
can be done in the NATDETECT ro
Hi Alberto
The magic you are looking for is in the "domain" module. The example
configuration files in the distribution have it set up and ready to use
if you add the line "#!define WITH_MULTIDOMAIN"
The important bit for your scenario is in the example config:
# - domain params -
.
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
mailto:paul.sm...@claritytele.com
me]@[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 exam
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