I experience something really strange when i change the dispatcher list to
have the same priority both of the servers. I am getting 401 (unauthorised)
to the extensions trying to register. Even though they in asterisk they
seem to be registered, the debug shows 401. Not able to call from one
Ah,
I see the problem, just change list file in following way:
1 sip:192.168.0.100:5080 0 0 maxload=20
1 sip:192.168.0.101:5080 0 0 maxload=20
Jurijs
On Tue, Jul 21, 2020 at 3:28 PM Jurijs Ivolga
wrote:
> Hi Aristedis,
>
> Sorry, indeed you have module parameters.
>
> When one asterisk is
Hi Aristedis,
Sorry, indeed you have module parameters.
When one asterisk is down what you see when you run:
kamcmd dispatcher.list
Jurijs
On Tue, Jul 21, 2020 at 3:19 PM Aristeidis Tsitras
wrote:
> i know that there is something wrong, but i can not figure it out.
> Especially for the
i know that there is something wrong, but i can not figure it out.
Especially for the modparam settings that Mr Jurijs Ivolga is proposing, I
already had them. it is the kamailio.cfg that I originally attached.
Unfortunately I did not manage to find anything in the parameters that will
solve the
Hi Aristeidis,
In your case Dispatcher module is misconfigured and it missing crutials
parts like:
modparam("dispatcher", "flags", 2) # without this flag no failover will
happen, as you experiencing
modparam("dispatcher", "xavp_dst", "_dsdst_") # this xavps will hold the
list with addresses
First thing is trying to get both servers status on opensips and make sure
opensips sees them up:
https://opensips.org/html/docs/modules/2.3.x/dispatcher.html#idp5739696
On Tue, 21 Jul 2020 at 12:12, Aristeidis Tsitras wrote:
> I have a Kamailio and 2 asterisk servers. All users are created
I have a Kamailio and 2 asterisk servers. All users are created in both of
the asterisk servers. I am forwarding the registration to asterisk. The
problem is that it is always used on only one server from the list. Even if
one goes to shutdown, then there is not any registration sent to the
Siremis is just a control panel, mostly for the DB data used by some Kamailio
modules. So, your question is really a Kamailio question; you can’t make
routing logic changes from Siremis.
As far as your actual question, how are you resolving the service domain to
“the actual PBX”? What’s being
I have recently installed Siremis
And have set up a domain pointing to it, how do I configure it so registration
requests to this domain are passed thru to the actual PBX
Adam
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Hi,
To clarify, we are talking about the PAI header from Caller..
On Tue, Jul 21, 2020 at 12:23 PM BALL SUN wrote:
>
> Hi
>
> We have a question. We are preparing a setup to bridge the SIP call
> from Kamailio IMS to PSTN, but we are puzzling how we can present the
> P-Asserted-Identity in the
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