22 Jul
2020
22 Jul
'20
8:16 a.m.
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
extension to the other.
here is the console error:
*<--- Transmitting (NAT) to 192.168.0.99:5060 <http://192.168.0.99:5060>
--->SIP/2.0 401 UnauthorizedVia: SIP/2.0/UDP
192.168.0.99;branch=z9hG4bKe78c.2793112e87a37f021cda7a582ba8c703.0;received=192.168.0.99;rport=5060Via:
SIP/2.0/UDP
192.168.0.117:5060;received=192.168.0.117;branch=z9hG4bK823694525;rport=5060From:
<sip:500@192.168.0.99 <sip%3A500(a)192.168.0.99>>;tag=580308996To0308996To:
<sip:500@192.168.0.99 <sip%3A500(a)192.168.0.99>>;tag=as4657cc95Call-ID95Call-ID:
822746835-5060-1(a)BJC.BGI.B.BBHCSeq: 2005 REGISTERServer: Asterisk PBX
11.25.3Allow: INVITE, ACK, CANCEL, OPTIONS, BYE, REFER, SUBSCRIBE, NOTIFY,
INFO, PUBLISH, MESSAGESupported: replaces, timerWWW-Authenticate: Digest
algorithm=MD5, realm="asterisk", nonce="73fd892f",
stale=trueContent-Length: 0*
On the other hand if i change the priority to 0 & 1 the servers
respectively, then no errors in asterisk console and they can call between
each other.
This is really odd.
Στις Τρί, 21 Ιουλ 2020 στις 3:32 μ.μ., ο/η Jurijs Ivolga <
jurijs.ivolga(a)gmail.com> έγραψε:
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 <jurijs.ivolga(a)gmail.com>
wrote:
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 <tsitras(a)gmail.com>
wrote:
Kamailio (SER) - Users
Mailing List
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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 problem as proposed by Villasmil and Semenov. I have given a
try on changes but nothing good came up.
Στις Τρί, 21 Ιουλ 2020 στις 2:48 μ.μ., ο/η Arsen Semenov <
arsperger(a)gmail.com> έγραψε:
_______________________________________________ Hi Aristeidis,
David is right, first would be good to check the status of the
destinations.
In your configuration there are couple of things to have in mind: need
to set correct flag param
https://kamailio.org/docs/modules/5.5.x/modules/dispatcher.html#dispatcher.…
select the algorithm used to select the destination (ds_select_dst) and
have a faulure_route where the next destination will be tried in case first
is down.
You can find the examples in the module doc.
Cheers,
On Tue, Jul 21, 2020 at 4:11 PM Aristeidis Tsitras <tsitras(a)gmail.com>
wrote:
> 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 available server. Even if *some *of the extensions can be seen
> registered in both of the asterisk's, if the secondary goes down, then
> there are no services for the phones.
>
> I am attaching the kamailio.cfg. My dispatch list is:
> 1 sip:192.168.0.100:5080 0 0 maxload=20
> 2 sip:192.168.0.101:5080 0 0 maxload=20
>
> In both of the asterisk servers i am using sip.conf to create users
> and s sip trunk for the Kamailio. Nothing special about it.
>
> I am looking to find what is wrong with my config and i cannot
> loadbalance/failover to the asterisk servers.
>
>
>
> _______________________________________________
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>
--
Arsen Semenov
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