22 Jul
2020
22 Jul
'20
8:37 a.m.
Hi Aristedis,
To use both servers simultaneously you need to have group id as 1 for both
of them, because ds_select_dst("1", "4") will choose only servers
from
group what is in the first argument(in this case 1) and group 2 will never
be used in your case.
To make this work probably you need to look into Asterisk Realtime, in that
case Asterisk will have to have a shared DB with shared registrations. Some
time ago realtime asterisk didn't support the path, but it was in version
13 as I remember, so I hope it is fixed now.
Jurijs
On Wed, Jul 22, 2020 at 9:24 AM Daniel-Constantin Mierla <miconda(a)gmail.com>
wrote:
Hello,
if you send first register to one asterisk and it replies 401, then the
following up register has to be sent to same asterisk, otherwise the
authentication fails if you send to 2nd asterisk and it doesn't recognize
the nonce, no matter the user password is the same.
Maybe asterisk has options to "share" (generate and recognize same) nonce
values across different instance, if yes, be sure it is enabled. In
kamailio that is possible by setting the appropriate parameters in auth
module.
Cheers,
Daniel
On 22.07.20 07:16, Aristeidis Tsitras wrote:
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 Unauthorized Via: SIP/2.0/UDP
192.168.0.99;branch=z9hG4bKe78c.2793112e87a37f021cda7a582ba8c703.0;received=192.168.0.99;rport=5060
Via: SIP/2.0/UDP
192.168.0.117:5060;received=192.168.0.117;branch=z9hG4bK823694525;rport=5060
From: <sip:500@192.168.0.99 <sip%3A500(a)192.168.0.99>>;tag=580308996 To:
<sip:500@192.168.0.99 <sip%3A500(a)192.168.0.99>>;tag=as4657cc95 Call-ID:
822746835-5060-1(a)BJC.BGI.B.BBH <822746835-5060-1(a)BJC.BGI.B.BBH> CSeq: 2005
REGISTER Server: Asterisk PBX 11.25.3 Allow: INVITE, ACK, CANCEL, OPTIONS,
BYE, REFER, SUBSCRIBE, NOTIFY, INFO, PUBLISH, MESSAGE Supported: replaces,
timer WWW-Authenticate: Digest algorithm=MD5, realm="asterisk",
nonce="73fd892f", stale=true Content-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:
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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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