[SR-Users] Dispatcher does not go to 2nd server

Wed Jul 22 14:15:26 CEST 2020

So, in order to have it working i need 2 realtime asterisk that are in a
master-slave setup and have users created in both. Am i correct?

Στις Τετ, 22 Ιουλ 2020, 11:36 ο χρήστης Sergio Charrua <
sergio.charrua at voip.pt> έγραψε:

> Try to install SNGREP command line application. It greatly helped me to
> understand what was going on in kamailio.
> Install it in your Kamailio server, and run it. You will then be able to
> see and understand all SIP requests and responses.
>
> Also, as Daniel said, the SIP requests between client->kamailio->asterisk
> must be kept the same at all times when trying to register. Additionally,
> install Asterisk with Realtime module, and share the same SIP users across
> your asterisk boxes, by the means of a DB. This is the best approach.
>
> Hope this helps,
>
> *Sérgio Charrua*
>
>
> *www.voip.pt <http://www.voip.pt/>*
> Tel.: +351  <callto:+351+91+104+12+66>21 130 71 77
>
> Email : *sergio.charrua at voip.pt <sergio.charrua at voip.pt>*
>
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>
> On Wed, Jul 22, 2020 at 6:19 AM Aristeidis Tsitras <tsitras at gmail.com>
> 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 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 at 192.168.0.99 <sip%3A500 at 192.168.0.99>>;tag=580308996To:
>> <sip:500 at 192.168.0.99 <sip%3A500 at 192.168.0.99>>;tag=as4657cc95Call-ID:
>> 822746835-5060-1 at 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 at 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 at 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 at gmail.com>
>>>> wrote:
>>>>
>>>>> 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 at 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.p.flags
>>>>>> 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 at 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.
>>>>>>>
>>>>>>>
>>>>>>>
>>>>>>> _______________________________________________
>>>>>>> Kamailio (SER) - Users Mailing List
>>>>>>> sr-users at lists.kamailio.org
>>>>>>> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
>>>>>>>
>>>>>>
>>>>>>
>>>>>> --
>>>>>> Arsen Semenov
>>>>>>
>>>>>> _______________________________________________
>>>>>> Kamailio (SER) - Users Mailing List
>>>>>> sr-users at lists.kamailio.org
>>>>>> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
>>>>>>
>>>>> _______________________________________________
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>>>>>
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