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

Sergio Charrua sergio.charrua at voip.pt
Wed Jul 22 15:33:24 CEST 2020


you can set the DB as Master-Master, and add a virtual IP for the DB
Cluster. Point your Asterisk boxes to connect to that VIP, and you're
good to go.


*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 1:22 PM Aristeidis Tsitras <tsitras at gmail.com>
wrote:

> 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>*
>>
>> This message and any files or documents attached are strictly
>> confidential or otherwise legally protected.
>>
>> It is intended only for the individual or entity named. If you are not
>> the named addressee or have received this email in error, please inform the
>> sender immediately, delete it from your system and do not copy or disclose
>> it or its contents or use it for any purpose. Please also note that
>> transmission cannot be guaranteed to be secure or error-free.
>>
>>
>>
>>
>>
>>
>>
>>
>> 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
>>>>>>>
>>>>>> _______________________________________________
>>>>>> 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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