22 Jul
2020
22 Jul
'20
3:15 p.m.
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(a)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(a)voip.pt <sergio.charrua(a)voip.pt>*
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On Wed, Jul 22, 2020 at 6:19 AM Aristeidis Tsitras <tsitras(a)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@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:
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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:
> 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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>>
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