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@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@192.168.0.99>;tag=580308996 To: <sip:500@192.168.0.99 sip%3A500@192.168.0.99>;tag=as4657cc95 Call-ID: 822746835-5060-1@BJC.BGI.B.BBH 822746835-5060-1@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@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@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@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@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... 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@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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