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 <mailto:sip%3A500@192.168.0.99>>;tag=580308996
To: <sip:500@192.168.0.99 <mailto:sip%3A500@192.168.0.99>>;tag=as4657cc95
Call-ID: 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 <mailto:jurijs.ivolga@gmail.com>> έγραψε:
Ah,
I see the problem, just change list file in following way:
1 sip:192.168.0.100:5080 <http://192.168.0.100:5080> 0 0 maxload=20
1 sip:192.168.0.101:5080 <http://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 <mailto: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(a)gmail.com <mailto: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(a)gmail.com <mailto: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.…
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 <mailto: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
<http://192.168.0.100:5080> 0 0 maxload=20
2 sip:192.168.0.101:5080
<http://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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