21 Jul
2020
21 Jul
'20
9:09 p.m.
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.
21 Jul
21 Jul
9:18 p.m.
First thing is trying to get both servers status on opensips and make sure
opensips sees them up:
https://opensips.org/html/docs/modules/2.3.x/dispatcher.html#idp5739696
On Tue, 21 Jul 2020 at 12:12, 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.
_______________________________________________
Kamailio (SER) - Users Mailing List
sr-users(a)lists.kamailio.org
https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
--
Regards,
David Villasmil
email: david.villasmil.work(a)gmail.com
phone: +34669448337
9:42 p.m.
Hi Aristeidis,
In your case Dispatcher module is misconfigured and it missing crutials
parts like:
modparam("dispatcher", "flags", 2) # without this flag no failover
will
happen, as you experiencing
modparam("dispatcher", "xavp_dst", "_dsdst_") # this xavps
will hold the
list with addresses and associated properties, without it no failover will
happen
modparam("dispatcher", "xavp_ctx", "_dsctx_") # The name of
the XAVP which
will hold some attributes specific to dispatcher routing context. I'm not
100% sure, but I think it will not work without this xavp too, in all my
configurations it is there.
Jurijs
On Tue, Jul 21, 2020 at 2:22 PM David Villasmil <
david.villasmil.work(a)gmail.com> wrote:
First thing is trying to get both servers status on
opensips and make sure
opensips sees them up:
https://opensips.org/html/docs/modules/2.3.x/dispatcher.html#idp5739696
On Tue, 21 Jul 2020 at 12:12, 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.
_______________________________________________
Kamailio (SER) - Users Mailing List
sr-users(a)lists.kamailio.org
https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
--
Regards,
David Villasmil
email: david.villasmil.work(a)gmail.com
phone: +34669448337
_______________________________________________
Kamailio (SER) - Users Mailing List
sr-users(a)lists.kamailio.org
https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
9:43 p.m.
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.
_______________________________________________
Kamailio (SER) - Users Mailing List
sr-users(a)lists.kamailio.org
https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
--
Arsen Semenov
10:17 p.m.
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.
_______________________________________________
Kamailio (SER) - Users Mailing List
sr-users(a)lists.kamailio.org
https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
--
Arsen Semenov
_______________________________________________
Kamailio (SER) - Users Mailing List
sr-users(a)lists.kamailio.org
https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
10:28 p.m.
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:
_______________________________________________
Kamailio (SER) - Users Mailing List
sr-users(a)lists.kamailio.org
https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
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(a)lists.kamailio.org
https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
--
Arsen Semenov
_______________________________________________
Kamailio (SER) - Users Mailing List
sr-users(a)lists.kamailio.org
https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
10:30 p.m.
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:
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:
_______________________________________________
Kamailio (SER) - Users Mailing List
sr-users(a)lists.kamailio.org
https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
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(a)lists.kamailio.org
https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
--
Arsen Semenov
_______________________________________________
Kamailio (SER) - Users Mailing List
sr-users(a)lists.kamailio.org
https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
22 Jul
22 Jul
3:16 p.m.
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:
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:
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:
Kamailio (SER) - Users
Mailing List
sr-users(a)lists.kamailio.org
https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
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.
>
>
>
> _______________________________________________
> Kamailio (SER) - Users Mailing List
> sr-users(a)lists.kamailio.org
> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
>
--
Arsen Semenov
_______________________________________________
Kamailio (SER) - Users Mailing List
sr-users(a)lists.kamailio.org
https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
_______________________________________________
Kamailio (SER) - Users Mailing List
sr-users(a)lists.kamailio.org
https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
4:21 p.m.
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.
_______________________________________________
Kamailio (SER) - Users Mailing List
sr-users(a)lists.kamailio.org
<mailto:sr-users@lists.kamailio.org>
https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
--
Arsen Semenov
_______________________________________________
Kamailio (SER) - Users Mailing List
sr-users(a)lists.kamailio.org
<mailto:sr-users@lists.kamailio.org>
https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
_______________________________________________
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sr-users(a)lists.kamailio.org
<mailto:sr-users@lists.kamailio.org>
https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
_______________________________________________
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Daniel-Constantin Mierla -- www.asipto.com
www.twitter.com/miconda -- www.linkedin.com/in/miconda
Funding: https://www.paypal.me/dcmierla
4:37 p.m.
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(a)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(a)192.168.0.99>>;tag=580308996 To:
<sip:500@192.168.0.99 <sip%3A500(a)192.168.0.99>>;tag=as4657cc95 Call-ID:
822746835-5060-1(a)BJC.BGI.B.BBH <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> 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(a)gmail.com>
wrote:
_______________________________________________
Kamailio (SER) - Users Mailing
Listsr-users@lists.kamailio.orghttps://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
--
Daniel-Constantin Mierla -- www.asipto.comwww.twitter.com/miconda --
www.linkedin.com/in/miconda
Funding: https://www.paypal.me/dcmierla
_______________________________________________
Kamailio (SER) - Users Mailing List
sr-users(a)lists.kamailio.org
https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
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:
Kamailio (SER) - Users
Mailing List
sr-users(a)lists.kamailio.org
https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
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.
>>
>>
>>
>> _______________________________________________
>> Kamailio (SER) - Users Mailing List
>> sr-users(a)lists.kamailio.org
>> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
>>
>
>
> --
> Arsen Semenov
>
> _______________________________________________
> Kamailio (SER) - Users Mailing List
> sr-users(a)lists.kamailio.org
> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
>
_______________________________________________
Kamailio (SER) - Users Mailing List
sr-users(a)lists.kamailio.org
https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
_______________________________________________ 
6:32 p.m.
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>*
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(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:
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:
_______________________________________________
Kamailio (SER) - Users Mailing List
sr-users(a)lists.kamailio.org
https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
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:
Kamailio (SER) - Users
Mailing List
sr-users(a)lists.kamailio.org
https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
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.
>>
>>
>>
>> _______________________________________________
>> Kamailio (SER) - Users Mailing List
>> sr-users(a)lists.kamailio.org
>> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
>>
>
>
> --
> Arsen Semenov
>
> _______________________________________________
> Kamailio (SER) - Users Mailing List
> sr-users(a)lists.kamailio.org
> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
>
_______________________________________________
Kamailio (SER) - Users Mailing List
sr-users(a)lists.kamailio.org
https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
_______________________________________________
10:15 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>*
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(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:
_______________________________________________
Kamailio (SER) - Users Mailing List
sr-users(a)lists.kamailio.org
https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
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:
_______________________________________________
Kamailio (SER) - Users Mailing List
sr-users(a)lists.kamailio.org
https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
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.
>>>
>>>
>>>
>>> _______________________________________________
>>> Kamailio (SER) - Users Mailing List
>>> sr-users(a)lists.kamailio.org
>>> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
>>>
>>
>>
>> --
>> Arsen Semenov
>>
>> _______________________________________________
>> Kamailio (SER) - Users Mailing List
>> sr-users(a)lists.kamailio.org
>> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
>>
> _______________________________________________
> Kamailio (SER) - Users Mailing List
> sr-users(a)lists.kamailio.org
> https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
>
_______________________________________________
Kamailio (SER) - Users Mailing
List
sr-users(a)lists.kamailio.org
https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
11:33 p.m.
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(a)voip.pt <sergio.charrua(a)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 1:22 PM Aristeidis Tsitras <tsitras(a)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(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>*
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(a)gmail.com>
wrote:
_______________________________________________
Kamailio (SER) - Users Mailing List
sr-users(a)lists.kamailio.org
https://lists.kamailio.org/cgi-bin/mailman/listinfo/sr-users
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:
> 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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>>
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12 comments
6 participants
participants (6)
-
Aristeidis Tsitras -
Arsen Semenov -
Daniel-Constantin Mierla -
David Villasmil -
Jurijs Ivolga -
Sergio Charrua