[SR-Users] Load balancing traffic based on SIP URI

[Alberto Sagredo]

Great. You can share with us how did you did it :)

Others would thank you also

BR

2016-04-04 15:22 GMT+02:00 NITESH BANSAL <nitesh.bansal at outlook.com>:

> Thanks guys for your ideas, I finally think that I have an idea on how to
> do it.
>
> Nitesh
>
> ------------------------------
> Date: Fri, 1 Apr 2016 16:28:02 +0200
> From: alberto.sagredo at avanzada7.com
> To: sr-users at lists.sip-router.org
> Subject: Re: [SR-Users] Load balancing traffic based on SIP URI
>
>
> Hi Federico. In mi case i have to force some RURIs to a determinate
> Asterisk . There was some logic to do that instead letting kamailio to
> handle that situations
>
> I haven't tested option 3 , but it must do what you expect i think. Maybe
> you would need to adjust
>  ds_hash_expire
> Regards
>
> 2016-04-01 14:46 GMT+02:00 Federico Cabiddu <federico.cabiddu at gmail.com>:
>
> Hi,
> if I understand correctly your problem is that the first INVITE for a
> specific URI will go to a server and the conference will be instantiated
> there. Then you must route all the INVITEs from the other participants to
> the same server, is it right?
> If this is the case why not use algorithm 3 in ds_select_ds (or
> ds_select_domain) which does hashing over Request-URI?
> (
> http://www.kamailio.org/docs/modules/devel/modules/dispatcher.html#dispatcher.f.ds_select_dst
> )
>
> Regards,
>
> Federico
>
> On Fri, Apr 1, 2016 at 12:26 PM, NITESH BANSAL <nitesh.bansal at outlook.com>
> wrote:
>
> Hello,
>
> I want to use Kamailio to load balance traffic across multiple asterisks.
> The problem is that my SIP traffic is related to conference, so if a SIP
> call
> for a particular URI is sent to an Asterisk instance, any subsequent calls
> for that
> SIP URI should be sent to the same Asterisk instance.
> Is there anyway to achieve this in Kamailio?
>
> Thanks,
> Nitesh
>
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