[SR-Users] two way routing

[Daniel-Constantin Mierla]

Sipsak or other tool pointing via Contact header to Kamailio is still an
option, but you can use directly uac module -- you have to set the
details for remote registrations inside uac_reg database table and
configure uac module, some details also at:

  - https://kamailio.org/docs/modules/5.0.x/modules/uac.html#idp38102932

Cheers,
Daniel


On 14.06.17 18:17, Jean Cérien wrote:
>
> Thanks, I went for the permission + dispatcher, and so far it looks good.
>
> One last thing.... I need the kamailio server to explicitly register
> to a provider. Googling the archive, I could only find an old
> reference to use sipsak. Does that mean that sipsak (or any other sip
> crafting tool like sipsak) should send a register request to Kamailio,
> which in turn will forward it to the voip provider ?
>
> Here is the
> message: https://lists.kamailio.org/pipermail/sr-users/2004-November/041304.html
>
> Rgds
> J
>
>
> On Tue, Jun 13, 2017 at 3:01 AM, Daniel-Constantin Mierla
> <miconda at gmail.com <mailto:miconda at gmail.com>> wrote:
>
>     Hello,
>
>     I typically blend several modules to achieve your routing needs:
>
>       - permissions module with address table to detect if calls are
>     coming from providers or asterisk farm
>
>       - based on result use load balancer to send out to asterisk farm
>
>       - or use pdt (if each provider has one gateway) or
>     mtree+dispatcher (mtree to translate from a prefix to a dispatcher
>     set id of the provider with many gateways) to send out to providers
>
>     If least cost routing is needed towards providers, then you can
>     decide to use one of next modules:
>
>        - lcr
>
>        - carrierroute
>
>        - drouting
>
>     Cheers,
>     Daniel
>
>
>     On 12.06.17 21:21, Jean Cérien wrote:
>>
>>     Hello
>>
>>     I am looking at the following configuration: 
>>
>>     1/ voip providers, with different external sip servers, will send
>>     calls to kamailio, that will dispatch them to asterisk servers,
>>     with load balancing & failover. The IP of the asterisk servers
>>     are known and fixed.
>>
>>     2/ the asterisk servers will also place calls, going through
>>     kamailio, that will dispatch them to the voip provider sip
>>     servers, based on R-URI / Prefix.
>>
>>     I've achieved #1 with dispatcher module, I've seen carrierroute
>>     module that can achieve #2, but I dont really grasp how to put
>>     the two together.
>>
>>     I have the feeling that #1 can also be done with carrierroute,
>>     and that I only need to check if the 'from' domain is one my
>>     asterisk, and chose a different routing table, but I dont see how
>>     to do that
>>
>>     is that a valid approch ? where should I look to do that ?
>>
>>     thanks
>>
>>     J.
>>
>>
>>     _______________________________________________
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>
>     -- 
>     Daniel-Constantin Mierla
>     www.twitter.com/miconda <http://www.twitter.com/miconda> -- www.linkedin.com/in/miconda <http://www.linkedin.com/in/miconda>
>     Kamailio Advanced Training - www.asipto.com <http://www.asipto.com>
>     Kamailio World Conference - www.kamailioworld.com <http://www.kamailioworld.com>
>
>     _______________________________________________ Kamailio (SER) -
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>
-- 
Daniel-Constantin Mierla
www.twitter.com/miconda -- www.linkedin.com/in/miconda
Kamailio Advanced Training - www.asipto.com
Kamailio World Conference - www.kamailioworld.com
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