[SR-Users] How to route to calls to asterisk

[David Villasmil]

Hello,

It’s actually pretty simple.
As you pointed out; just direct your softphones to specific dns names,
I.e.: ast.domain.com, ast2, etc. all of which resolve to that Kamailio
proxy.
In Kamailio add aliases for those dns names, I.e.:

alias=ast.domain.com
alias=ast2.domain.com

And when you get an INVITE, just look at the RURI and forward to the
specific asterisk server.

On Sat, 21 Nov 2020 at 13:15, Mihai Cezar <cezar at mokalife.ro> wrote:

> any clues here?
>
> On Wed, Nov 18, 2020 at 10:17 PM Mihai Cezar <cezar at mokalife.ro> wrote:
>
>> Hi all,
>>
>> I need some help on getting to work my scenario:
>>
>> I have 3 asterisk servers, and every agents (soft phones like zoiper) is
>> connecting  to which server they need by contacting the PublicIP of each
>> server. (the problem is that I need to reuse the IP address on others
>> servers) .
>>
>> I want if is possible to use a proxy like "haproxy" but for sip to
>> redirect the agents (softphones),
>> to their coresponding server.
>>
>> For example:
>> - james at ast.example.com -> must register to asterisk server 1,
>> - sarah at ast2.exampe.com -> must register to asterisk server 2
>> - andy at ast3.example.com-> must register to asterisk server 3
>>
>> ast.example.com, ast2.example.com, ast3.example.com = will share all
>> the same ip,
>>
>> when the request arrives in the proxy it should check the vhost and
>> redirect to the coresponding server for registering.
>>
>> Any hints and how to do this scenario?
>>
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-- 
Regards,

David Villasmil
email: david.villasmil.work at gmail.com
phone: +34669448337
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