[SR-Users] SIP Trunks Location

[Stoyan Mihaylov]

Surely there must be a better way.
For example, if you are a good client for your provider, you can ask him to
forward all calls to your IP. This way, Kamailio will accept all calls and
forward them to multiple Asterisk boxes for processing.
Also - you can put one Asterisk to do only one thing - accept all calls
(from provider), and forward them (to other Asterisk boxes, or to
Kamailio).
May be there are better ways, but in this moment, I cannot imagine more.
PS
There are many providers, and some of them can forward calls. It is good
idea to see if your provider can or cannot really forward calls to your IP.


On Tue, Jun 25, 2013 at 8:24 AM, Jose Suero <ms at mstn.com> wrote:

> Stoyan thanks for your reply, i've been doing some research before
> replying (which has taken a while) and there's something I don't
> understand. I apologize in advance if I'm asking something that makes no
> sense.
>
> my provider does in fact requires registration, and they provide a single
> sip that accepts hundreds of concurrent calls.
>
> If I have a kamailio in front of several pbx servers, in order to have
> redundancy (if server fails) and be able to handle thousands of calls, but
> have to route all outgoint (pstn) calls to a single asterisk/freeswitch
> server that's actually registered with my provider, wouldn't I loose all my
> redundancy and concurrency capability??
>
> Is there a better way??
>
> thanks in advance
>
>
>
> On 2013-06-19 13:19, Stoyan Mihaylov wrote:
>
>> It depends.
>> I can imagine next scenarios:
>> 1. Under SIP trunks you mean calls from your provider to you
>> A) In case your provider can send calls to you - then you can use
>> Kamailio, accepting all calls from your provider - based on IP.
>> B) In case your provider expects registration from your system - then,
>> at least I - dont know how to do only with Kamailio - Asterisk can
>> register easily to every provider.
>> 2. Under SIP trunks you mean calls from you to World through your
>> provider.
>> A) Your provider can accept all calls from you based on IP - Kamailio
>> can directly forward calls to your provider.
>> B) Your provider expects authentication - then again I dont know how
>> this can be done through Kamailio, but Asterisk can do it easily.
>>
>> My suggestion is - you can use Kamailio for registration of users and
>> load balancing, and asterisk servers for everything else.
>>
>> On Wed, Jun 19, 2013 at 7:30 PM, Jose Suero <ms at mstn.com [3]> wrote:
>>
>>  Hi
>>>
>>>
>>> Im planning to set kamailio in front of an farm of pbx servers
>>> (havent decided on freeswitch or asterisk) theres a million
>>> tutorials on how to do this, what I havent found is what part of my
>>>
>>> setup actually handles the sip trunks my phone company provides me
>>> with.
>>>
>>> Whats the best practice when It comes to this?
>>>
>>>
>>> Is kamailio going to be receiving the calls from the trunk and
>>> passing them to the PBX or is it the other way around?
>>>
>>> please advice
>>>
>>> Thanks in advance
>>>
>>> Jose Suero
>>>
>>> ______________________________**_________________
>>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing
>>> list
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>>>
>>
>>
>>
>> Links:
>> ------
>> [1] mailto:sr-users at lists.sip-**router.org<sr-users at lists.sip-router.org>
>> [2] http://lists.sip-router.org/**cgi-bin/mailman/listinfo/sr-**users<http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users>
>> [3] mailto:ms at mstn.com
>>
>
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