[SR-Users] Kamailio Proxy and Asterisk

[Stoyan Mihaylov]

In Asterisk - sip.conf
[MyKamailioUser]
type = friend
host = 192.168.2.2 - internal IP of Kamailio
insecure = port,invite
context = {where you would like to get calls - you can use default, or not
use this part at all}
In Asterisk - only authentication is based on IP of Kamailio server.

In Kamailio, I am using:
ds_select_dst("1","4");
to get Asterisk from pool (if you have more then one asterisk behind
Kamailio)
and then - t_relay, instead of forward, because my Asterisk has only
internal IP. Calls are sent back to Kamailio in case of calling other user,
or directly to provider. Asterisk also is connected via registrar to
provider for incomming calls.
In my configuration, I am using some dirty solutions for problems with ACK
and BYE, but more or less everything works fine. rtpproxy - solves problems
with NAT, and users must not use STUN.


On Tue, Jan 31, 2012 at 8:50 AM, Sammy Govind <govoiper at gmail.com> wrote:

> Hi,
>
> Kamailio is definitely the exact tool for this purpose, I have exactly the
> same setup running as yours and for scalability we started using Kamailio
> in front of our asterisk servers. Long story short, read these articles.
>
> http://kb.asipto.com/asterisk:realtime:kamailio-3.1.x-asterisk-1.6.2-astdb
>
> http://saevolgo.blogspot.com/2011/11/increasing-voip-services-capacity.html
>
>
> Once after setting up the environments you don't need Public IPs on your
> asterisk servers. Just only for Kamailio server(s)
>
> Regards,
> Sammy.
>
>
> On Tue, Jan 31, 2012 at 11:16 AM, Greg Mannie <greg at latigi.com> wrote:
>
>> Hello,
>>
>> We have been using Asterisk for sometime and over the last year have
>> started hosting instances for our clients on a vmware platform.  These
>> virtual pbx are located on public ip addresses and each customer has their
>> own SIP trunk arrangements with various providers.   We have decided to
>> pool our resources and would like to start aggregating traffic.
>>
>> From reading online I have installed Kamailio 3.1.x with Asterisk 1.6.x.
>> On top of this I have installed Siremis 3.2 in hopes of using it as a
>> graphical front end.
>>
>> I am wanting to use Kamailio as a proxy to the SIP providers and allow
>> the Asterisk to register only with Kamailio.  Is there a link to some
>> example configs with a tutorial pertaining to this type of deployment?
>>
>> If this is a good solution, it is my intention to recommend a consultant
>> be hired to assist with a production server.  I just need to learn allot
>> more, before I can recommend Kamailio for the job.
>>
>> Do you think it's a good fit?
>>
>> Thank you for your kind responses,
>>
>> Greg
>>
>>
>>
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>>
>
>
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