[SR-Users] SBC

[andrzej.ciupek at asterisk.edu.pl]

Hello

Or start with this tutorial:

http://kb.asipto.com/asterisk:realtime:kamailio-3.3.x-asterisk-10.7.0-astdb

later You can change it to use dispatcher module to have load balance  
between couple od Asterisk GW.

But try to change configuration to use WITH_PSTN, not WITH_ASTERISK,  
that won't forward registration to Asterisk, only INVITE's, in PSTN  
Route implement dispatcher module.

Greetings
Andrzej

Cytowanie davy van de moere <davy.van.de.moere at gmail.com>:

> Stampeding on an open door here, and not wanting to start a pun war ;)
>
> http://www.amazon.com/Building-Telephony-Systems-OpenSIPS-1-6/dp/1849510741/ref=sr_1_1?ie=UTF8&qid=1361373796&sr=8-1&keywords=kamailio
>
> Offcourse, it's not *AS* good as Kamailio, but if your into the book  
>  thing, it might help you to get your head around how the thing   
> actually works.
>
> The question you ask, is actually a close to default setting of   
> Kamailio, so I think you need to or:
>
> 1/ get your head around routing sip packets, which at starters is   
> mindblowing, so give yourself some time.
> 2/ pay someone to set a kamailio up for you which does what you   
> want. (I think almost anyone on the mailinglist here, can do that in  
>  a matter of hours)
>
> Enjoy!
>
>
> On 19 Feb 2013, at 10:52, Keith wrote:
>
>> Hi,
>> I am looking to build an SBC/SIP router made up from Kamailio and   
>> FreePBX. Calls will then need to be passed to multiple Asterisk   
>> media gateways. Can anyone point me to some documentation or help   
>> on how to do this?
>>
>> --
>> Keith
>> _______________________________________________
>> SIP Express Router (SER) and Kamailio (OpenSER) - sr-users mailing list
>> sr-users at lists.sip-router.org
>> http://lists.sip-router.org/cgi-bin/mailman/listinfo/sr-users
>
>