Hi Stoyan,

Does that mean that you use Asterisk as pure SIP PSTN Gateways? I imagined
Asterisk as a physical PSTN gateway but thought that Kamailio/RTPProxy
would scale out better for pure SIP. I was planning to use Asterisk or
FreeSWITCH as a media server for hold, VM, conference and IVR.

Rob

On 3 April 2012 13:23, Stoyan Mihaylov <[email protected]> wrote:

> We do just that way - Kamailo to handle load balancing and clients, and
> Asterisk servers for routing, gateway etc....
> You can forward calls to your main gateway which then will work with other
> gateways or calls from gateways to clients registered in Kamailio.
>
> On Tue, Apr 3, 2012 at 3:15 PM, Rob Watkin <[email protected]> wrote:
>
>> Hi,
>>
>> I am just getting started with Kamailio and have been following the book
>> "Building Telephony Systems with OpenSER" by Flavio E. Goncalves. The book
>> describes an architecture with a SIP Proxy handling registrations and
>> handing calls to a PSTN Gateway. I now have a basic test network running
>> where calls are routed via the SIP Proxy (Kamailio) to a third party PSTN
>> Gateway. I feel that a better design would be to implement my own PSTN
>> Gateway using Kamailio. This single gateway would then handle all third
>> party PSTN gateways. Thus one Kamailio server would be facing my clients
>> while another would be facing my suppliers.
>>
>> Is this a sensible architecture and are there any sample configurations
>> for Kamailio performing this role?
>>
>> Thanks
>> Rob Watkin
>>
>>
>>
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>
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