3 Apr
2012
3 Apr
'12
3 p.m.
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 <stoyan.v.mihaylov(a)gmail.com> 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 <robwatkin(a)gmail.com> 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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