You can use OpenSIPS call_control module to terminate the calls on
both side.
Adrian
On Jan 14, 2009, at 10:26 AM, ram wrote:
On 1/14/09, Mark Sayer <[email protected]> wrote:
I suggest using the pieces as they work best. Let OpenSIPs handle the
registration & NAT. Let Asterisk handle the media & connections to
terminators or PSTN. The only issue is that Asterisk will only handle
about 200 concurrent calls per box so a large installation might have
a single OpenSIPs box and multiple Asterisk boxes. Relatively simple
to setup and manage, stable, proven. Asterisk itself can be
"partitioned" through careful construction of the extensions.conf
file to do what you want.
Hi
thanks for the suggestion
thats what iam trying to achieve Asterisk (or freeswitch)
the suggestions again needed here
use Dispatcher Module or Drouting or LCR is again question
if its post paid fine
If its prepaid, how does that work of the Asterisk disconnect the call
how the Opensip react on the same
may be some are odd question, these all i have to understand
Ram
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