[Freeswitch-users] Contemplating a jump from Asterisk to Freeswitch

[David Knell]

Hi Geoff,

> One of the benefits of our architecture is that our business logic is
> completely abstracted from the asterisk system. We use a combination
> of FastAGI and AMI to control channels on the asterisk server. We have
> a Java based server which interfaces with the higher level call
> routing engines. It looks to me like the Mod_event_socket would
> probably satisfy my requirements for controlling the calls via an
> external process, although it doesn't look as cut/dry as the FastAGI
> model. I haven't seen anything which would let me know the equivalent
> of the FastAGI 'script' being requested.

Three possibilities spring to mind:-
* have each distinct 'script' listen on a different socket;
* set a variable in the dialplan to a script name or other identifier
before making the outbound socket connection;
* have your event socket handler work out what to do itself based on the
dialled number, or whatever other criteria you'd use.

> The other thing I haven't seen is how to dynamically create
> conferences on the fly and redirect channels into them. We use
> app_conference on asterisk to avoid the ztdummy issue. Once the higher
> level intelligence engine determines two channels need to speak with
> each other, they are both redirected via AMI Redirect into a dynamic
> Conference created just for that particular call.

Choose a (unique) conference ID, and execute
conference <id>
on each of the channels.

> Also - what is the status of call progress on FreeSwitch? Some things
> that are important to me are answering machine detection as well as
> detecting SIT intercept tones in the early media stream... any love
> here?

Not sure on these, but I'm *am* sure that someone else will be ;-)

Cheers --

Dave

-- 
David Knell, Director, 3C Limited
T: +44 20 3298 2000
E: dave at 3c.co.uk
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