[asterisk-biz] Call Recording System information request

[Steve Totaro]

On Mon, Jun 30, 2008 at 8:15 PM, Alex Balashov
<abalashov at evaristesys.com> wrote:
> Steve Totaro wrote:
>
>> OrecX will have no value with a Definity G3.  What you want to do is
>> front end your Definity system with Asterisk.
>
> It does if you bounce the calls in and out of SIP channels.

How do you do that on a Definity and still make call routing work?  I
have worked on several older systems, and configuration of a simple T1
and trunk group are difficult enough.  I think "bouncing the calls in
and out of SIP channels" sounds really really difficult, elegant, and
unneeded, but I may be wrong.  Plus, I am not sure how you would
correspond a call to an extension with all this bouncing going on.

>
>>
>> With your call volume, Asterisk's native monitor application will more
>> than suffice on any modern server.  The I/O threshold is ~60-70
>> simultaneous calls before audio starts breaking up.
>
> I agree;  this is probably a more practical route for this call volume.
>  I'm just used to Monitor() being considered inadequate for any sort of
> nontrivial load, but last time I touched it, Asterisk was neither this
> mature (pre-1.2) nor hardware this good.

To add to this OrecX would be the next step if you pass the I/O
threshold (hopefully you do, means business it good ;-)

Plus I cannot stress the added flexibilty in the way queues are
handled and the reporting of such data.

I would first put Asterisk in the middle and just get the recording
portion working, once you feel that is stable, I would consider
migrating the queue function to Asterisk as well.

Thanks,
Steve T

>
> --
> Alex Balashov
> Evariste Systems
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