Re: [asterisk-biz] Call Recording System information request

As far as I know, the paid version of Orecx can record from a T1 passively.
This is not clear in the Orecx website, please contact Orecx for further
details. So it should work with the Definity G3.

Flavio

----- Original Message -----
From: "Steve Totaro" <stotaro@totarotechnologies.com>
To: "Commercial and Business-Oriented Asterisk Discussion"
<asterisk-biz@lists.digium.com>
Sent: Monday, June 30, 2008 9:38 PM
Subject: Re: [asterisk-biz] Call Recording System information request

> On Mon, Jun 30, 2008 at 8:15 PM, Alex Balashov
> <abalashov@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
>> Web : http://www.evaristesys.com/
>> Tel : (+1) (678) 954-0670
>> Direct : (+1) (678) 954-0671
>> Mobile : (+1) (706) 338-8599
>>
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