[asterisk-biz] Call Recording System information request

[Steve Totaro]

And then how do you associate the agent with the call?

Thanks,
Steve T

On Mon, Jun 30, 2008 at 10:05 PM, Matt Florell <astmattf at gmail.com> wrote:
> If you are using a Sangoma card you can use OrecX to record all calls
> from a T1 interface(set up as a T1 passthru).
>
> The Sangoma wanpipe drivers have an RTP-tap feature that takes the T1
> audio channels at the kernel driver level and formats them as RTP
> streams that OrecX can use to record the audio separated into calls.
>
> MATT---
>
> On 6/30/08, flavio <flavio at asteriskguide.com> wrote:
>> 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 at totarotechnologies.com>
>>  To: "Commercial and Business-Oriented Asterisk Discussion"
>>  <asterisk-biz at 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 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
>>  >> Web    : http://www.evaristesys.com/
>>  >> Tel    : (+1) (678) 954-0670
>>  >> Direct : (+1) (678) 954-0671
>>  >> Mobile : (+1) (706) 338-8599
>>  >>
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