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

Hello,

That depends on the capabilities of the system that you are passing
the calls through to. If it logs the channel and time then it is easy
to match up the calls to their recordings. If not, then you have a
problem.

In the end, the best decision is to move to an all-Asterisk solution
of some kind. But there are options is that is not possible.

MATT---

On 6/30/08, Steve Totaro <stotaro@totarotechnologies.com> wrote:
> 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@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@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@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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