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

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
>> >>
>> >> _______________________________________________
>> >> --Bandwidth and Colocation Provided by http://www.api-digital.com--
>> >>
>> >> AstriCon 2008 - September 22 - 25 Phoenix, Arizona
>> >> Register Now: http://www.astricon.net
>> >>
>> >> asterisk-biz mailing list
>> >> To UNSUBSCRIBE or update options visit:
>> >>

http://lists.digium.com/mailman/listinfo/asterisk-biz
>> >>
>> >
>> > _______________________________________________
>> > --Bandwidth and Colocation Provided by http://www.api-digital.com--
>> >
>> > AstriCon 2008 - September 22 - 25 Phoenix, Arizona
>> > Register Now: http://www.astricon.net
>> >
>> > asterisk-biz mailing list
>> > To UNSUBSCRIBE or update options visit:
>> >

http://lists.digium.com/mailman/listinfo/asterisk-biz
>>
>>
>> _______________________________________________
>> --Bandwidth and Colocation Provided by http://www.api-digital.com--
>>
>> AstriCon 2008 - September 22 - 25 Phoenix, Arizona
>> Register Now: http://www.astricon.net
>>
>> asterisk-biz mailing list
>> To UNSUBSCRIBE or update options visit:
>>

http://lists.digium.com/mailman/listinfo/asterisk-biz
>>
>
> _______________________________________________
> --Bandwidth and Colocation Provided by http://www.api-digital.com--
>
> AstriCon 2008 - September 22 - 25 Phoenix, Arizona
> Register Now: http://www.astricon.net
>
> asterisk-biz mailing list
> To UNSUBSCRIBE or update options visit:
>

http://lists.digium.com/mailman/listinfo/asterisk-biz
>

_______________________________________________
--Bandwidth and Colocation Provided by http://www.api-digital.com--

AstriCon 2008 - September 22 - 25 Phoenix, Arizona
Register Now: http://www.astricon.net

asterisk-biz mailing list
To UNSUBSCRIBE or update options visit:

http://lists.digium.com/mailman/listinfo/asterisk-biz