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" <[EMAIL PROTECTED]> 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 > <[EMAIL PROTECTED]> 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