<html><head><style type="text/css"><!-- DIV {margin:0px;} --></style></head><body><div style="font-family:arial, helvetica, sans-serif;font-size:10pt"><DIV style="FONT-SIZE: 10pt; FONT-FAMILY: arial, helvetica, sans-serif">It is possible to pass parameters such as a credit card numbers using the WEB API Interface.</DIV>
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<DIV style="FONT-SIZE: 10pt; FONT-FAMILY: arial, helvetica, sans-serif">Reference, press release - <A href="http://www.orecx.com/web/press/press-2008-03_ProdYear.pdf">http://www.orecx.com/web/press/press-2008-03_ProdYear.pdf</A></DIV>
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<DIV style="FONT-SIZE: 10pt; FONT-FAMILY: arial, helvetica, sans-serif">Flavio E. Goncalves<BR><BR></DIV>
<DIV style="FONT-SIZE: 12pt; FONT-FAMILY: times new roman, new york, times, serif">----- Mensagem original ----<BR>De: Matt Florell <astmattf@gmail.com><BR>Para: Commercial and Business-Oriented Asterisk Discussion <asterisk-biz@lists.digium.com><BR>Enviadas: Terça-feira, 1 de Julho de 2008 0:06:03<BR>Assunto: Re: [asterisk-biz] Call Recording System information request<BR><BR>Hello,<BR><BR>That depends on the capabilities of the system that you are passing<BR>the calls through to. If it logs the channel and time then it is easy<BR>to match up the calls to their recordings. If not, then you have a<BR>problem.<BR><BR>In the end, the best decision is to move to an all-Asterisk solution<BR>of some kind. But there are options is that is not possible.<BR><BR>MATT---<BR><BR>On 6/30/08, Steve Totaro <<A href="mailto:stotaro@totarotechnologies.com" ymailto="mailto:stotaro@totarotechnologies.com">stotaro@totarotechnologies.com</A>>
wrote:<BR>> And then how do you associate the agent with the call?<BR>><BR>> Thanks,<BR>> Steve T<BR>><BR>><BR>> On Mon, Jun 30, 2008 at 10:05 PM, Matt Florell <<A href="mailto:astmattf@gmail.com" ymailto="mailto:astmattf@gmail.com">astmattf@gmail.com</A>> wrote:<BR>> > If you are using a Sangoma card you can use OrecX to record all calls<BR>> > from a T1 interface(set up as a T1 passthru).<BR>> ><BR>> > The Sangoma wanpipe drivers have an RTP-tap feature that takes the T1<BR>> > audio channels at the kernel driver level and formats them as RTP<BR>> > streams that OrecX can use to record the audio separated into calls.<BR>> ><BR>> > MATT---<BR>> ><BR>> > On 6/30/08, flavio <<A href="mailto:flavio@asteriskguide.com" ymailto="mailto:flavio@asteriskguide.com">flavio@asteriskguide.com</A>>
wrote:<BR>> >> As far as I know, the paid version of Orecx can record from a T1 passively.<BR>> >> This is not clear in the Orecx website, please contact Orecx for further<BR>> >> details. So it should work with the Definity G3.<BR>> >><BR>> >><BR>> >> Flavio<BR>> >><BR>> >><BR>> >><BR>> >> ----- Original Message -----<BR>> >> From: "Steve Totaro" <<A href="mailto:stotaro@totarotechnologies.com" ymailto="mailto:stotaro@totarotechnologies.com">stotaro@totarotechnologies.com</A>><BR>> >> To: "Commercial and Business-Oriented Asterisk Discussion"<BR>> >> <<A href="mailto:asterisk-biz@lists.digium.com" ymailto="mailto:asterisk-biz@lists.digium.com">asterisk-biz@lists.digium.com</A>><BR>> >>
Sent: Monday, June 30, 2008 9:38 PM<BR>> >> Subject: Re: [asterisk-biz] Call Recording System information request<BR>> >><BR>> >><BR>> >> > On Mon, Jun 30, 2008 at 8:15 PM, Alex Balashov<BR>> >> > <<A href="mailto:abalashov@evaristesys.com" ymailto="mailto:abalashov@evaristesys.com">abalashov@evaristesys.com</A>> wrote:<BR>> >> >> Steve Totaro wrote:<BR>> >> >><BR>> >> >>> OrecX will have no value with a Definity G3. What you want to do is<BR>> >> >>> front end your Definity system with Asterisk.<BR>> >> >><BR>> >> >> It does if you bounce the calls in and out of SIP channels.<BR>> >> ><BR>> >> > How do you do that on a Definity and
still make call routing work? I<BR>> >> > have worked on several older systems, and configuration of a simple T1<BR>> >> > and trunk group are difficult enough. I think "bouncing the calls in<BR>> >> > and out of SIP channels" sounds really really difficult, elegant, and<BR>> >> > unneeded, but I may be wrong. Plus, I am not sure how you would<BR>> >> > correspond a call to an extension with all this bouncing going on.<BR>> >> ><BR>> >> >><BR>> >> >>><BR>> >> >>> With your call volume, Asterisk's native monitor application will more<BR>> >> >>> than suffice on any modern server. The I/O threshold is ~60-70<BR>> >> >>> simultaneous calls before
audio starts breaking up.<BR>> >> >><BR>> >> >> I agree; this is probably a more practical route for this call volume.<BR>> >> >> I'm just used to Monitor() being considered inadequate for any sort of<BR>> >> >> nontrivial load, but last time I touched it, Asterisk was neither this<BR>> >> >> mature (pre-1.2) nor hardware this good.<BR>> >> ><BR>> >> > To add to this OrecX would be the next step if you pass the I/O<BR>> >> > threshold (hopefully you do, means business it good ;-)<BR>> >> ><BR>> >> > Plus I cannot stress the added flexibilty in the way queues are<BR>> >> > handled and the reporting of such data.<BR>> >> ><BR>>
>> > I would first put Asterisk in the middle and just get the recording<BR>> >> > portion working, once you feel that is stable, I would consider<BR>> >> > migrating the queue function to Asterisk as well.<BR>> >> ><BR>> >> > Thanks,<BR>> >> > Steve T<BR>> >> ><BR>> >> >><BR>> >> >> --<BR>> >> >> Alex Balashov<BR>> >> >> Evariste Systems<BR>> >> >> Web : <A href="http://www.evaristesys.com/" target=_blank>http://www.evaristesys.com/</A><BR>> >> >> Tel : (+1) (678) 954-0670<BR>> >> >> Direct : (+1) (678) 954-0671<BR>> >> >> Mobile : (+1) (706)
338-8599<BR>> >> >><BR>> >> >> _______________________________________________<BR>> >> >> --Bandwidth and Colocation Provided by <A href="http://www.api-digital.com--/" target=_blank>http://www.api-digital.com--</A><BR>> >> >><BR>> >> >> AstriCon 2008 - September 22 - 25 Phoenix, Arizona<BR>> >> >> Register Now: <A href="http://www.astricon.net/" target=_blank>http://www.astricon.net</A><BR>> >> >><BR>> >> >> asterisk-biz mailing list<BR>> >> >> To UNSUBSCRIBE or update options visit:<BR>> >> >> <A href="http://lists.digium.com/mailman/listinfo/asterisk-biz" target=_blank>http://lists.digium.com/mailman/listinfo/asterisk-biz</A><BR>> >> >><BR>>
>> ><BR>> >> > _______________________________________________<BR>> >> > --Bandwidth and Colocation Provided by <A href="http://www.api-digital.com--/" target=_blank>http://www.api-digital.com--</A><BR>> >> ><BR>> >> > AstriCon 2008 - September 22 - 25 Phoenix, Arizona<BR>> >> > Register Now: <A href="http://www.astricon.net/" target=_blank>http://www.astricon.net</A><BR>> >> ><BR>> >> > asterisk-biz mailing list<BR>> >> > To UNSUBSCRIBE or update options visit:<BR>> >> > <A href="http://lists.digium.com/mailman/listinfo/asterisk-biz" target=_blank>http://lists.digium.com/mailman/listinfo/asterisk-biz</A><BR>> >><BR>> >><BR>> >>
_______________________________________________<BR>> >> --Bandwidth and Colocation Provided by <A href="http://www.api-digital.com--/" target=_blank>http://www.api-digital.com--</A><BR>> >><BR>> >> AstriCon 2008 - September 22 - 25 Phoenix, Arizona<BR>> >> Register Now: <A href="http://www.astricon.net/" target=_blank>http://www.astricon.net</A><BR>> >><BR>> >> asterisk-biz mailing list<BR>> >> To UNSUBSCRIBE or update options visit:<BR>> >> <A href="http://lists.digium.com/mailman/listinfo/asterisk-biz" target=_blank>http://lists.digium.com/mailman/listinfo/asterisk-biz</A><BR>> >><BR>> ><BR>> > _______________________________________________<BR>> > --Bandwidth and Colocation Provided by <A href="http://www.api-digital.com--/"
target=_blank>http://www.api-digital.com--</A><BR>> ><BR>> > AstriCon 2008 - September 22 - 25 Phoenix, Arizona<BR>> > Register Now: <A href="http://www.astricon.net/" target=_blank>http://www.astricon.net</A><BR>> ><BR>> > asterisk-biz mailing list<BR>> > To UNSUBSCRIBE or update options visit:<BR>> > <A href="http://lists.digium.com/mailman/listinfo/asterisk-biz" target=_blank>http://lists.digium.com/mailman/listinfo/asterisk-biz</A><BR>> ><BR>><BR>> _______________________________________________<BR>> --Bandwidth and Colocation Provided by <A href="http://www.api-digital.com--/" target=_blank>http://www.api-digital.com--</A><BR>><BR>> AstriCon 2008 - September 22 - 25 Phoenix, Arizona<BR>> Register Now: <A href="http://www.astricon.net/" target=_blank>http://www.astricon.net</A><BR>><BR>>
asterisk-biz mailing list<BR>> To UNSUBSCRIBE or update options visit:<BR>> <A href="http://lists.digium.com/mailman/listinfo/asterisk-biz" target=_blank>http://lists.digium.com/mailman/listinfo/asterisk-biz</A><BR>><BR><BR>_______________________________________________<BR>--Bandwidth and Colocation Provided by <A href="http://www.api-digital.com--/" target=_blank>http://www.api-digital.com--</A><BR><BR>AstriCon 2008 - September 22 - 25 Phoenix, Arizona<BR>Register Now: <A href="http://www.astricon.net/" target=_blank>http://www.astricon.net</A><BR><BR>asterisk-biz mailing list<BR>To UNSUBSCRIBE or update options visit:<BR> <A href="http://lists.digium.com/mailman/listinfo/asterisk-biz" target=_blank>http://lists.digium.com/mailman/listinfo/asterisk-biz</A><BR></DIV></div></body></html>