Tuesday, July 1, 2008

Re: [asterisk-biz] anyone want to take a look at a RFP?

Hi Bret,

Yep this was the right email list to post to. I’m sure a number of people would be interested in bidding.

 

It might help if you post your location and a rough idea about how large your call center is/how many extensions you plan on implementing etc. You might get more meaningful responses.

 

 

Regards,

Dean Collins
dean@cognation.net

+1-212-203-4357 (Direct)
+1-917-207-3420 (Mobile)
+61-2-9016-5642 (Sydney in-dial)
http://www.Cognation.net

From: asterisk-biz-bounces@lists.digium.com [mailto:asterisk-biz-bounces@lists.digium.com] On Behalf Of Bret Peterson
Sent: Tuesday, 1 July 2008 12:46 PM
To: Commercial and Business-Oriented Asterisk Discussion
Subject: Re: [asterisk-biz] anyone want to take a look at a RFP?

 

Hi All,

 

I am not sure if this is the right list to post this to but the company that I work for is getting ready to release an RFP for a new phone system and personally, I think that Asterisk/Trixbox would be a great fit given all the "we have to have" capabilities that they are requesting (in house conference calling...video conferencing...remote office call tracking...call recording...and other call center capabilities). My recommendations have been pushed aside in the past and will probably be pushed aside again but I would like to get at least a proposal or two to throw into the ring to show the powers that be that this is a viable solution and cost effective. The funny thing is that the new network administrator that is writing the proposal had a salesman just stop by his desk and start touting asterisk as "the perfect solution for conference calling for you all! Have you ever heard of it?" I couldn't resist and had to pipe in but he is not selling any of the other capabilities. Just made me sort of laugh. I also had a Cisco guy last year ask me if I had ever heard of Trixbox or Asterisk...they were starting to sell it as a low end solution for those who couldn't afford Cisco Call Manager...

 

I hope that Jim Capp and his gang take a look since Jim is an old friend and a "wicked smart guy" but let me know and I will send a copy of the RFP to anyone interested...

 

Thanks for your time,

 

Bret

Posted by at No comments:

Re: [asterisk-biz] anyone want to take a look at a RFP?

Hi All,
 
I am not sure if this is the right list to post this to but the company that I work for is getting ready to release an RFP for a new phone system and personally, I think that Asterisk/Trixbox would be a great fit given all the "we have to have" capabilities that they are requesting (in house conference calling...video conferencing...remote office call tracking...call recording...and other call center capabilities). My recommendations have been pushed aside in the past and will probably be pushed aside again but I would like to get at least a proposal or two to throw into the ring to show the powers that be that this is a viable solution and cost effective. The funny thing is that the new network administrator that is writing the proposal had a salesman just stop by his desk and start touting asterisk as "the perfect solution for conference calling for you all! Have you ever heard of it?" I couldn't resist and had to pipe in but he is not selling any of the other capabilities. Just made me sort of laugh. I also had a Cisco guy last year ask me if I had ever heard of Trixbox or Asterisk...they were starting to sell it as a low end solution for those who couldn't afford Cisco Call Manager...
 
I hope that Jim Capp and his gang take a look since Jim is an old friend and a "wicked smart guy" but let me know and I will send a copy of the RFP to anyone interested...
 
Thanks for your time,
 
Bret
Posted by at No comments:

[asterisk-biz] The S word: Asterisk security

Hi all,

As I mentioned briefly in the SIP takeover thread, I'd like to try to
talk about security this coming Friday. I realize it is a holiday in
the USA, but do geeks ever take a day off, especially
security-conscious geeks? Mark Spencer once said "The Bug Tracker is
never on vacation!".

We will try to start this subject this Friday, but I have no
experience at all with this. If you know anyone who is good in this
area and would like to share their expertise and talk about security
in the asterisk and voip contexts, I'd like to hear from them,
especially next Friday July 4th.

tia,

Randy

_______________________________________________
--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

Posted by at No comments:

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

Again, I am not sure you are going to get a Definity to use a "WEB API
Interface".

I also wouldn't consider RTP TAP to be "passive". Passive is using
mirrored ports on a switch and sniffing packets, which is totally
invisible to both systems. I would consider RTP TAP to be very
active.

The bottom line is that OP is trying to capture 20 simultaneous calls,
Asterisk can handle that with no problem.

Thanks,
Steve Totaro

On Tue, Jul 1, 2008 at 7:25 AM, Flavio Goncalves
<flavio@asteriskguide.com> wrote:
> It is possible to pass parameters such as a credit card numbers using the
> WEB API Interface.
>
> Reference, press release -
> http://www.orecx.com/web/press/press-2008-03_ProdYear.pdf
>
> Flavio E. Goncalves
>
> ----- Mensagem original ----
> De: Matt Florell <astmattf@gmail.com>
> Para: Commercial and Business-Oriented Asterisk Discussion
> <asterisk-biz@lists.digium.com>
> Enviadas: Terça-feira, 1 de Julho de 2008 0:06:03
> Assunto: 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
>> >>

_______________________________________________
--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

Posted by at No comments:

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

On Mon, Jun 30, 2008 at 11:06 PM, Matt Florell <astmattf@gmail.com> wrote:
> 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.

The Definity systems I have worked on did not have that ability.

Going all Asterisk was kind of the route I was suggesting without
suggesting it outright.

Taking small steps in that direction. Most people that have Definity
systems (in my experience) do not want to migrate away since the have
such a large investment in their gear already.

When they realize that by adding Asterisk in the middle, they can take
advantage of VoIP and all things Asterisk, it begins to become a no
brainer to stop buying/replacing the digital sets for the Definity
with Polycom (or whatever) SIP phones and hang them right off the
Asterisk box, or get the admin staff on Polycoms and give them an
"Upgrade". Remote agents.....

Thanks,
Steve T

Thanks,
Steve

>
> 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---
>> >

_______________________________________________
--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

Posted by at No comments:

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

It is possible to pass parameters such as a credit card numbers using the WEB API Interface.
 
 
Flavio E. Goncalves

----- Mensagem original ----
De: Matt Florell <astmattf@gmail.com>
Para: Commercial and Business-Oriented Asterisk Discussion <asterisk-biz@lists.digium.com>
Enviadas: Terça-feira, 1 de Julho de 2008 0:06:03
Assunto: 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
>  >>  >>
>  >>  >> _______________________________________________
>  >>  >> --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
>

_______________________________________________
--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
Posted by at No comments:

Re: [asterisk-biz] Datacenters

On 30 Jun 2008, at 18:44, emist wrote:

> I'm looking around at the datacenters in my area(Florida). Can
> anyone in the list recommend a good place to collocate asterisk
> boxes in the east coast?

You are lucky to have Nap of the Americas local to you

http://www.terremark.com/technology-platform/nap-of-the-americas.aspx

This facility is a fantastic place to pick up a wide range of
connectivity options, and you can also interconnect widely to a number
of providers for resiliency and performance.

Andy

--
// www.netsumo.com // Expert ISP Consultancy //

_______________________________________________
--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

Posted by at No comments:
Subscribe to: Posts (Atom)