Monday, June 30, 2008

[asterisk-biz] H323 installation needed ($$$)

I am after someone to help me to config H323 on asterisk if possible since I
am far too busy stuck on another project. Interested parties please msn me
on sam _ _ tam AT hotmail.com please take out all space and change AT to @

If you are unsure then you can always email me with your contact via my
gmail account.
Sam


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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:
> >> >

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

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> >>
> >
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> >
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> > Register Now: http://www.astricon.net
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> >

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> >
>
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>
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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
>
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>

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>

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

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
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>

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>

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

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:
>

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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
>

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Re: [asterisk-biz] Datacenters

Thanks for the help everyone.
Btw anyone has any experience with www.carrierhouse.us?


Thanks again,

Igor H.

Max Clark wrote:
> I would say Host.net, there is also Verio but I don't have experience with them.
>
> On Mon, Jun 30, 2008 at 1:16 PM, emist <emistz@gmail.com> wrote:
>> Miami, but anywhere in a 100 miles radius will probably do.
>>
>> Max Clark wrote:
>>> Where in Florida are you?
>>>
>>> On Mon, Jun 30, 2008 at 10:44 AM, emist <emistz@gmail.com> wrote:
>>>> Hey,
>>>>
>>>> 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?
>>>>
>>>> Have a good one,
>>>>
>>>> Igor H.
>>>>
>>>> _______________________________________________
>>>> --Bandwidth and Colocation Provided by http://www.api-digital.com--
>>>>
>>>> AstriCon 2008 - September 22 - 25 Phoenix, Arizona
>>>> Register Now: http://www.astricon.net
>>>>
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>>>> 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
>>>
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>>> To UNSUBSCRIBE or update options visit:
>>>

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>>>
>>
>> _______________________________________________
>> --Bandwidth and Colocation Provided by http://www.api-digital.com--
>>
>> AstriCon 2008 - September 22 - 25 Phoenix, Arizona
>> Register Now: http://www.astricon.net
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>>

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

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.

>
> 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.

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

On Mon, Jun 30, 2008 at 4:02 PM, Alex Balashov
<abalashov@evaristesys.com> wrote:
> Erick Perez wrote:
>> HI,
>> I am looking for an open source with paid support or a fully developed
>> call recording solution based on Asterisk + digium or sangoma cards
>> (no Dialogic please). It must support E1, no support for T1 or ISDN is
>> available in our area.
>>
>> Our customer has a contact center that must record all calls, all the
>> time. for about 20 agents.
>> We can assemble the server here if needed on HP Proliant (rack or
>> tower) hardware.
>> Customer need reporting on calls for the contact center manager as
>> well as for the CIO, ability to retrieve a saved call. daily backup,
>> capacity to move backup to offline storage (tape or DVD-R).
>> System will be installed agains an Alcatel 4100 system. It has two E1
>> coming in shared among the administration offices and the contact
>> center. It also has an unused E1 card. All phones are digital.
>>
>> Thanks in advance.
>
> Consider OrecX (www.orecx.com).
>
>

OrecX will have no value with a Definity G3. What you want to do is
front end your Definity system with Asterisk.

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 suggest a good server with dual power supplies, transparent RAID and
a great deal of HD space. Basically, you are going to be acting as
the "man in the middle" and using the monitor app to record the calls.

Then you can process them somewhere else, such as a NAS device. If
your HD fills with recordings, the system stops....

There are several packages to put your recordings into a GUI. You
could actually use Asterisk's queues to deliver calls directly to the
Definity extensions, so the recordings will match up with the agent.

Another benefit to this is using queuemetrics, a home brewed queue
stat program, or some of the other open queue analyzers out there.
You will also have greater flexibility over how your queues are
configured.

This is a very interesting project. I would love to help you get it setup.

Thanks,
Steve Totaro

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

Erick,

Where are you located?

Also, what sort of reporting does the client want? Being specific, are
they just looking for all calls to be recorded, or do they want each
call mapped back to a specific extension?

Kind Regards
Stuart

Erick Perez wrote:
> HI,
> I am looking for an open source with paid support or a fully developed
> call recording solution based on Asterisk + digium or sangoma cards
> (no Dialogic please). It must support E1, no support for T1 or ISDN is
> available in our area.
>
> Our customer has a contact center that must record all calls, all the
> time. for about 20 agents.
> We can assemble the server here if needed on HP Proliant (rack or
> tower) hardware.
> Customer need reporting on calls for the contact center manager as
> well as for the CIO, ability to retrieve a saved call. daily backup,
> capacity to move backup to offline storage (tape or DVD-R).
> System will be installed agains an Alcatel 4100 system. It has two E1
> coming in shared among the administration offices and the contact
> center. It also has an unused E1 card. All phones are digital.
>
> Thanks in advance.
>
>
>


--
Stuart Elvish
www.stuartelvish.com

Telephone 03 8888 5361 (+61 3 8888 5361)
Mobile 0408 873 601 (+61 408 873 601)
Facsimile 03 9018 4304 (+61 3 9018 4304)
Email stuart.elvish@stuartelvish.com

Visit www.stuartelvish.com for simple, seamless, smart solutions.


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Re: [asterisk-biz] Datacenters

I would say Host.net, there is also Verio but I don't have experience with them.

On Mon, Jun 30, 2008 at 1:16 PM, emist <emistz@gmail.com> wrote:
> Miami, but anywhere in a 100 miles radius will probably do.
>
> Max Clark wrote:
>> Where in Florida are you?
>>
>> On Mon, Jun 30, 2008 at 10:44 AM, emist <emistz@gmail.com> wrote:
>>> Hey,
>>>
>>> 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?
>>>
>>> Have a good one,
>>>
>>> Igor H.
>>>
>>> _______________________________________________
>>> --Bandwidth and Colocation Provided by http://www.api-digital.com--
>>>
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>>> To UNSUBSCRIBE or update options visit:
>>>

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

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>>
>
>
> _______________________________________________
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>

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Re: [asterisk-biz] Datacenters

Terramark - Miami
Host.net - Delray Beach - lchesal@host.net

Alan
www.group2call.com

--- On Mon, 6/30/08, emist <emistz@gmail.com> wrote:
From: emist <emistz@gmail.com>
Subject: Re: [asterisk-biz] Datacenters
To: "Commercial and Business-Oriented Asterisk Discussion" <asterisk-biz@lists.digium.com>
Date: Monday, June 30, 2008, 1:16 PM

Miami, but anywhere in a 100 miles radius will probably do.

Max Clark wrote:
> Where in Florida are you?
>
> On Mon, Jun 30, 2008 at 10:44 AM, emist <emistz@gmail.com> wrote:
>> Hey,
>>
>> 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?
>>
>> Have a good one,
>>
>> Igor H.
>>
>> _______________________________________________
>> --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
>>
>
> _______________________________________________
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>
> AstriCon 2008 - September 22 - 25 Phoenix, Arizona
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>
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> http://lists.digium.com/mailman/listinfo/asterisk-biz
>


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Re: [asterisk-biz] Datacenters

Miami, but anywhere in a 100 miles radius will probably do.

Max Clark wrote:
> Where in Florida are you?
>
> On Mon, Jun 30, 2008 at 10:44 AM, emist <emistz@gmail.com> wrote:
>> Hey,
>>
>> 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?
>>
>> Have a good one,
>>
>> Igor H.
>>
>> _______________________________________________
>> --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
>>
>
> _______________________________________________
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>
> AstriCon 2008 - September 22 - 25 Phoenix, Arizona
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>

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>


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

Erick Perez wrote:
> HI,
> I am looking for an open source with paid support or a fully developed
> call recording solution based on Asterisk + digium or sangoma cards
> (no Dialogic please). It must support E1, no support for T1 or ISDN is
> available in our area.
>
> Our customer has a contact center that must record all calls, all the
> time. for about 20 agents.
> We can assemble the server here if needed on HP Proliant (rack or
> tower) hardware.
> Customer need reporting on calls for the contact center manager as
> well as for the CIO, ability to retrieve a saved call. daily backup,
> capacity to move backup to offline storage (tape or DVD-R).
> System will be installed agains an Alcatel 4100 system. It has two E1
> coming in shared among the administration offices and the contact
> center. It also has an unused E1 card. All phones are digital.
>
> Thanks in advance.

Consider OrecX (www.orecx.com).


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

I am truly sorry.
I meant a Definity PBX. please forget the alcatel thing.

On Mon, Jun 30, 2008 at 2:51 PM, Erick Perez <eaperezh@gmail.com> wrote:
> HI,
> I am looking for an open source with paid support or a fully developed
> call recording solution based on Asterisk + digium or sangoma cards
> (no Dialogic please). It must support E1, no support for T1 or ISDN is
> available in our area.
>
> Our customer has a contact center that must record all calls, all the
> time. for about 20 agents.
> We can assemble the server here if needed on HP Proliant (rack or
> tower) hardware.
> Customer need reporting on calls for the contact center manager as
> well as for the CIO, ability to retrieve a saved call. daily backup,
> capacity to move backup to offline storage (tape or DVD-R).
> System will be installed agains an Alcatel 4100 system. It has two E1
> coming in shared among the administration offices and the contact
> center. It also has an unused E1 card. All phones are digital.
>
> Thanks in advance.
>
>
> --
> ------------------------------------------------------------
> Erick Perez
> ------------------------------------------------------------
>

--
------------------------------------------------------------
Erick Perez
Panama Sistemas
Integradores de Telefonia IP y Soluciones Para Centros de Datos
Panama, Republica de Panama
Cel Panama. +(507) 6694-4780
------------------------------------------------------------

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[asterisk-biz] Call Recording System information request

HI,
I am looking for an open source with paid support or a fully developed
call recording solution based on Asterisk + digium or sangoma cards
(no Dialogic please). It must support E1, no support for T1 or ISDN is
available in our area.

Our customer has a contact center that must record all calls, all the
time. for about 20 agents.
We can assemble the server here if needed on HP Proliant (rack or
tower) hardware.
Customer need reporting on calls for the contact center manager as
well as for the CIO, ability to retrieve a saved call. daily backup,
capacity to move backup to offline storage (tape or DVD-R).
System will be installed agains an Alcatel 4100 system. It has two E1
coming in shared among the administration offices and the contact
center. It also has an unused E1 card. All phones are digital.

Thanks in advance.


--
------------------------------------------------------------
Erick Perez
------------------------------------------------------------

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Re: [asterisk-biz] Datacenters

Where in Florida are you?

On Mon, Jun 30, 2008 at 10:44 AM, emist <emistz@gmail.com> wrote:
> Hey,
>
> 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?
>
> Have a good one,
>
> Igor H.
>
> _______________________________________________
> --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
>

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[asterisk-biz] Datacenters

Hey,

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?

Have a good one,

Igor H.

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Sunday, June 29, 2008

[asterisk-biz] +599 country codes services?

A friend of mine has asked me to do some research, Anyone on list offering +599 country code SIP indial numbers?

 

If so please email me pricing and details. Single number Low volume service required.

 


Cheers,

Dean

 

[asterisk-biz] We are looking for Toll Free DIDs from Brazil & Turkey via SIP

Thanks
Moshe

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[asterisk-biz] Druid Open Source Events - Druid Miami Meetup (18 Jul), OSCON (21-25 Jul), Druid London Meetup (22 Jul) & LinuxWorld (4-7 Aug)

Dear Asterisk users,

Voiceroute will be at exhibiting and presenting at the below open source communications related conferences, Druid meetups speaking about Druid & Asterisk. We would like to meet with other fellow Asterisk enthusiasts who may be at OSCON & LinuxWorld.

Mark Spencer will be speaking at OSCON 2008
http://en.oreilly.com/oscon2008/public/schedule/speaker/6807

1) Druid Meetup Miami Florida
Date: 18 Jul 2008, 6pm-8pm EST, Redfone Communications Miami Florida
For more details and sign up http://druidmiami.eventbrite.com

2) OSCON 2008 (Portland Oregon)
Date: 21-25 Jul 2008, Oregon Convention Center, Booth 221

Navin Kumar, will be giving a talk on Druid
Building an Open Source Unified Communications Solution - The Druid Project
5:20pm - 6:05pm Thursday, 24 jul 2008
http://en.oreilly.com/oscon2008/public/schedule/speaker/27379
For more details and sign up, http://druidoscon.eventbrite.com

3)  Druid Meetup (West End London, UK)
Date: 22 Jul 2008, 6pm-8pm BET
Location: Thames Valley University - Room TC43
For more details and sign up http://druidlondon.eventbrite.com

4) LinuxWorld 2008 (Moscone Center, San Francisco CA, USA)
Date: 4-7 Aug 2008, 10am-4pm PST
Location: Moscone Center, San Francisco, Booth 1626
For more details and sign up http://druidlinuxworld.eventbrite.com

Some of the hot new stuff we will be demoing on Open Source at these events
- Druid Communicator on Blackberry v1.5 launched!
- Druid Communicator Adobe Air Application: SugarCRM/SalesForce Integration desktop application
- Cool stuff like Blackberry & Desktop Integration using Druid SOAP API
- Druid SOAP API: Your own Druid & Asterisk integration application in 10 mins for Druid

Regards,
Ming


--
Ming Yong
CEO, www.voiceroute.org
Druid - Open Source Unified Communications
DID: +1-877-242-3704
Office: +1-866-915-2407 ext 301
SIP/email: ming@voiceroute.net

Saturday, June 28, 2008

Re: [asterisk-biz] Free VoIP Predictive Dialer

Sam -

If you're needing voice blasting for the US only consider off loading your messaging to us and leverage our asterisk farm that is capable of thousands of calls a minute, complete messaging APIs (including API to record messages) and full reporting. I can get you a good price based on volume of calls.

Thanks.
Alan
www.group2call.com

--- On Thu, 6/26/08, Sam Tam <samtam888@gmail.com> wrote:
From: Sam Tam <samtam888@gmail.com>
Subject: [asterisk-biz] Free VoIP Predictive Dialer
To: "'Commercial and Business-Oriented Asterisk Discussion'" <asterisk-biz@lists.digium.com>
Date: Thursday, June 26, 2008, 10:41 PM

 

Hello

I am trying to after a Predictive dialer that can generate thousand of calls over a few VOIP lines. Ideally I don't need it to do much, just play a message and wait for a random of 2-3 mins then it hang up.

Does anybody know anything that does this?

Sam
 

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Re: [asterisk-biz] dear asterisk-biz@lists.digium.com june 80% 0ff

** EMAIL BLOCKED: Reason: tagged by SpamAssassin
**

Your Email To: asterisk-biz@lists.digium.com
Subject: [asterisk-biz] dear asterisk-biz@lists.digium.com june 80%
0ff

Your email has been rejected because our system has identified the
message as spam, unsolicited, or unwanted. If you believe this is
in error, please review the reason, listed above, and re-send your
message.

[asterisk-biz] Dear asterisk-biz@lists.digium.com June 80% 0FF

Click Here!

Friday, June 27, 2008

Re: [asterisk-biz] Free VoIP Predictive Dialer

Hi!

Please contact us for Hosted Predictive Dialer that fits your need, It can dial thousands of numbers depends on lines (you can define number of channels available for dialing) and can play pre-recorded voice message to either human Answered or it can leave message to answering machine.

TeleRep Performance Optimizer is not a free Predictive Dialer but it's cheap like Free Predictive Dialer.


Thank
Regards,
Zulqarnain
MSN: zulqarnain@globalitvision.com
email: zulqarnain@gitv.pk



Hello

I am trying to after a Predictive dialer that can generate thousand of calls over a few VOIP lines. Ideally I don't need it to do much, just play a message and wait for a random of 2-3 mins then it hang up.

Does anybody know anything that does this?

Sam
 

shady dial

http://voip-info.mirrors.bsd.net/wiki/view/ShadyDial.html

>
>
> Hello
>
> I am trying to after a Predictive dialer that can generate thousand of
calls over a few
> VOIP lines. Ideally I don´t need it to do much, just play a message and
wait for a random
> of 2-3 mins then it hang up.
>
> Does anybody know anything that does this?
>
> Sam
>



Rehan Ahmed AllahWala
Msn/Yahoo/GoogleTalk/Email: Rehan@Rehan.com
http://www.supertec.com/ - Internet Telephony Solutions
Http://www.DIDX.net - DID Number Market Place.
Don't Remember Me ? Visit http://www.Rehan.com

~~~~~~~~~~~~~~~~~~~
"First they ignore you, then they laugh at you, then they fight you, then
you win."
By Gandhi.

"Live as if you were to die tomorrow. Learn as if you were to live
forever." - Gandhi

Re: [asterisk-biz] Free VoIP Predictive Dialer

shady dial

http://voip-info.mirrors.bsd.net/wiki/view/ShadyDial.html

>
>
> Hello
>
> I am trying to after a Predictive dialer that can generate thousand of calls over a few
> VOIP lines. Ideally I don´t need it to do much, just play a message and wait for a random
> of 2-3 mins then it hang up.
>
> Does anybody know anything that does this?
>
> Sam
>

Rehan Ahmed AllahWala
Msn/Yahoo/GoogleTalk/Email: Rehan@Rehan.com
http://www.supertec.com/ - Internet Telephony Solutions
Http://www.DIDX.net - DID Number Market Place.
Don't Remember Me ? Visit http://www.Rehan.com

~~~~~~~~~~~~~~~~~~~
"First they ignore you, then they laugh at you, then they fight you, then you win."
By Gandhi.

"Live as if you were to die tomorrow. Learn as if you were to live forever." - Gandhi


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Thursday, June 26, 2008

[asterisk-biz] Free VoIP Predictive Dialer

 

Hello

I am trying to after a Predictive dialer that can generate thousand of calls over a few VOIP lines. Ideally I don’t need it to do much, just play a message and wait for a random of 2-3 mins then it hang up.

Does anybody know anything that does this?

Sam
 

Re: [asterisk-biz] Astricon: Early Bird Special ends next week

Thanks

It's interesting to see that Sangoma is a Platinum Sponsor for the
Digium Event.

http://www.astricon.net/2008/glendale/web/attendRegister.php


Am I to assume this is an indication of a new tolerance policy by Digium
of alternative vendor product(s)?

Cheers


Craig Lawrence


-----Original Message-----
From: asterisk-biz-bounces@lists.digium.com
[mailto:asterisk-biz-bounces@lists.digium.com] On Behalf Of John Todd
Sent: Friday, 27 June 2008 4:38 AM
To: asterisk-biz@lists.digium.com
Subject: [asterisk-biz] Astricon: Early Bird Special ends next week


Astricon 2008 is less than three months away - the Early Bird
discounts will expire on the last day of the month, which is next
Tuesday - please get your registrations in by then to get up to $100
off the normal rates. Making hotel reservations now is also a good
idea, since while there is a good supply of rooms at the conference
hotel, the supply is limited at the conference rate ($134/night.)

The 2008 Astricon is really shaping up to be the most tech-heavy
conference ever! We've received a great list of speakers, and there
are talks that really seem to be getting into the details of
Asterisk's new features, how to implement large-scale services, and
coverage of some great third-party applications in the Asterisk
ecosystem.

What makes Astricon great is the speakers, but also the participants.
The ability to talk in person with people who are working in the same
areas that you are, who have solved the same problems you're
encountering, or who you've met online but never face-to-face - these
are some of the most valuable parts of the conference. The informal
parts of the conference are where you make connections, figure out
code or implementation problems, or solve business issues that
otherwise would be difficult or impossible to handle outside of such
a concentrated group of similarly-minded people.

Here is a tiny random sample of the 50+ topics we've got in the agenda:

- A Carrier Grade VoIP Project with Asterisk.
- OpenR2 in Asterisk - MFC/R2 free of headaches or your money back
- Clustering Methods with Asterisk
- Druid: Open Source Unified Communications
- Asterisk Checks Into The Hotel
- ISDN PRI Capabilities and the Asterisk Implementation
- AT&T SIP Trunk Compatibility Testing for Asterisk
- Selling Asterisk-based Phone Systems In The Legacy World
- CEL: an introduction to Asterisk's new call logging mechanism
- Intro to Unified Communications: Two words, many challenges.
- Carrier Class Routing using Asterisk ExternalIVR and the Griffin
Routing Engine
- Asterisk, meet Lua: An Introduction to pbx_lua
- Measuring Signal Quality in Hybrid Systems (VOIP/PSTN)

Hope to see you in September!
- The Digium Astricon Staff

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[asterisk-biz] Astricon: Early Bird Special ends next week

Astricon 2008 is less than three months away - the Early Bird
discounts will expire on the last day of the month, which is next
Tuesday - please get your registrations in by then to get up to $100
off the normal rates. Making hotel reservations now is also a good
idea, since while there is a good supply of rooms at the conference
hotel, the supply is limited at the conference rate ($134/night.)

The 2008 Astricon is really shaping up to be the most tech-heavy
conference ever! We've received a great list of speakers, and there
are talks that really seem to be getting into the details of
Asterisk's new features, how to implement large-scale services, and
coverage of some great third-party applications in the Asterisk
ecosystem.

What makes Astricon great is the speakers, but also the participants.
The ability to talk in person with people who are working in the same
areas that you are, who have solved the same problems you're
encountering, or who you've met online but never face-to-face - these
are some of the most valuable parts of the conference. The informal
parts of the conference are where you make connections, figure out
code or implementation problems, or solve business issues that
otherwise would be difficult or impossible to handle outside of such
a concentrated group of similarly-minded people.

Here is a tiny random sample of the 50+ topics we've got in the agenda:

- A Carrier Grade VoIP Project with Asterisk.
- OpenR2 in Asterisk - MFC/R2 free of headaches or your money back
- Clustering Methods with Asterisk
- Druid: Open Source Unified Communications
- Asterisk Checks Into The Hotel
- ISDN PRI Capabilities and the Asterisk Implementation
- AT&T SIP Trunk Compatibility Testing for Asterisk
- Selling Asterisk-based Phone Systems In The Legacy World
- CEL: an introduction to Asterisk's new call logging mechanism
- Intro to Unified Communications: Two words, many challenges.
- Carrier Class Routing using Asterisk ExternalIVR and the Griffin
Routing Engine
- Asterisk, meet Lua: An Introduction to pbx_lua
- Measuring Signal Quality in Hybrid Systems (VOIP/PSTN)

Hope to see you in September!
- The Digium Astricon Staff

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Re: [asterisk-biz] Looking for IAX2 A-Z providers



2008/5/12 Vahan Yerkanian <vahan@arminco.com>:
Greetings all,

We are looking for A-Z IAX2 European providers with capabilities. Please
contact me off-list if you are interested.

I'd certainly recommend Magrathea Telecom in the UK.


Steve

Wednesday, June 25, 2008

Re: [asterisk-biz] Opinions requested: message blasting

Hi!

Recently in Pakistan we have Launched and tested similar Voice Broadcasting Service that were heavily tested for sending millions of pre-recorded voice messages to Pakistani People for getting their feedback on Election 2008 Campaign ending with the option to press 1 to gave their feedback about favorite Political party in Election Campaign.

Our hosted service is so cheaper then any similar service provider in the world so we called it as the "World Cheapest Voice Broadcasting Solution" because the pricing are as low as $0.04c/minute for answered message. The most valuable feature is that there is no restriction to use it for any Geographical region you are free to use for calling your client anywhere in the world with your own SIP Carrier/Trunk (administration through web interface).

Thanks
Regards

Muhammad Zulqarnain
email: zulqarnain@gitv.pk
http://www.gitv.pk
Msn: zulqarnain@globalitvision.com




Re: [asterisk-biz] New faxing protocol. Good/Bad ?

The IEEE's Printer Working Group (PWG) was working on such a protocol.

Google for IEEE PWG IFX.

It was to use a (bitmap) PDF subset as the image format, and TCP as the
transport. (I can't recall whether SCTP was considered as an alternative.)

AFAICT it failed due to a disconnect between two proponents: one group
wanted to use it over the 'net and the other for communicating over the
LAN with multi-function devices. The former wanted open inward calling
whereas the latter wanted to lock down the devices.

(I could be wrong about it failing, but that was the impression I got
from the mailing list.)

If you do want to do something different, use connection oriented
sockets (TCP or SCTP) rather than datagram sockets (UDP or DCCP) and
use PDF for the image format.

-JimC
--
James Cloos <cloos@jhcloos.com> OpenPGP: 1024D/ED7DAEA6

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[asterisk-biz] Asterisk Gigs in CT

I have a colleague that is looking to hire for two high level Asterisk engineer positions.

 

-Senior Telephony Engineer

-Asterisk Systems Architect

 

Firm is located in Connecticut, and ideally they are looking for people who can relocate.  If interested, shoot me an email and I will pass you their contact info and more detail regarding the positions.

 

Regards,

 

Cory J Andrews

Director, New Market Initiatives

 

Sayers Media Group

 

454 Sonwil Drive

Buffalo, NY 14225

716-250-3402 OFFICE

716-630-1548 FAX

716-601-4474 MOBILE

Candrews@SayersMedia.com

 

NOTICE: The information contained in this email and any document attached hereto is intended only for the named recipient(s). It is the property of the VoIP Supply, LLC and shall not be used, disclosed or reproduced without the express written consent of VoIP Supply, LLC. If you are not the intended recipient, nor the employee or agent responsible for delivering this message in confidence to the intended recipient(s), you are hereby notified that you have received this transmittal in error, and any review, dissemination, distribution or copying of this transmittal or its attachments is strictly prohibited. If you have received this transmittal and/or attachments in error, please notify me immediately by reply e-mail or telephone and then delete this message, including any attachments. Our mailing address is 454 Sonwil Drive, Buffalo, NY 14225 USA.

 

Re: [asterisk-biz] Opinions requested: message blasting

On Wed, 2008-06-25 at 08:29 -0400, Dean Collins wrote:
> I got a robo-call from Duane Reade (pharmacy chain in the USA), saying
> that one of my scripts was up for renewal that week and provided the
> nearest address for pickup.
>
>
>
> I was chuffed enough about it to write a blog pos a few months back.
>
> http://deancollinsblog.blogspot.com/2008/05/duane-reade-robo-call.html

>
> I think it was brilliant.
>
>

That isnt something I knew about when I was commenting on the online
pharmacy, but yes generally I think that is a good idea, particularly
since people are busy, some medications or conditions cause people to
forget things and often lapses in medication can be dangerous.

Scripts.com iirc is the online one that uses the same thing to say "its
on its way" I think they also do it based on return receipt stuff, which
if you do enough you can probably get online notices from usps.com. I
got one of their calls, which really is just a recording.

The ones I liked more were the appointment ones that first make you
press some numbers indicating that you are the intended receiptient.
Then they will tell you that your appt is tomorrow, and you have to
press something to acknowledge it, if you dont it gets canceled. This
coincides with the 24 hour cancelation policy as well, since its the day
before usually if you dont acknowledge the appointment you arent billed
for a missed one.

A huge percent of americans are over the age of 50 now. Many will be
taking medications, seeing doctors, and forgetting to get refills or
show up for appointments. By implementing these types of systems it
helps a lot of people. Even by reducing the number of people required
in the office to see the same number of patients that means that they
arent as busy and can acutally focus a little more on patient care and
not making phone calls to inform someone that they didnt show up at all
to the office as scheduled.

Really this service is just a "hotel wake up call" on steroids, another
use for the same software :)


>
--
Trixter http://www.0xdecafbad.com

Bret McDanel
Belfast +44 28 9099 6461 US +1 516 687 5200
http://www.trxtel.com the phone company that pays you!


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Re: [asterisk-biz] Opinions requested: message blasting

I got a robo-call from Duane Reade (pharmacy chain in the USA), saying that one of my scripts was up for renewal that week and provided the nearest address for pickup.

 

I was chuffed enough about it to write a blog pos a few months back.

http://deancollinsblog.blogspot.com/2008/05/duane-reade-robo-call.html

I think it was brilliant.

 

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

 

 

-----Original Message-----
From: asterisk-biz-bounces@lists.digium.com [mailto:asterisk-biz-bounces@lists.digium.com] On Behalf Of Trixter aka Bret McDanel
Sent: Wednesday, 25 June 2008 8:12 AM
To: Commercial and Business-Oriented Asterisk Discussion
Subject: Re: [asterisk-biz] Opinions requested: message blasting

 

Doctors offices regularly use them to remind people of their

appointments to reduce missed ones.  A major US based online pharmacy

(yeah US based!) also does it to remind people their medication is on

its way so they know to expect it and report it missing if it does not

arrive. 

 

--

Trixter http://www.0xdecafbad.com     Bret McDanel

Belfast +44 28 9099 6461        US +1 516 687 5200

http://www.trxtel.com the phone company that pays you!

 

 

_

Re: [asterisk-biz] VoIP Provider

On Tue, 2008-06-24 at 23:09 -0700, John van Oppen wrote:
> I should probably also point out that the public internet links also
> tend to be far larger than the private interconnections and thus
> infinitely easier to scale than needing to continually flip production
> voice traffic onto ever larger private circuits.
>

That may be however if its a true private link with a bandwidth
guarantee contention rates arent the same, and DDoS attacks and worms on
the net at large shouldnt affect their performance. I am not talking
about just a VPN but a real circuit with real bandwidth guarantees.

If you get dark fiber for your private links then you can upgrade the
speed anytime you want by replacing the end equipment. This allows you
to scale without contention. Many ISPs have such a high contention
rate, especially the closer you get to being an end user, and why some
are now traffic shaping to control the bandwidth consumption issues.
They have oversold and people actually wanted to use what they bought.
This happens even on the internet at large. Especially with specific
peering points.


> In this case, the network to which Miles is referring, has gigE to five
> providers and one public peering point and thus has sub 1ms latency to
> the networks of many of the major origination and termination players
> (via multiple paths) as such could easily be construed as better
> connected than a private link.
>

Ahh but "better connected" is a qualified statement, better connected to
whom? A private link with guarnateed bandwidth may be better connected
to a POP that gets regional DIDs and forwards them off but not as well
connected to say average joe user.


--
Trixter http://www.0xdecafbad.com

Bret McDanel
Belfast +44 28 9099 6461 US +1 516 687 5200
http://www.trxtel.com the phone company that pays you!


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Re: [asterisk-biz] Opinions requested: message blasting

On Tue, 2008-06-24 at 20:51 -0400, Cory Andrews wrote:
> Chris - I think it depends on the usage scenario....here in the states,
> automated outbound messaging is a popular tactic amongst political
> candidates to deliver campaign statements and GOTV messaging to
> registered voters.
>
> In the mid 90's it was used as a telemarketing lead gen tool, but has
> decreased in popularity due to DNC regulations in the states.
>
not DNC, but telemarketing in general, in the US its illegal to play a
recording for anything but religious and political messages. So if they
want to generate leads or do sales they are only allowed to (and infact
required to) give very brief information that cannot include a sales
pitch (doesnt mean people dont do it, just means that they can be
reported and fined if they do).


> It's also a great tool for "value added" services like appointment
> reminders, school related notifications, group related notifications,
> service reminders, etc. As a lead generation tool IMO its effectiveness
> is limited, and it is generally regarded in a similar manner as spam.
>

Doctors offices regularly use them to remind people of their
appointments to reduce missed ones. A major US based online pharmacy
(yeah US based!) also does it to remind people their medication is on
its way so they know to expect it and report it missing if it does not
arrive.

In 1999-2000 I worked for a company that was selling systems to airlines
(and anyone else that would buy the entire system) that would let
consumers choose how to be notified of their travel plans, letting them
choose pager, fax, sms, voice, email, etc.


If people thought of message blasting like email lists they probably
wouldnt be so upset over it as a whole and rather only be upset over it
when its unsolicited and probably sales related. Instead the average
person generally thinks of it as just phone spam because that is all
they think about and forget if they get such a call reminding them of a
real appointment or some other service.


--
Trixter http://www.0xdecafbad.com

Bret McDanel
Belfast +44 28 9099 6461 US +1 516 687 5200
http://www.trxtel.com the phone company that pays you!


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Re: [asterisk-biz] Opinions requested: message blasting

On Wed, 2008-06-25 at 01:32 +0100, Chris Bagnall wrote:
> Greetings list,
>
> I've seen a lot of posts over the last couple of years on both -users and -biz with people requesting help with "message blasting" setups - that is to say, asterisk dials a number, waits for an answer, then plays a pre-recorded message.
>
> Everyone I've ever spoken to about this here in the UK finds it thoroughly offensive. If they answer the phone, they expect to find a human being at the other end. Systems like this seem to draw far more ire from people than even multi-level IVR systems (which aren't exactly popular either).
>
well systems like that are good for some situations, did you ask them
the following:
A volunteer fire department in a rural area uses an automated system to
call firemen to an emergency, do you have a problem with that?

A school calls parents to let them know of a closure while school
officials deal with inbound calls from parents. Do you have a problem
with that?

Or did you merely ask someone calls you unsolicited and plays a
recording (illegal to do in the US at least, probably UK too if its for
sales/marketing stuff) if that would be anoying?

How you ask the question can greatly influence the response. Why polls
should declare the wording of the questions used and not just the
responses.

--
Trixter http://www.0xdecafbad.com

Bret McDanel
Belfast +44 28 9099 6461 US +1 516 687 5200
http://www.trxtel.com the phone company that pays you!


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Tuesday, June 24, 2008

Re: [asterisk-biz] VoIP Provider

I should probably also point out that the public internet links also
tend to be far larger than the private interconnections and thus
infinitely easier to scale than needing to continually flip production
voice traffic onto ever larger private circuits.

In this case, the network to which Miles is referring, has gigE to five
providers and one public peering point and thus has sub 1ms latency to
the networks of many of the major origination and termination players
(via multiple paths) as such could easily be construed as better
connected than a private link.

What does astonish me is what people will do with their production voip
traffic in terms of running it across either small links with no QOS or
providers known for peering issues. I can tell you I would never run
production voice over a consumer circuit like verizon Fios or random T1
link and expect flawless results. I use voip in production at my home
office and even there I have provider-side QOS enabled on the link to my
house, without control of the link the connection is coming in on there
is no guarantee of a congestion-free (or QOS enabled) path which is
required for great voice quality.

There also appears to be a bit of a misconception of the "downsides" of
running this kind of traffic over the public internet. As miles noted
originally, the congestion very rarely happens at the core, it happens
at the edges. This is not just a "guess" this is fact, I have tracked
latency towards big destinations from our network for quite some time
and I can tell you that the peak-to-lull latency change for most large
destinations is around 150-200 microseconds (yes, 0.15 to 0.2 ms) if you
are doing the tests from a well connected place (the above described
network in this case) whereas many consumer-type connections have large
latency variations depending on load. The only time I have ever seen
congestion in the core was across peers between large tier1 providers
and in the last three or four years I have only seen that on peers to
AS7018 (ATT) or AS174 (Cogent) which at least in our case accounts for
virtually none of our voice traffic.

Anyway, that is it for my contribution. As miles notes, we offer
wholesale origination services using our own IP backbone with SIP
proxies located in the Westin Building in Seattle, WA. Feel free to
try traceroutes or speed tests.

http://as11404.net or
http://as11404.net/speedtest (be nice to www.as11404.net as it is only
on 100 mbit/sec Ethernet).


It is getting late, so I hope all of that made sense.

John van Oppen
Spectrum Networks LLC

-----Original Message-----
From: asterisk-biz-bounces@lists.digium.com
[mailto:asterisk-biz-bounces@lists.digium.com] On Behalf Of Miles
Scruggs
Sent: Tuesday, June 24, 2008 10:31 AM
To: Commercial and Business-Oriented Asterisk Discussion
Subject: Re: [asterisk-biz] VoIP Provider

You can't really change the way a provider offers their circuits, no
matter how much you want to speak a certain way for personal
preferences. They offered it over T1s or DS3s, you can try hammering
out semantics with them, but I doubt it will help.

On Jun 24, 2008, at 10:14 AM, Steve Totaro wrote:

> It would seem if someone is speaking of T1s then they should also use
> T3, if they are speaking of DS1 then DS3 would be the proper naming
> convention.


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