[Asterisk-Users] VoIP dialtone?
Mark Spencer
markster at digium.com
Thu Aug 21 21:42:00 MST 2003
As far as people are thinking of sharing their telephony, we could let
people start exposing them through iaxtel. If anyone has areacodes &
prefixes they want to make available, e-mail me and I can set you up on
iaxtel.
Mark
On Thu, 21 Aug 2003, Brian West wrote:
> NUFONE R0X!
>
> Took him 30 seconds or so to set me up when I got services with him! :P
>
> bkw
>
> On Thu, 21 Aug 2003, Jeremy McNamara wrote:
>
> > Our phones have been working perfectly fine all day. I've personally
> > supported quite a few new users over the phone today and even set a
> > couple up.
> >
> >
> > Jeremy McNamara
> >
> >
> >
> >
> > Steve Lane wrote:
> >
> > >Nufone won't answer their phones. I am very interested in finding out
> > >pricing from them as Jeremy stated they are very good with their rates.
> > >
> > >Steve
> > >
> > >-----Original Message-----
> > >From: asterisk-users-admin at lists.digium.com
> > >[mailto:asterisk-users-admin at lists.digium.com] On Behalf Of Adam Roach
> > >Sent: Thursday, August 21, 2003 10:23 AM
> > >To: 'asterisk-users at lists.digium.com'
> > >Subject: RE: [Asterisk-Users] VoIP dialtone?
> > >
> > >I think Jeff Pulver (pulver.com) was trying to do this
> > >with his Free World Dialup program at one point. Haven't
> > >been paying that much attention, though. You might
> > >poke around http://www.pulver.com/ to see if there's
> > >something there that interests you.
> > >
> > >/a
> > >
> > >
> > >
> > >>-----Original Message-----
> > >>From: Dan Austin [mailto:Dan_Austin at Phoenix.com]
> > >>Sent: Wednesday, August 20, 2003 22:07
> > >>To: asterisk-users at lists.digium.com
> > >>Subject: RE: [Asterisk-Users] VoIP dialtone?
> > >>
> > >>
> > >>This idea has been floating around in my head. I don't think the
> > >>needed 'critical mass' has been reached, but I suspect at some
> > >>point a co-op style arrangement could be reached.
> > >>
> > >>disclaimer:
> > >> I have played with *, and am deploying Cisco Call Manager.
> > >>I don't see any technical reason why the following would not work,
> > >>but it is open for abuse, so there may be enough socio-political
> > >>reasons to not even try.
> > >>
> > >>Ingredients:
> > >> 1. A * server
> > >> 2. A friend with an * server in another city/state/country
> > >> 3. A way to locate like minded individuals/orginizations
> > >> 4. Moderately over-built local PSTN connectivity
> > >>
> > >> Mix it together with a gentlemans agreement, or strongly
> > >>worded contract. Co-ordinate or advertise local number ranges.
> > >>
> > >>
> > >>Problems:
> > >> People looking to save ~$30 per line won't be thrilled to
> > >>order T1(s) to share with the co-op.
> > >> Keeping a structured dial-plan to provide for reasonable
> > >>overlap without massive meltdowns.
> > >> There are many businesses springing up to fill this void,
> > >>and they will be better suited to manage and grow the infrastructure.
> > >>
> > >>I've watched the discussions about IAX/SIP service providers, and
> > >>most seem to be geared exclusively to the single user/line household.
> > >>I know a number of small businesses that would jump to a VoIP carrier
> > >>that allowed concurrent calls, heck my family has one. And I suspect
> > >>that a number of the smaller/newer VoIP carriers might be entertaining
> > >>partnerships with their competitors whose footprint compliments their
> > >>own.
> > >>
> > >>Oh, and let's not forget that the traditional carriers are
> > >>not ignorant
> > >>of what is happening with VoIP or customer interest. There
> > >>is no doubt
> > >>that they are aware that if they don't find a way to deliver
> > >>this service,
> > >>someone else will.
> > >>
> > >>Dan (who, if he had a decent PSTN connected * box, would be
> > >>willing to share)
> > >>
> > >>
> > >>
> > >>-----Original Message-----
> > >>From: Mike Ciholas [mailto:mikec at ciholas.com]
> > >>Sent: Wednesday, August 20, 2003 3:42 PM
> > >>To: Ernest W. Lessenger
> > >>Cc: asterisk-users at lists.digium.com
> > >>Subject: RE: [Asterisk-Users] VoIP dialtone?
> > >>
> > >>
> > >>
> > >>On Wed, 20 Aug 2003, Ernest W. Lessenger wrote:
> > >>
> > >>
> > >>
> > >>>At 04:48 PM 8/20/2003 -0500, you wrote:
> > >>>
> > >>>
> > >>>
> > >>>>Now, if that is possible, how does the VoIP dial tone provider
> > >>>>get my inbound local and toll calls? I would want my "local"
> > >>>>phone number to work, of course.
> > >>>>
> > >>>>
> > >>>You would need to redirect your local number to them. This
> > >>>ALWAYS assumes that the VoIP provider has a switch in your
> > >>>local CO or an agreement with someone who does. Vonage and
> > >>>Voicepulse, for example, do not have a presence in my area. I
> > >>>intend to maintain several POTS lines for incoming calls, and
> > >>>use a VoIP provider for all outgoing calls.
> > >>>
> > >>>
> > >>Oh well. I'm would expect no one would have presence here.
> > >>This sounds so suboptimal, you have to provision *two* systems,
> > >>one for inbound (local CO) and one for outbound (VoIP provider).
> > >>Of course, the outbound can be just your internet connection, but
> > >>this still seems annoying because most of the money is in the
> > >>local CO service.
> > >>
> > >>Hmm, perhaps *all* incoming calls can be toll free? I would
> > >>maintain the one local CO POTS line for 911 out bound, and then
> > >>only use my toll free number for inbound. For the money I would
> > >>save on local CO lines I can buy a *lot* of toll free minutes!
> > >>Then the VoIP dial tone provider can route my toll free number to
> > >>me over the internet. Presumably, then, there is no real limit
> > >>on the number of "lines" coming in. It isn't hard coded like the
> > >>CO lines are.
> > >>
> > >>This all seems pretty fanciful at the moment...
> > >>
> > >>--
> > >>Mike Ciholas (812) 476-2721 voice
> > >>CIHOLAS Enterprises (812) 476-2881 fax
> > >>2626 Kotter Ave, Unit D mikec at ciholas.com
> > >>Evansville, IN 47715 http://www.ciholas.com
> > >>
> > >>
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> > >>Asterisk-Users at lists.digium.com
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> > >>
> > >>
> > >>
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