OK, i installed trixbox (centos/asterisk/freepbx) from an .iso into a kvm
under ubuntu.  No networking yet...

This is a good excuse to learn about vm's...


On Sat, Feb 20, 2010 at 9:38 PM, Andre Courchesne
<[email protected]>wrote:

> For an easy and clean installation, simply install CentOS, then configure
> the digium repository and install asterisk and freepbx via yum:
>
> http://www.asterisk.org/downloads/yum
>
> ---
>
> Andre Courchesne - Consultant
>
> http://www.net-forces.com
>
> MSN: [email protected]
>
> Skype: VoipForces
>
>
>
> L'information contenue dans le présent document est la propriété de Andre
> Courchesne. Et est divulguée en toute confidentialité. Cette information ne
> doit pas être utilisée, divulguée à d'autres personnes ou reproduite sans le
> consentement écrit explicite de Andre Courchesne.
>
> The information contained in this document is confidential and property of
> Andre Courchesne. It shall not be used, disclosed to others or reproduced
> without the express written consent of Andre Courchesne.
>
>
>
>
>
> On 2010-02-20, at 4:48 PM, Peter Silva wrote:
>
> > So I have a spa2102 (SIP -> 2 FXS ports) that talks to the plain old
> telephones in my house.  Line 2 is set to use a voip provider (callcentric,
> as recommended by mlug!) so far so good.  What I would really like to do is
> have some sort of routing where a little server does voicemail and forwards
> it to my email account.  I'd also like laptop softphone and home phone both
> ring when someone phones home, and be able to have glorious voice menus to
> allow per person extensions and distinctive rings, and ... I dunno what, but
> it'll be fun.
> >
> > My router/firewall is debian, and apt-get install asterisk worked, but
> the configuration looks really complicated. It doesn't seem to have any
> getting started type documentation, just sort of reference material.  Since
> I have the SPA box, I don't actually need any POTS/PSTN function.  without
> any PSTN to do, do I even need asterisk?  Is there pure SIP-based switch
> that would be easier / faster / simpler ?
> > I have the ekiga softphone on my laptop, and an asterix prompt...
> >
> > I guess the first thing I'd like to do is create an account on asterix
> for the ekiga phone, and another
> > one for "line2" on the SPA, and phone one from the other... (leaving
> line1, the real phone, alone for now.)
> > Do I just mess with sip.conf for that?  decent "getting started" type
> guides?
> >
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> >
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