[asterisk-biz] PRI confusion
Igor H
emistz at gmail.com
Sat Nov 24 15:00:55 CST 2007
Pure sip meaning I would buy some DID's for the customers to call in
and have it routed to my asterisk box via sip?
How reliable are those usually?
On Nov 24, 2007 1:46 PM, Moshe Maeir <moshe at maeir.com> wrote:
> Hi,
> Why do you want to go the PRI route?
> Why not pure SIP? That is the way we do it. Pretty easy to set up, no
> initial investments in h/w and you
> are not limited at all by the number of channels.
> If you are interested, we can offer you your own partition on our server.
>
> Good Luck
> Moshe Maeir
> The Flat Planet Phone Co.
>
>
> On Nov 24, 2007 7:41 AM, emist <emistz at gmail.com> wrote:
> >
> >
> >
> > Hey guys,
> >
> > I was hoping someone could clarify this for me since i've been trying to
> > find out for a while now to no avail. Im thinking of deploying a call
> > routing service through asterisk. Basically I want people to be able to
> > call a number through the PSTN and then call whatever extension to be
> > routed through a voip termination provider.
> >
> > Im guessing using a PRI is the best way to do this. However, im confused
> > as to how it all works. Say a PRI has 23 usable channels, does that mean
> > that I will be able to route 23 calls at the same time or does it mean
> > that I would have to split 11 channels for incoming voice traffic(from
> > PSTN) and 11 channels for outgoing voip traffic?
> >
> > Im stomped =\
> >
> >
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