[asterisk-biz] PRI confusion

Igor H emistz at gmail.com
Sat Nov 24 00:26:37 CST 2007


Thanks again Ming,

I see what you mean, what kind of connection do you think would be
optimal to handle the outgoing route of the calls using say g.711
codec?

On Nov 24, 2007 1:13 AM, Ming Yong <ming at voiceroute.net> wrote:
> Igor,
> Honestly, I would do 23 channels incoming DID with pure SIP VOIP trunking
> via internet WAN to my terminating VOIP provider.
> I do not quite understand why you need another 11 channel for outgoing voip
> traffic when simple Ethernet SIP outgoing connections will do.
> Ming
>
>
>
> On Nov 24, 2007 2:10 PM, Igor H <emistz at gmail.com> wrote:
> > Thanks Ming,
> >
> > Another question is would it be possible to handle 23 simultenous
> > calls through the PRI or would I need 11 pri channels for the incoming
> > calls and 11 pri channels for the outgoing voip traffic?
> >
> >
> >
> >
> >
> > On Nov 24, 2007 1:05 AM, Ming Yong <ming at voiceroute.net> wrote:
> > > This is Ming from Voiceroute.
> > > There is a very simple way to do this.
> > > PRI will send 11 callerIDs into the PRI card with 1 context. This
> context
> > > can be an inbound route context that will route 11 different DIDs into
> > > auto-attendant or DISA that will route to an outbound dialing pattern to
> the
> > > trunk you want.
> > >
> > > Let me know if you need more help.
> > > Btw, the above can be done using Druid very easily. Many VOIP providers
> have
> > > implemented the above using Druid Enterprise communications server ECS.
> > > Features & Benefits
> > > http://www.voiceroute.net/site/druidecs/features
> > > Eg of VOIP provider customers
> > > http://www.voiceroute.net/site/druidecs/customers
> > >
> > > We have free trials for the software.
> > >
> > > Ming
> > >
> > >
> > >
> > > On Nov 24, 2007 1:41 PM, 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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> > >
> > >
> > >
> > > --
> > > Ming Yong
> > > CEO, www.voiceroute.net
> > > DID: +1-650-331-1732 ext 301
> > > SIP/email: ming at voiceroute.net
> >
>
>
>
> --
>
>
> Ming Yong
> CEO, www.voiceroute.net
> DID: +1-650-331-1732 ext 301
> SIP/email: ming at voiceroute.net



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