[asterisk-biz] PRI confusion

[Igor H]

Thanks a bunch Ming, your help is really appreciated!

Take care,

Igor H.

On Nov 24, 2007 1:32 AM, Ming Yong <ming at voiceroute.net> wrote:
> For least bandwidth, get quality PRI card eg. Sangoma with echo cancellation
> + onboard DSP encode the audio in and send out using g729 (10kbps per
> channel) SIP.
>
> For best quality, take PRI channels as ulaw & alaw out using 23 x 32kbps =
> 736kbps or 1Mbps will be good enough
>
> Ming
>
>
>
> On Nov 24, 2007 2:26 PM, Igor H <emistz at gmail.com> wrote:
> > 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 =\
> > > > > >
> > > > > > _______________________________________________
> > > > > > --Bandwidth and Colocation Provided by
> http://www.api-digital.com--
> > > > > >
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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
> >
>
>
>
> --
>
>
> Ming Yong
> CEO, www.voiceroute.net
> DID: +1-650-331-1732 ext 301
> SIP/email: ming at voiceroute.net