Well,  for calling party ID and message waiting light stuff on centralized
voice mail, Lucent canuse it's own form of ISDN PRI called DCS+ .  The way
they usually set it up for PBX to PBX connectivity is the customer purchases
a clear channel T1 and Lucent runs its own PRI over it with one side being
the network protocol side (isdn switch) and the other being the user.  They
use channel 24 for signaling and sometimes channel 23 for centralized call
accounting.  The rest of the time slots are bearer channels.  Cisco can
emulate this with a technique of VOIP called Transparent Common Channel
Signaling (TCCS)

However, I am almost sure this won't work between different PBX venders
because DCS+ is proprietary.

Another approach is QSIG.  This is a semi standard type of ISDN PRI that is
supposed to signal all the stuff you need like calling party display,
Message waiting light display, etc.

The way you set it up is to set up the controller for the E1 interface for
the proper linecode,clocksource, and framing you want to use.  Then the
command "PRI-timeslots 1-30.

This creates a serial interface that looks like "interface serial  "mod/port
: 16"   I can't remember if it's timeslot 15 or 16 on E1 that is the D
channel.  (I'm a t1 kind of guy)  Under this port you call out the "ISDN
switch-type QSIG"  A cisco router is supposed to emulate the network
protocol side of a PRI but I haven't had much luck with it.  Just let each
PBX be network side.

Then it just a matter of dial peers for your dial plan.  There is lots of
info on CCO on how to configure that.

The QSIG switch type is supposed to be vender independent and is supposed to
pass along all the information you need

You replicate a similar config on the other side obviously.  Don't forget
QOS techniques for whatever layer 2 wan transport your using

The G729 codec (if that is the one you want to choose for your compression)
will give toll quality voice and real low bit rate.

I hope this gives you an start on what has to happen.  There is a TON of
stuff on CCO on toll bypass setups.  I prefer VOIP because of the
flexability.

Good luck
Tony M.
#6172
----- Original Message -----
From: 
To: 
Sent: Wednesday, May 23, 2001 7:11 AM
Subject: RE: connecting NM-HDV-1E1-30E to Alcatel 4300 PABX [7:5566]


>
> Hi Tony.
> Let me explain my case.
>
> Customer has 2 different sites.
> One site has Lucent PABX the other has Alcatel 4200 and 4300 PABX.
> In Both sides Cisco 3661 and 1 Mbps Frame Relay connection between them.
>
> They want to reduce their phone costs. So they decide to carry voice
traffic
> on data network.
> VoIP or VoFR ?  We havent decided yet. Probably VoIP.
>
> NM-HDV-1E1-30E modules on 3661 routers will be used when integrating PABX
to
> data network.
>
> They dont plan to use IP phones.
> Calling party display is needed actually.
>
> Thanks for your help.
> Best Regards.
>
> Burcin
>
>
>
> -----Original Message-----
> From: Tony Medeiros [mailto:[EMAIL PROTECTED]]
> Sent: Thursday, May 24, 2001 8:02 AM
> To: [EMAIL PROTECTED]; [EMAIL PROTECTED]
> Subject: Re: connecting NM-HDV-1E1-30E to Alcatel 4300 PABX [7:5566]
>
>
> I have never done an Alcatel, but I here the same things apply, just
> different signaling techiques apply. Are you emulating a PBX to PBX PRI
> trunk?  Do you have centralized email and need MWL fuctioniality? Need
> calling party name display ?Are you useing the router to tie into a Cisco
IP
> telephony solution?  Using voice over ip or frame?
>
> All these uses require different techniques.  If you give  little more
info
> I'll let you know what I think needs to be done.
>
> Tony M.
> #6172




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