<!DOCTYPE HTML PUBLIC "-//W3C//DTD HTML 4.0 Transitional//EN"><HTML DIR=ltr><HEAD><META HTTP-EQUIV="Content-Type" CONTENT="text/html; charset=iso-8859-1"></HEAD><BODY><DIV><FONT face='Arial' color=#000000 size=2>Ok,</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>I'm about to take the plunge, and am trying to
decide between Channelized T1 E&M and PRI. I'm getting an "Integrated
T1" which will have data and voice capability, all plugged directly into my
digium single T1 card. In either case the data piece looks pretty
straighforward, just setup the channel properly, hand it off to the linux hdlc
layer, and route away.... the voice side seems a little more complex -- I'm
looking for clarification and/or advice:</FONT></DIV>
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<DIV><FONT face=Arial size=2>It seems to me that the major differences between
the two different voice delivery mechanisms (other than cost) is caller id
functionality and call setup delay. With the PRI, I'll have practically
instant call setup and the ability to pass CNAM (caller name) and CID (caller
ID) information in BOTH directions. The PRI will give me
the ability to have additional directory numbers (typically called DIDs)
assigned against my voice trunks and will provide the full ANI (automatic
number identification) and DNIS (dialed number identificaton service) over
the PRI signalling trunk. Each voice channel will also be 64k clear
channel, so I could (theoretically) provide 56k dial-in modem service from the
same box (anyone actually doing this?? seems like a neat application for the dsp
software guys) I also lose one 64k channel to signalling.</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>Sounds like the way to go, but basically the PRI
ends up being $100/month more expensive than the Channelized T1
E&M.</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>The T1 E&M approach will still give me CID (but
not CNAM???) over the in-band call setup mechanism (ie: quick DTMF tones during
the wink). Each voice channel will actually be 56k because it uses RBS
(robbed bit signalling -- not sure what its using this for, as the call setup is
delivered via wink???). As a result, this approach would also keep me from
implementing a 56k dial-in modem service, but I could still use an
"ordinary" modem or fax dsp to provide 33.6k dial-in. This setup
can support DID, but its appended (or prepended, depending on the provider) to
the DTMF call setup (which extends the time for calls to actually
connect). Not sure if CID or CNAM can be provided for outgoing
calls (I think some providers can enable me to be able to wink to them
the number to pass as caller id??) </FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>I believe in either case, the normal call features
(3-way, forwarding, etc) can be provisioned.</FONT></DIV>
<DIV> </DIV>
<DIV><FONT face=Arial size=2>Do I have it about right?? Is it pretty
normal for providers to charge a premium for the PRI? Any
thoughts/clarifications to my above assumptions?? Are there other
pros/cons of each setup?</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>Thanks in advance!</FONT></DIV>
<DIV><FONT face=Arial size=2></FONT> </DIV>
<DIV><FONT face=Arial size=2>-Matt</FONT></DIV>
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