[Asterisk-Users] T1 E&M vs PRI question

[Keith Burns]

Depending on the switch they are using, there are a limited number of
D-channels (or D-channel licenses).
 
CAS signaling needs RBS (it's the winking in this case).
 
 
-----Original Message-----
From: asterisk-users-bounces at lists.digium.com
[mailto:asterisk-users-bounces at lists.digium.com] On Behalf Of Matt Beebe
Sent: Monday, January 24, 2005 2:47 PM
To: asterisk-users at lists.digium.com
Subject: [Asterisk-Users] T1 E&M vs PRI question
 
Ok,
 
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:
 
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.
 
Sounds like the way to go, but basically the PRI ends up being
$100/month more expensive than the Channelized T1 E&M.
 
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??) 
 
I believe in either case, the normal call features (3-way, forwarding,
etc) can be provisioned.
 
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?
 
Thanks in advance!
 
-Matt
 
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