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

[David Boyd]

Responses embedded below!

On Mon, 2005-01-24 at 18:49, Keith Burns wrote:
> 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:
> 
> 

PLease no Flame, just a correction if required.

There seemed to be issue using Data/Voice on the digium cards, but I
believe it is a setup issue not a technical limitation on the card
itself.


>  
> 
> 
> 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.
> 
Actually DNIS can be provisioned over e&m trunking also, the separation
of digits is done with *'s or KP/ST. So the digiti dump would be
something like:
DTMF
OH ->

	<- Wink 

		digit dump *703727131229*8004231212*->
							<-wink
								<-Answer		

The breakdown of the digits is ani + Info digits then DNIS

The *'s would be replaced with KP/ST pulses if MF.  KP start sequence, &
ST stop sequence.

Sorry for the crude drawing, and the disclaimer is its been 4 years
since I have looked at the digit sequence for an E&M t1 :)

> 	
>  
> 
> 
> 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 don't know of a way for outbound or inbound CNAM to be provided on a
T1 unless you are using SS7 or some like control protocol. 

The setup time is in milliseconds for PRI and potentially could be 1.2
seconds in E&M including wink times, and outpulse dump. This can be
decreased if the carrier can accept fast outpulse, and also be decreased
if you use MF with KP & ST pulses instead of DTMF.

Robbed bit allows for the current channel condition to be maintained in
the signalling stream. When a channel hangs up the onhook condition has
to be able to be passed to the other end of the t1 for disconnect.  The
wink and digits dump at the start of the call only provides call setup
capability.

> 
>  
> 
> 
> I believe in either case, the normal call features (3-way, forwarding,
> etc) can be provisioned.
> 
> 
Additional features are usually  handled within the switching/* system
once the call has been setup. There are some features that are available
via ISDN, however in my experiences most carriers don't/won't support
them.

>  
> 
> 
> 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?
> 
> 
Yes it is normal for increased cost, however IMHO I would spend the
additional money (assuming one can afford it) for improved throughput
and performance. 

>  
> 
> 
> Thanks in advance!
> 
> 
>  
> 
> 
> -Matt
> 
> 
>  
> 
> 
> 
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