Correct, CAS can supply DNIS but the call set up times are significantly longer.
> -----Original Message----- > From: [EMAIL PROTECTED] [mailto:asterisk-users- > [EMAIL PROTECTED] On Behalf Of David Boyd > Sent: Monday, January 24, 2005 7:12 PM > To: Asterisk Users Mailing List - Non-Commercial Discussion > Subject: RE: [Asterisk-Users] T1 E&M vs PRI question > > 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: [EMAIL PROTECTED] > > [mailto:[EMAIL PROTECTED] On Behalf Of Matt > > Beebe > > Sent: Monday, January 24, 2005 2:47 PM > > To: asterisk-users@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 > > > > > > > > > > > > > > > ____________________________________________________________________ > __ > > _______________________________________________ > > Asterisk-Users mailing list > > Asterisk-Users@lists.digium.com > > http://lists.digium.com/mailman/listinfo/asterisk-users > > To UNSUBSCRIBE or update options visit: > > http://lists.digium.com/mailman/listinfo/asterisk-users > > _______________________________________________ > Asterisk-Users mailing list > Asterisk-Users@lists.digium.com > http://lists.digium.com/mailman/listinfo/asterisk-users > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users _______________________________________________ Asterisk-Users mailing list Asterisk-Users@lists.digium.com http://lists.digium.com/mailman/listinfo/asterisk-users To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users