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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050124/8e524668/attachment.htm
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@lists.digium.com [mailto:asterisk-users-bounces@lists.digium.com] 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: 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 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20050124/a90506b7/attachment.htm
Please don't post to the list using HTML. On January 24, 2005 04:47 pm, Matt Beebe wrote:> 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 canThis is not true -- I helped run a small ISP for years on CAS T1 (robbed bit signaling) lines and could achieve 53kbps without even trying. 56k modems already detect the timeslots and can "route around" the 1/6 frame "damage" that inband signaling takes. -A.