A carrier I like will be introducing PRI over IP, presumably going thru some sort of gateway box (I'm guessing by Adtran but no data yet). Has anybody set up successfully to work directly with such a feed without bothering to take it down to T1 and use a T1/PRI card? Thanks -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110707/6c6dd32c/attachment.htm>
On 07/07/2011 04:41 PM, eric weaver wrote:> A carrier I like will be introducing PRI over IP, presumably going > thru some sort of gateway box (I'm guessing by Adtran but no data > yet). Has anybody set up successfully to work directly with such a > feed without bothering to take it down to T1 and use a T1/PRI card?Are you talking about a TDMoIP solution? Or are you talking about trunking calls over an IP medium with PRI as the last-mile handoff at both ends? -- Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/
----- Original Message -----> A carrier I like will be introducing PRI over IP, presumably going > thru some sort of gateway box (I'm guessing by Adtran but no data > yet). Has anybody set up successfully to work directly with such a > feed without bothering to take it down to T1 and use a T1/PRI card? >Since you're getting this delivered via IP, I assume you already have 'Internet connectivity' of some sort. So, this "PRI over IP" is likely for voice? In most instances, this just means the carrier is giving you 23 simultaneous channels (or fractional) of VoIP connectivity and calling it a "PRI" for marketing speak. Some examples: http://www.ipcomms.net/html/package-virtualt1.html http://www.didlive.com/virtual_t1.htm I'm not saying this is a bad thing, just that it really isn't anything 'groundbreaking' or 'special'. I *have* used such a service before. In one case, the 'virtual PRI' was terminated to me via SIP via my Asterisk PBX box. In another instance, it was handed off via SIP to a Cisco gateway which presented a standard PRI port to the customer PBX equipment. In either case, the general idea is that you already have IP transport, this is for voice, and the channels are provided by SIP. Your termination equipment could likely be anything that handles SIP. If this turns out to be something *not* disguised as just "SIP in X number of channels form", I'd be interested to hear details. Maybe some sort of TDMoIP service? --Tim
---------------------------------------- From: "eric weaver" <ecweaver at gmail.com> Sent: Thursday, July 07, 2011 4:41 PM To: asterisk-users at lists.digium.com Subject: [asterisk-users] Anybody doing PRI over IP? A carrier I like will be introducing PRI over IP, presumably going thru some sort of gateway box (I'm guessing by Adtran but no data yet). Has anybody set up successfully to work directly with such a feed without bothering to take it down to T1 and use a T1/PRI card? Thanks eric I agree with others that likely what you are getting is a product that is SIP based and it is just being priced and bundled to compete with a PRI connection as most bussiness owners and phone guys know what a PRI is.. We have pri's into gateways that run on our VOIP network and we have sip trunks and we mix services out to our customers based on what the routes require. Most of our up line CLEC's can now deliver their TDM and SIP services in both forms so in most cases we take the SIP version and where the vendor does not support SIP correctly we take their PRI version and convert it to SIP ourselves on our gateways. zktech -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110707/66b5bc5e/attachment.htm>
Good point, folks, and thank you. I don't know yet whether they'll using something that is really a DS1-over-IP but that's what it sounds like. Since this would be a new Asterisk install rather than a legacy PBX, I probably prefer to go with somebody who can do plain SIP/IAX trunking (and not eat 1.5+ MBPS on the net connection constantly). -------------- next part -------------- An HTML attachment was scrubbed... URL: <http://lists.digium.com/pipermail/asterisk-users/attachments/20110709/2e1308f8/attachment.htm>