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I will 5th the TA900s. These are the most reliable analog/PRI/CAS devices we have used.<BR>
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Brian<BR> <BR>
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<DIV id=SkyDrivePlaceholder></DIV>> From: mylists@battleop.com<BR>> To: RDawson@alliedtelecom.net; matthew@corp.crocker.com; nathana@fsr.com<BR>> Date: Thu, 14 Feb 2013 14:13:35 -0500<BR>> CC: voiceops@voiceops.org<BR>> Subject: Re: [VoiceOps] SIP-to-TDM gateway appliance<BR>> <BR>> I'll 4th this as well. I have had a couple of TA900s die from various<BR>> causes but I am not convinced these were Adtran problems. In every case we<BR>> open a ticket with Adtran and they issue an RMA without a hassle. Their<BR>> support has been great and they don't charge you for support and updates.<BR>> We use the TA900s on the majority all of our PRI/CAS hand offs or when we<BR>> need to do T38.<BR>> <BR>> We make limited use of the Cisco 2431-8fxs and 2431-16fxs for analog POTS<BR>> type hand offs. They are cheap on the secondary market (you can't find<BR>> TA900s secondary now) but support is very limited since no one really knows<BR>> much about them.<BR>> <BR>> <BR>> Richey<BR>> <BR>> -----Original Message-----<BR>> From: voiceops-bounces@voiceops.org [mailto:voiceops-bounces@voiceops.org]<BR>> On Behalf Of Robert Dawson<BR>> Sent: Tuesday, February 12, 2013 12:45 PM<BR>> To: Matthew Crocker; Nathan Anderson<BR>> Cc: 'voiceops@voiceops.org'<BR>> Subject: Re: [VoiceOps] SIP-to-TDM gateway appliance<BR>> <BR>> I'll second (or third . . .) the Adtran TA900 series. We use them for PRI,<BR>> T1-CAS, analog, pretty much anything you would want to do with them they can<BR>> handle. They support PAI, you can set the number of digits transferred or<BR>> you can perform extensive manipulation of DNIS/ANI, pretty much rock solid<BR>> on t.38, great devices.<BR>> <BR>> Good support and (knocking on wood) have never had one actually "fail".<BR>> <BR>> On 2/6/13 5:47 PM, "Matthew Crocker" <matthew@corp.crocker.com> wrote:<BR>> <BR>> ><BR>> ><BR>> >On Feb 6, 2013, at 5:42 PM, Nathan Anderson <nathana@fsr.com> wrote:<BR>> ><BR>> >> (remember to "Reply All"! :-))<BR>> >> <BR>> >> Holy crap. I don't know how I missed the pricing for AdTran Total <BR>> >>Access. I guess after I saw what AudioCodes and MediaTrix and Sangoma <BR>> >>go for on average, I must have made an assumption about AdTran pricing.<BR>> >>That totally blows Digium's seemingly-aggressive pricing out of the <BR>> >>water, especially if it covers all of my use-cases (which I already <BR>> >>know the Digium doesn't).<BR>> ><BR>> >The 10 year warranty doesn't suck either ;)<BR>> ><BR>> >I love the Adtran TA-9xx. It is a swiss-army knife of VoIP.<BR>> ><BR>> >The only issue is they don't handle 208v very well (i.e at all). we <BR>> >released the magic blue smoke in our lab. The warranty covered the <BR>> >repair though :)<BR>> ><BR>> >> <BR>> >> -- Nathan<BR>> >> <BR>> >> -----Original Message-----<BR>> >> From: David Wessell [mailto:david@ringfree.biz]<BR>> >> Sent: Wednesday, February 06, 2013 2:15 PM<BR>> >> To: Nathan Anderson<BR>> >> Subject: Re: [VoiceOps] SIP-to-TDM gateway appliance<BR>> >> <BR>> >> Seconded. This is a killer topic. We've just closed our first deal <BR>> >>for this type of situation. I had planned on going with a Adtran 904 <BR>> >>($725 on NewEgg) but am very interested to hear other options.<BR>> >> <BR>> >> Thanks<BR>> >> David<BR>> >> <BR>> >> <BR>> >> <BR>> >> <BR>> >> <BR>> >> David Wessell<BR>> >> Chief Packet Slinger<BR>> >> Ringfree Communications, LLC<BR>> >> t: 828-575-0030<BR>> >> e:david@ringfree.biz <mailto:david@ringfree.biz><BR>> >> w: ringfree.biz<BR>> >> <BR>> >> <BR>> >> <BR>> >> <BR>> >> On Feb 6, 2013, at 5:04 PM, Nathan Anderson <nathana@fsr.com><BR>> >> wrote:<BR>> >> <BR>> >> <BR>> >> I know this has been a topic of conversation in the past, but things<BR>> <BR>> >>might have changed since the last discussion and I'm wondering what <BR>> >>the market is currently like for such devices.<BR>> >> <BR>> >> We deliver voice strictly via SIP/RTP, but naturally there are some <BR>> >>potential customers out there that still have an older, non-IP-aware <BR>> >>PBX that they're not ready to throw out yet. What are the best and <BR>> >>most cost-effective gateway options out there at this time? We are <BR>> >>specifically looking for one that has a single T1 interface that can <BR>> >>operate in either CAS or PRI modes.<BR>> >> <BR>> >> Special requirements:<BR>> >> <BR>> >> 1) We need to be able to do DID manipulation between T1 and SIP; I <BR>> >>presume this is a rather standard feature in most gateways given that <BR>> >>most SIP trunk providers will send at least 10-digit DNIS (in the <BR>> >>INVITE and "To" fields) but DNIS on PRI is often only the last 3 or 4 <BR>> >>digits of the TN.<BR>> >> <BR>> >> 2) There may be certain situation where we want to leave the PBX <BR>> >>configuration as untouched/unchanged as possible (drop-in replacement <BR>> >>service), and where there is no correllation between target DID and <BR>> >>the telephone number (e.g., 212-555-1212 is called, PBX is sent 4001). <BR>> >>We'd like a gateway where static mappings like that for DID <BR>> >>manipulation are possible, rather than just a general rule that says <BR>> >>"strip the first 6 digits off before sending to the PRI".<BR>> >> <BR>> >> 3) For outgoing calls, the device needs to put the calling DID (the <BR>> >>desired Caller-ID/ANI) in the PAI header, and also needs to be able to <BR>> >>be configured to override "From" with a static alphanumeric value (so <BR>> >>"From" and PAI should not match; "From" will not contain the desired <BR>> >>ANI).<BR>> >> <BR>> >> 4) In T1 CAS singalling modes such as E&M Wink where it is possible <BR>> >>to transmit CLID and target DID information via DTMF to the PBX, <BR>> >>different PBXes potentially have different formats that they want to <BR>> >>see this information in; for example, a Nortel Norstar would expect to <BR>> >>see<BR>> >>*CALLERID*DNIS* (e.g., *2125550001*1212* where the caller is<BR>> >>212-555-0001 and the destination is 212-555-1212). Are there any <BR>> >>gateways that support this?<BR>> >> <BR>> >> 5) It needs to have a T.38 gateway mode that can recognize a fax <BR>> >>call, either send or accept a re-INVITE with a T.38 SDP as <BR>> >>appropriate, and perform the "transcoding" from/to T.38 between the T1 <BR>> >>channel and the RTP session. Just resorting to G.711 for fax <BR>> >>passthrough is not desireable...any gateway can do that.<BR>> >> <BR>> >> 6) If in T1 CAS mode, and the PBX takes a channel "off-hook" to <BR>> >>place an outbound call, the gateway should generate an audible dialtone.<BR>> >> <BR>> >> ...and, of course, it would be nice if we could find such a device <<BR>> <BR>> >>$1,000. :-P<BR>> >> <BR>> >> I know I could build one myself with a mini PC and a single-span T1 <BR>> >>card that was running Asterisk 10 and easily hit that price point, but <BR>> >>I'd rather find a supported, off-the-shelf solution to sell to our <BR>> >>customers, if possible.<BR>> >> <BR>> >> There are the "usual suspects", of course: AdTran, MediaTrix, <BR>> >>AudioCodes, and so forth. AdTran seems to get talked about a lot here.<BR>> >>Let's say price was no object for a second. Does anyone know if there <BR>> >>is a model amongst any of the ones these manufacturers produce that <BR>> >>fulfills the above list of requirements?<BR>> >> <BR>> >> Does anybody have any experience with Digium's relatively new line <BR>> >>of gateways (G100/G200)? I think it would support some of these <BR>> >>scenarios<BR>> >>(#1 and #3) but I'm not sure about the remaining ones. Unfortunately, <BR>> >>although it most certainly runs on an Asterisk core, that core is only <BR>> >>exposed to you through a clever but still-limited GUI; with direct <BR>> >>access to the dialing plan (extensions.conf) I could accomplish all of <BR>> >>these things myself. The price is certainly right, though.<BR>> >> <BR>> >> If only somebody made a reasonably-priced single-board-computer that<BR>> <BR>> >>ran raw, embedded Asterisk and had a single-span T1 interface on it. <BR>> >>Oh wait, somebody does!:<BR>> >> <BR>> >> <BR>> >> <BR>> >>http://switchvoice.com/index.php?page=shop.product_details&flypage=fly<BR>> >>pa<BR>> >>ge-ask.tpl&product_id=9&category_id=2&option=com_virtuemart&Itemid=30<BR>> >> <BR>> >> <BR>> >> <BR>> >>http://www.odints.com/pages/prod/completesolutions/alvis-pbx/alvisfs.h<BR>> >>tm<BR>> >> <BR>> >> Only problem is that the first company doesn't have a U.S.<BR>> >>distributor, and the second doesn't have a distributor that sells in <BR>> >>single-unit quantities.<BR>> >> <BR>> >> Would love to hear y'all's thoughts on this subject.<BR>> >> <BR>> >> Thanks,<BR>> >> <BR>> >> -- <BR>> >> Nathan Anderson<BR>> >> First Step Internet, LLC<BR>> >> nathana@fsr.com<BR>> >> _______________________________________________<BR>> >> VoiceOps mailing list<BR>> >> VoiceOps@voiceops.org<BR>> >> https://puck.nether.net/mailman/listinfo/voiceops<BR>> >> <BR>> >> <BR>> >> <BR>> >> <BR>> >> _______________________________________________<BR>> >> VoiceOps mailing list<BR>> >> VoiceOps@voiceops.org<BR>> >> https://puck.nether.net/mailman/listinfo/voiceops<BR>> >> <BR>> ><BR>> ><BR>> >_______________________________________________<BR>> >VoiceOps mailing list<BR>> >VoiceOps@voiceops.org<BR>> >https://puck.nether.net/mailman/listinfo/voiceops<BR>> <BR>> <BR>> _______________________________________________<BR>> VoiceOps mailing list<BR>> VoiceOps@voiceops.org<BR>> https://puck.nether.net/mailman/listinfo/voiceops<BR>> <BR>> _______________________________________________<BR>> VoiceOps mailing list<BR>> VoiceOps@voiceops.org<BR>> https://puck.nether.net/mailman/listinfo/voiceops<BR></DIV> </div></body>
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