[VoiceOps] SIP-to-TDM gateway appliance

Thu Feb 14 16:19:32 EST 2013

I will 5th the TA900s.  These are the most reliable analog/PRI/CAS devices we have used.

Brian

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