[VoiceOps] SIP-to-TDM gateway appliance

Thu Feb 14 23:01:24 EST 2013

And just to be another voice echoing, a 6th.. have 100's of TA's deployed
for CAS or PRI translation at the edge..

One thing to note, 30 dsp channels on the base box.  Pretty sure it won't
run out on a full T, 23 or 24 for CAS channels
doing G711/G729/RFC2833.  TA's aren't dirt cheap, but roughly equivalent in
virtually every capability to something like a Cisco 2431 IAD, for at least
50% of the price, if not less.  As others have said, complimentary 10 years
hardware and firmware update support out of the box.  Their support groups
are fairly small, they do a solid job, and I've had times when I was a
little frustrated for a minute, but generally they've done a pretty bang up
job.  I love dealing with them, compared to other vendors..

Things to watch out for:

Netflow cache sample rate, you want it to almost anytime you've got ip flow
export destination ip.ad.re.ss configured you'll want to.

Turn on ip ffe anywhere you can show proc queue on a lightly loaded T1
doing a bunch of short calls and see why (hint, PacketRouting goes about
87, you drop packets).

debug isdn l2-formatted is your best friends, but you'd best be in via ssh
and ready to kick your connection if you get overloaded.

"no events" after logging in will get rid of the scroll messages.  Most
stuff configures almost exactly like IOS on AOS, but the voice stuff makes
a lot more sense.  You can virtually go through the options and see how
it's done pretty quick.  Configs on these devices unless you're going to do
massive amounts of DNS or 18 SIP trunks feeding the PRI, are a couple to a
few K.  Some other cool stuff you can do when looking to control the
routing when there are multiple PRI's or CAS hitting the customer equipment
are things like accept and reject statements on the CAS/PRI trunks.

ACL's are pretty straightforward as is SNMP, Syslog, Traps.. not much to
complain about.. a couple of things here an there that you can't poll that
you should be able like, IMO having to do with like active calls, call
processing and stuff, but I haven't looked at AOS enterprise MIBs too much
since A1.xx - the T1 performance stats are readily available, both current
and interval.. Like that stuff when landing T1's from the Adtran to
someone's gear 350' away on a flaky cable.

Peter

On Thu, Feb 14, 2013 at 1:19 PM, Brian R <briansupport at hotmail.com> wrote:

>  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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> > >> VoiceOps at voiceops.org
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> > >>
> > >>
> > >>
> > >>
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> > >>
> > >
> > >
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-- 
Peter Serwe
http://truthlightway.blogspot.com/
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