[VoiceOps] Fwd: Re: SIP-to-TDM gateway appliance

Paul Timmins paul at timmins.net
Thu Feb 14 01:51:33 EST 2013 

My experience has been the opposite, even when I found a difficult bug in their DNS cache implementation. I had to push hard at first to get them to realize it was a bug, but once I had a test case, they were very, very interested, and ultimately put out a release JUST to fix the bug I found. (I think it was A2.04, if I recall - it had to do with the first DNS lookup after a DNS TTL expiry would cause a DNS lookup failure. It manifested itself as a failed inbound call attempt about once every 10 minutes.

Adtran finally realized that the reason they didn't see it more often was we run 5 minute TTLs on our DNS so we can change it quickly, and a lot of people run 60 minute or even 6 hour or 24 hour TTLs on their records, so the adtrans would only fail an inbound call once an hour, once every 6 hours, or once a day, and nobody noticed.

Anyway, like I was saying, I'm kind of surprised about the bug patch. Did you follow up by requesting a supervisor, or going through your sales team? Maybe you got a bad tech.

-Paul

On Feb 12, 2013, at 17:51 , Joe Fratantoni <jfratantoni at cygnustel.com> wrote:

> 
> 
> I have to comment that I was pretty dissatisfied with AdTran's customer support and unwillingness to patch a software bug we found in the TotalAccess line (It affects bridging).
> This bad taste in our mouth has caused us to seek out another vendor to meet our needs.
> 
> On 2013-02-06 16:42, Nathan Anderson 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).
>> 
>> -- 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=flypage-ask.tpl&product_id=9&category_id=2&option=com_virtuemart&Itemid=30
>> 
>> 
>> 	http://www.odints.com/pages/prod/completesolutions/alvis-pbx/alvisfs.htm
>> 
>> 	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
>> 	_______________________________________________
>> 	VoiceOps mailing list
>> 	VoiceOps at voiceops.org
>> 	https://puck.nether.net/mailman/listinfo/voiceops
>> 
>> 
>> 
>> 
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> 
> -- 
> Joe Fratantoni
> Cygnus Communications
> 19635 97th Ave
> Mokena, IL 60448
> 815.680.5686 x206
> Business Internet & Phone Services
> 
> -- 
> Joe Fratantoni
> Cygnus Communications
> 19635 97th Ave
> Mokena, IL 60448
> 815.680.5686 x206
> Business Internet & Phone Services
> _______________________________________________
> VoiceOps mailing list
> VoiceOps at voiceops.org
> https://puck.nether.net/mailman/listinfo/voiceops

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