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

Thu Feb 14 01:02:20 EST 2013

Out of curiosity, what was the bug and why would they refuse to fix it?  That seems rather odd.

-- Nathan 

-----Original Message-----
From: voiceops-bounces at voiceops.org [mailto:voiceops-bounces at voiceops.org] On Behalf Of Joe Fratantoni
Sent: Tuesday, February 12, 2013 2:52 PM
To: voiceops at voiceops.org
Subject: [VoiceOps] Fwd: Re: SIP-to-TDM gateway appliance

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
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>
>
>
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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
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