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Today's Topics:
1. Re: SIP-to-TDM gateway appliance (Matthew Crocker)
----------------------------------------------------------------------
Message: 1
Date: Wed, 6 Feb 2013 17:47:07 -0500
From: Matthew Crocker <[email protected]>
To: Nathan Anderson <[email protected]>
Cc: "'[email protected]'" <[email protected]>
Subject: Re: [VoiceOps] SIP-to-TDM gateway appliance
Message-ID: <[email protected]>
Content-Type: text/plain; charset=us-ascii
On Feb 6, 2013, at 5:42 PM, Nathan Anderson <[email protected]> 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:[email protected]]
> 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:[email protected] <mailto:[email protected]>
> w: ringfree.biz
>
>
>
>
> On Feb 6, 2013, at 5:04 PM, Nathan Anderson <[email protected]>
> 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
> [email protected]
> _______________________________________________
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>
>
>
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End of VoiceOps Digest, Vol 44, Issue 7
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