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Today's Topics:
1. Re: Fwd: Re: SIP-to-TDM gateway appliance (Paul Timmins)
2. Re: SIP-to-TDM gateway appliance (My List Account)
3. Re: SIP-to-TDM gateway appliance (Brian R)
----------------------------------------------------------------------
Message: 1
Date: Thu, 14 Feb 2013 01:51:33 -0500
From: Paul Timmins <[email protected]>
To: [email protected]
Cc: [email protected]
Subject: Re: [VoiceOps] Fwd: Re: SIP-to-TDM gateway appliance
Message-ID: <[email protected]>
Content-Type: text/plain; charset=us-ascii
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 <[email protected]> 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:[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]
>> _______________________________________________
>> VoiceOps mailing list
>> [email protected]
>> https://puck.nether.net/mailman/listinfo/voiceops
>>
>>
>>
>>
>> _______________________________________________
>> VoiceOps mailing list
>> [email protected]
>> https://puck.nether.net/mailman/listinfo/voiceops
>
> --
> 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
> [email protected]
> https://puck.nether.net/mailman/listinfo/voiceops
------------------------------
Message: 2
Date: Thu, 14 Feb 2013 14:13:35 -0500
From: "My List Account" <[email protected]>
To: "'Robert Dawson'" <[email protected]>, "'Matthew Crocker'"
<[email protected]>, "'Nathan Anderson'" <[email protected]>
Cc: [email protected]
Subject: Re: [VoiceOps] SIP-to-TDM gateway appliance
Message-ID: <[email protected]>
Content-Type: text/plain; charset="us-ascii"
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: [email protected] [mailto:[email protected]]
On Behalf Of Robert Dawson
Sent: Tuesday, February 12, 2013 12:45 PM
To: Matthew Crocker; Nathan Anderson
Cc: '[email protected]'
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" <[email protected]> wrote:
>
>
>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=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
>> [email protected]
>> _______________________________________________
>> VoiceOps mailing list
>> [email protected]
>> https://puck.nether.net/mailman/listinfo/voiceops
>>
>>
>>
>>
>> _______________________________________________
>> VoiceOps mailing list
>> [email protected]
>> https://puck.nether.net/mailman/listinfo/voiceops
>>
>
>
>_______________________________________________
>VoiceOps mailing list
>[email protected]
>https://puck.nether.net/mailman/listinfo/voiceops
_______________________________________________
VoiceOps mailing list
[email protected]
https://puck.nether.net/mailman/listinfo/voiceops
------------------------------
Message: 3
Date: Thu, 14 Feb 2013 13:19:32 -0800
From: Brian R <[email protected]>
To: <[email protected]>, <[email protected]>,
<[email protected]>, <[email protected]>
Cc: voiceops <[email protected]>
Subject: Re: [VoiceOps] SIP-to-TDM gateway appliance
Message-ID: <[email protected]>
Content-Type: text/plain; charset="iso-8859-1"
I will 5th the TA900s. These are the most reliable analog/PRI/CAS devices we
have used.
Brian
> From: [email protected]
> To: [email protected]; [email protected]; [email protected]
> Date: Thu, 14 Feb 2013 14:13:35 -0500
> CC: [email protected]
> 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: [email protected] [mailto:[email protected]]
> On Behalf Of Robert Dawson
> Sent: Tuesday, February 12, 2013 12:45 PM
> To: Matthew Crocker; Nathan Anderson
> Cc: '[email protected]'
> 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" <[email protected]> wrote:
>
> >
> >
> >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=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
> >> [email protected]
> >> _______________________________________________
> >> VoiceOps mailing list
> >> [email protected]
> >> https://puck.nether.net/mailman/listinfo/voiceops
> >>
> >>
> >>
> >>
> >> _______________________________________________
> >> VoiceOps mailing list
> >> [email protected]
> >> https://puck.nether.net/mailman/listinfo/voiceops
> >>
> >
> >
> >_______________________________________________
> >VoiceOps mailing list
> >[email protected]
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End of VoiceOps Digest, Vol 44, Issue 12
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