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Today's Topics:
1. VoIP number can't be verified for Google ad campaigns
(Carlos Alvarez)
2. Re: SIP-to-TDM gateway appliance (Robert Dawson)
3. Re: VoIP number can't be verified for Google ad campaigns
(Carlos Alvarez)
4. Fwd: Re: SIP-to-TDM gateway appliance (Joe Fratantoni)
5. Re: VoIP number can't be verified for Google ad campaigns
(James Cloos)
6. Re: Fwd: Re: SIP-to-TDM gateway appliance (Nathan Anderson)
----------------------------------------------------------------------
Message: 1
Date: Mon, 11 Feb 2013 14:58:04 -0700
From: Carlos Alvarez <[email protected]>
To: [email protected]
Subject: [VoiceOps] VoIP number can't be verified for Google ad
campaigns
Message-ID:
<CAFn1dUEv26GSt8uq=vozboukakuj6frhn+keg_uz5xdemr2...@mail.gmail.com>
Content-Type: text/plain; charset="iso-8859-1"
We're a small ITSP, and get most of our numbers through
Onvoy/Zayo/360networks. We also use a variety of aggregators and other
carriers. Today a customer called and said that when he tried to create a
Google ad campaign with one of his DIDs, it said the number couldn't be
verified. The error message said this happens with prepaid cell phones and
"some" VoIP services. Neither of us know what Google is looking for. Is
it the CNAM for the number, or a white pages listing, or something else?
As a normal practice we don't put CNAM on DIDs if they are not used for
CLID on outbound calls, and of course we don't do directory entries for DID
numbers. Any ideas?
The specific number in question is through Onvoy.
--
Carlos Alvarez
TelEvolve
602-889-3003
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Message: 2
Date: Tue, 12 Feb 2013 17:44:50 +0000
From: Robert Dawson <[email protected]>
To: Matthew Crocker <[email protected]>, Nathan Anderson
<[email protected]>
Cc: "'[email protected]'" <[email protected]>
Subject: Re: [VoiceOps] SIP-to-TDM gateway appliance
Message-ID: <cd3fe847.27abf%[email protected]>
Content-Type: text/plain; charset="us-ascii"
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=flypa
>>ge-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
>>
>
>
>_______________________________________________
>VoiceOps mailing list
>[email protected]
>https://puck.nether.net/mailman/listinfo/voiceops
------------------------------
Message: 3
Date: Mon, 11 Feb 2013 19:39:13 -0700
From: Carlos Alvarez <[email protected]>
To: [email protected]
Subject: Re: [VoiceOps] VoIP number can't be verified for Google ad
campaigns
Message-ID:
<cafn1duhruuva+djulz4t96ewa_falxqbnabju0qsrygqfeu...@mail.gmail.com>
Content-Type: text/plain; charset="iso-8859-1"
On Mon, Feb 11, 2013 at 5:34 PM, James Cloos <[email protected]> wrote:
>
> AIUI, they try to call the number and, upon answer, tts a verification
> code which one then submits back via a web form.
>
That's done on some Google products, but not all. Many actually want to
verify that you own a real business number. They also won't accept a
prepaid cell for these services.
> They may refuse to dial a number if the rate they'd have to pay is too
> high. I've seen that with gvoice vs rural non-Bell ILECs.
>
This is a major metro in LATA 666, one of the cheapest rate centers.
I've had a few discussions off-list with people from the list. I
especially want to acknowledge the two people from 360 and Zayo who helped
a lot. We all tend to think that the next step I should try is setting
CNAM and directory listing for a number and have the customer try again
once that's in place.
--
Carlos Alvarez
TelEvolve
602-889-3003
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Message: 4
Date: Tue, 12 Feb 2013 16:51:58 -0600
From: Joe Fratantoni <[email protected]>
To: <[email protected]>
Subject: [VoiceOps] Fwd: Re: SIP-to-TDM gateway appliance
Message-ID: <[email protected]>
Content-Type: text/plain; charset=UTF-8; format=flowed
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
------------------------------
Message: 5
Date: Mon, 11 Feb 2013 19:34:30 -0500
From: James Cloos <[email protected]>
To: Carlos Alvarez <[email protected]>
Cc: [email protected]
Subject: Re: [VoiceOps] VoIP number can't be verified for Google ad
campaigns
Message-ID: <[email protected]>
Content-Type: text/plain
>>>>> "CA" == Carlos Alvarez <[email protected]> writes:
CA> Today a customer called and said that when he tried to create a
CA> Google ad campaign with one of his DIDs, it said the number couldn't
CA> be verified.
AIUI, they try to call the number and, upon answer, tts a verification
code which one then submits back via a web form.
Ie, they same thing they do when one adds a target number to one's
gvoice account.
They may refuse to dial a number if the rate they'd have to pay is too
high. I've seen that with gvoice vs rural non-Bell ILECs.
-JimC
--
James Cloos <[email protected]> OpenPGP: 1024D/ED7DAEA6
------------------------------
Message: 6
Date: Wed, 13 Feb 2013 22:02:20 -0800
From: Nathan Anderson <[email protected]>
To: "'[email protected]'" <[email protected]>,
"'[email protected]'" <[email protected]>
Subject: Re: [VoiceOps] Fwd: Re: SIP-to-TDM gateway appliance
Message-ID:
<[email protected]>
Content-Type: text/plain; charset="us-ascii"
Out of curiosity, what was the bug and why would they refuse to fix it? That
seems rather odd.
-- Nathan
-----Original Message-----
From: [email protected] [mailto:[email protected]] On
Behalf Of Joe Fratantoni
Sent: Tuesday, February 12, 2013 2:52 PM
To: [email protected]
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:[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
------------------------------
_______________________________________________
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End of VoiceOps Digest, Vol 44, Issue 11
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