Sunday, May 31, 2009

Re: [asterisk-biz] UK SIP trunking

info wrote:
> Hi,
>
> This is standard and very simple service to set by Operators in UK who are
> trustworthy not to abuse "CLI Manipulation".
>
> It is normal for these services in application like "Virtual office" both
> voice and fax where return path lands into extension of caller in
> "Asterisk world"
>
> Provider will need to convince proper Carriers that they do not wish to
> run " Missing Calls Application" where gullible mobile callers return
> calls to missed calls which generate revenue to the called party.
>
> Cheers,
>
> Mo
>
>
> http://www.telpoint.co.uk
>
>> We are looking into the possibilities of hooking our asterisk system
>> into a SIP trunk for inbound and outbound calls in the UK.
>>
>> One of the requirements would be to set our caller id to any number used
>> by us (including 0845 / 0800)
>>
>> We run up to 50 simultaneous external calls, and are currently using 1.4
>> of asterisk.
>>
>> Does anyone know of such a provider in the UK ? Has anyone had any
>> dealings with Spitfire
>> (http://www.spitfire.co.uk/SIP_Trunking_tel.shtml?headerbar=1)
>>
>> Thanks!
>>
>> Julian

Hi

We maybe able to help. Pls contact steve at bicomsystems dot com

Thanks

Senad


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Re: [asterisk-biz] Malaysia DID Needed

Mr Abdul,

We are a wholesale carrier interconnected with different carriers throughout the world.

Moreover the current requirement of Malaysia DID is mainly for calling card and call back platform we are launching in Malaysia.

 

Regards

 

Vijay Gandhi

GIPL(An ISO 9001:2000 Company)

+91-9811688460

+44-2080992384

vijay@gandhiinfotech.com

www.gandhiinfotech.com

 

From: Abdul Hakeem [mailto:alhakeem@gmail.com]
Sent: Sunday, May 31, 2009 6:55 PM
To: vijay@gandhiinfotech.com; 'Commercial and Business-Oriented Asterisk Discussion'
Subject: RE: [asterisk-biz] Malaysia DID Needed

 

Hello,

 

Could you tell me a bit more about your Voip services ?

I am looking at it from the wholesale side.

Best regards,

Abdul Hakeem

 

From: asterisk-biz-bounces@lists.digium.com [mailto:asterisk-biz-bounces@lists.digium.com] On Behalf Of Vijay Gandhi
Sent: Saturday, May 30, 2009 5:20 PM
To: 'Commercial and Business-Oriented Asterisk Discussion'
Subject: [asterisk-biz] Malaysia DID Needed

 

We need DID’s to Malaysia for calling card and call back platform.

We need about 20-30 channels to start with, anyone in the list can offer us the same at a reasonable price with 99.9% uptime.

 

 

Regards

 

Vijay Gandhi

GIPL(An ISO 9001:2000 Company)

+91-9811688460

+44-2080992384

vijay@gandhiinfotech.com

www.gandhiinfotech.com

 

Re: [asterisk-biz] Malaysia DID Needed

Hello,

 

Could you tell me a bit more about your Voip services ?

I am looking at it from the wholesale side.

Best regards,

Abdul Hakeem

 

From: asterisk-biz-bounces@lists.digium.com [mailto:asterisk-biz-bounces@lists.digium.com] On Behalf Of Vijay Gandhi
Sent: Saturday, May 30, 2009 5:20 PM
To: 'Commercial and Business-Oriented Asterisk Discussion'
Subject: [asterisk-biz] Malaysia DID Needed

 

We need DID’s to Malaysia for calling card and call back platform.

We need about 20-30 channels to start with, anyone in the list can offer us the same at a reasonable price with 99.9% uptime.

 

 

Regards

 

Vijay Gandhi

GIPL(An ISO 9001:2000 Company)

+91-9811688460

+44-2080992384

vijay@gandhiinfotech.com

www.gandhiinfotech.com

 

Saturday, May 30, 2009

[asterisk-biz] Malaysia DID Needed

We need DID’s to Malaysia for calling card and call back platform.

We need about 20-30 channels to start with, anyone in the list can offer us the same at a reasonable price with 99.9% uptime.

 

 

Regards

 

Vijay Gandhi

GIPL(An ISO 9001:2000 Company)

+91-9811688460

+44-2080992384

vijay@gandhiinfotech.com

www.gandhiinfotech.com

 

Friday, May 29, 2009

Re: [asterisk-biz] White Routes

Do you have something for Benin ??

 

From: asterisk-biz-bounces@lists.digium.com [mailto:asterisk-biz-bounces@lists.digium.com] On Behalf Of Abdul Hakeem
Sent: May-29-09 5:40 AM
To: 'Commercial and Business-Oriented Asterisk Discussion'
Subject: Re: [asterisk-biz] White Routes

 

Patrick,

Could you send your company and contact details ?

Cheers,

Abdul Hakeem

IPEX Telecom

+447931800952

 

From: asterisk-biz-bounces@lists.digium.com [mailto:asterisk-biz-bounces@lists.digium.com] On Behalf Of Tamo Tamo
Sent: Thursday, April 09, 2009 5:30 PM
To: asterisk-biz@lists.digium.com
Subject: [asterisk-biz] White Routes

 

Hello,

 

We have the following routes available

 

Mexico Mobile   $0.0845   Billing 60/60  2 E1

Nigeria Mobile    $0.085    Billing 1/1      3 E1

 

Prepay Only.

 

Patrick.

Re: [asterisk-biz] Mobile extensions from Asterisk - HLR registration?

Hi John!

> Sounds promising, but I'm afraid that German is not a language I
> speak or understand, and various translation systems are not quite at
> the level where they are able to adequately interpret technical or
> marketing documentation. Do you have any idea where details might be
> found on their service, described in English?

I knew you'd ask for that, but it appears they simply don't have an
Englisch version of their web site. So apart from using the Google
translation service your option is to learn the language that once was
the scientific lingua franca. ;->

About their VoIP service: If you register with a SIP UA then calls
directed to your mobile number will ring on your VoIP phone; you un-
register and your mobile rings again (at least that's why I have
understood, didn't try their offering). This is a nice way to avoid
international roaming charges - despite the current efforts of the EU
commission those are still too high in Europe (and of course the rest of
the world, but that is outside the scope of the EC).

> >> If it were possible to selectively and temporarily re-map a mobile
> >> device so that inbound calls and SMS messages ended up on your Asterisk
> >> server, delivered via SIP and/or other IP-based transmission mechanisms
> >
> > The mobile carrier (reseller) solomo.de does that already for quite a
> > while.

Philipp

--
| aixvox GmbH
| Philipp von Klitzing
| Beratung und Technologie
|
| Monheimsallee 22
| 52062 Aachen · Germany
|
| T: +49.241.4133 141
| F: +49.241.4133 222
| E: mailto:pk@aixvox.com
|
| http://www.voice-compass.com
| http://www.aixvox.com


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Re: [asterisk-biz] Mobile extensions from Asterisk - HLR registration?

On Fri, 2009-05-29 at 11:56 -0400, John Todd wrote:
> Philipp -
> Sounds promising, but I'm afraid that German is not a language I
> speak or understand, and various translation systems are not quite at
> the level where they are able to adequately interpret technical or
> marketing documentation. Do you have any idea where details might be
> found on their service, described in English?
>
> JT
>
translate.google.com ?


--
Trixter http://www.0xdecafbad.com Bret McDanel
pgp key: http://pgp.mit.edu:11371/pks/lookup?op=get&search=0x8AE5C721

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Re: [asterisk-biz] Mobile extensions from Asterisk - HLR registration?

Philipp -
Sounds promising, but I'm afraid that German is not a language I
speak or understand, and various translation systems are not quite at
the level where they are able to adequately interpret technical or
marketing documentation. Do you have any idea where details might be
found on their service, described in English?

JT

On May 29, 2009, at 11:05 AM, Philipp von Klitzing wrote:

> Hi!
>
>> If it were possible to selectively and temporarily re-map a mobile
>> device so that inbound calls and SMS messages ended up on your
>> Asterisk server, delivered via SIP and/or other IP-based transmission
>> mechanisms
>
> The mobile carrier (reseller) solomo.de does that already for quite a
> while.
>
> Philipp
>
> --
> | aixvox GmbH
> | Philipp von Klitzing
> | Beratung und Technologie
> |
> | Monheimsallee 22
> | 52062 Aachen · Germany
> |
> | T: +49.241.4133 141
> | F: +49.241.4133 222
> | E: mailto:pk@aixvox.com
> |
> | http://www.voice-compass.com
> | http://www.aixvox.com

---
John Todd email:jtodd@digium.com
Digium, Inc. | Asterisk Open Source Community Director
445 Jan Davis Drive NW - Huntsville AL 35806 - USA
direct: +1-256-428-6083 http://www.digium.com/


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[asterisk-biz] Jamaica mobile

Good day,

I have Jamaica Mobile white route available. Rate is USD$0.145. Capacity 1
T1

Please contact me off list if interested.

Regards,

VoIP Carib


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[asterisk-biz] (7) Brand New Audiocodes MP108-FXS For Sale First $200/ea Takes Them

I have a client consignment of (7) Brand New Audiocodes MP108-FXS, eight port FXS SIP Gateways.  These are new retail, sealed.   Normal retail is $700-$800.  I will take $200/ea for these, prefer to sell them all to a single entity.  Can ship today.

 

Thanks

 

Cory J. Andrews

Director New Market Initiatives

 

Sayers Media Group

VoIP Supply, LLC

454 Sonwil Drive

Buffalo, NY 14225

716-250-3402 OFFICE

716-630-1548 FAX

716-601-4474 MOBILE

candrews@sayersmedia.com

 

 

Have I exceeded your expectations?  Please share your experience with my boss,  Benjamin P. Sayers, CEO

 

NOTICE: The information contained in this email and any document attached hereto is intended only for the named recipient(s). It is the property of the VoIP Supply, LLC and shall not be used, disclosed or reproduced without the express written consent of VoIP Supply, LLC. If you are not the intended recipient, nor the employee or agent responsible for delivering this message in confidence to the intended recipient(s), you are hereby notified that you have received this transmittal in error, and any review, dissemination, distribution or copying of this transmittal or its attachments is strictly prohibited. If you have received this transmittal and/or attachments in error, please notify me immediately by reply e-mail or telephone and then delete this message, including any attachments. Our mailing address is 454 Sonwil Drive, Buffalo, NY 14225 USA.

 

 

Re: [asterisk-biz] Mobile extensions from Asterisk - HLR registration?

Hi!

> If it were possible to selectively and temporarily re-map a mobile
> device so that inbound calls and SMS messages ended up on your
> Asterisk server, delivered via SIP and/or other IP-based transmission
> mechanisms

The mobile carrier (reseller) solomo.de does that already for quite a
while.

Philipp

--
| aixvox GmbH
| Philipp von Klitzing
| Beratung und Technologie
|
| Monheimsallee 22
| 52062 Aachen · Germany
|
| T: +49.241.4133 141
| F: +49.241.4133 222
| E: mailto:pk@aixvox.com
|
| http://www.voice-compass.com
| http://www.aixvox.com


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Re: [asterisk-biz] Mobile extensions from Asterisk - HLR registration?

I've had a few conversations in the last week with mobile carriers and
equipment vendors who have expressed interest in this concept. Some
of you have contacted me as potential consumers of such a service.
But the scale of interest is in question, and as I am not actually
developing this product I have only what is discussed on this list (at
the moment) as valid data to give to some organization who might be
considering such a service.

If it were possible to selectively and temporarily re-map a mobile
device so that inbound calls and SMS messages ended up on your
Asterisk server, delivered via SIP and/or other IP-based transmission
mechanisms, would you buy such a service? Assume it would be secure,
and authenticated both from the server perspective and from the end
device owner.

Private replies welcome.

JT


On May 20, 2009, at 11:00 AM, John Todd wrote:

>
> I've posed this question in person to people with some frequency over
> the last few years, and the answer has always been "No, I don't know
> of such a service." but I'll try on the list and see what I get.
>
> I think it would be a great asset to the Asterisk community to have a
> service provider (let's call them "SP-A") who is a mobile carrier who
> offered the following method: if I register a SIP entity with their
> servers, they would then register with the HLR of my mobile carrier
> ("SP-B") and act as if I was roaming into a mobile network operated
> by SP-A. SP-B would then take all calls and text messages destined
> for my mobile device and send them to SP-A. SP-A in turn would then
> relay those calls and messages to my Asterisk server, via SIP and/or
> XMPP.
>
> I would have pre-registered my mobile number with SP-B and
> authenticated that I was the owner of that mobile number. SP-A would
> hopefully charge very little for the calls - perhaps a slight premium
> on what I'd expect for a SIP carrier.
>
> This would, I believe, quickly make Asterisk a roaming-capable
> solution for mobile devices. Local Asterisk servers would be able to
> (as an example) detect dual-mode devices and then route calls in the
> office in the appropriate manner. Bluetooth could be used as the
> "trigger" for non-dual-mode phones. I have faith that Asterisk
> developers and administrators would descend upon this type of service
> like locusts. The trick would be to make it purchase-able by
> individuals, and not as some large-scale process that involved sales
> contracts and NDAs and the like. This needs to be a web form, a
> credit card/paypal account, and some good documentation.
>
> Potential problems: what if my mobile phone is registered with SP-A or
> some other provider already? Who gets the messages? How do HLRs
> manage multi-registration conflicts?
>
> JT

---
John Todd email:jtodd@digium.com
Digium, Inc. | Asterisk Open Source Community Director
445 Jan Davis Drive NW - Huntsville AL 35806 - USA
direct: +1-256-428-6083 http://www.digium.com/


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Re: [asterisk-biz] Austria Mobile - T-mobile

Hello,
Are you still offering T-Mobile Austria ?
If not let me know if you have access to simcards as I am able to provide
gsm gateway.
Best regards,
Abdul Hakeem
IPEX Telecom
+447931800952

-----Original Message-----
From: asterisk-biz-bounces@lists.digium.com
[mailto:asterisk-biz-bounces@lists.digium.com] On Behalf Of VoIP Carib
Sent: Friday, March 20, 2009 5:14 AM
To: asterisk-biz@lists.digium.com
Subject: [asterisk-biz] Austria Mobile - T-mobile


Hello,

We have Austria Cell (T-Mobile) for sale, 6 channels. Price USD$0.11

Please contact me if interested

Regards,

VoIP Carib

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Re: [asterisk-biz] White Routes

Patrick,

Could you send your company and contact details ?

Cheers,

Abdul Hakeem

IPEX Telecom

+447931800952

 

From: asterisk-biz-bounces@lists.digium.com [mailto:asterisk-biz-bounces@lists.digium.com] On Behalf Of Tamo Tamo
Sent: Thursday, April 09, 2009 5:30 PM
To: asterisk-biz@lists.digium.com
Subject: [asterisk-biz] White Routes

 

Hello,

 

We have the following routes available

 

Mexico Mobile   $0.0845   Billing 60/60  2 E1

Nigeria Mobile    $0.085    Billing 1/1      3 E1

 

Prepay Only.

 

Patrick.

Thursday, May 28, 2009

[asterisk-biz] Friday at 12 Noon EDT: Jim Van Meggelen on the VoIP Users Conference

Hi,

Like me, some of you probably remember Jim as one of the pioneers
along with Leif and Jarod. These guys "wrote the book", literally. Jim
is our guest tomorrow and he'll be talking about system building,
among other things. We always have a good time AND get stuff done on
the Conference so come by and join us:

Silently, on IRC: #voip-users-conference (freenode.net) - stop by any
time, there'll usually be a party starting an hour or so before noon
EDT
Streamingly: http://www.voipusersconference.org/listen-now/
SIP in g722 wideband via ZipDX or g711 via Talkshoe (see the site
http://tr.im/voip for instructions and URI)
PSTN: (724) 444-7444 and enter 22622# PIN# - Get your PIN at
Talkshoe.com or enter 1# instead

The conference gives us all a chance to hobnob with celebrities like
John Todd, find Asterisk people and groups in your neck of the woods
and just generally usher out the week with a little light-sided view
of some great technology.

Thanks again to all of you who've been there and made it happen.

For those of you who just getting in to Asterisk and VoIP, here's more
about Jim:

"Jim is probably a bit of a masochist, which would explain why he got
into the telecom business in the first place, and why he now loves
Asterisk. Jim is pretty friendly, kinda like a puppy that gets your
shoes dirty. His enthusiasm is infectious, but also a little bit
frightening if you stand too close. Jim is a partner in Core Telecom
Innovations Inc, and iConverged Inc. He lives in Toronto with his wife
and three kids, and loves writing, photography, speaking, improv,
choral singing, and old shoes."

About the shoes...

/r

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[asterisk-biz] T.38 Fax termination to Thailand (66)

Dear sir,
I can provide T.38 Fax termination to Thailand.
If you interest please email me directly.

Best regard.

Dome C.

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Re: [asterisk-biz] Carrier to spain

I may be getting rusty here, but I thought 6/6 was the least any reseller would sell to anyone, including mobile carriers and PTTs.  Wow, finally I am learning something I didn’t know!...  I will be contacting you privately for a test account.

 

CS

 

>>>

Why LOL?  We can offer termination to Spain (and most of the rest of the world), with CLI, billed 1/1. 

--Dave

 

 

From: asterisk-biz-bounces@lists.digium.com [mailto:asterisk-biz-bounces@lists.digium.com] On Behalf Of David Knell
Sent: Thursday, May 28, 2009 12:08 PM
To: Commercial and Business-Oriented Asterisk Discussion
Subject: Re: [asterisk-biz] Carrier to spain

----- Original Message -----

Sent: Thursday, May 28, 2009 4:22 PM

Subject: Re: [asterisk-biz] Carrier to spain

 

1/1 ? lol

 

From: asterisk-biz-bounces@lists.digium.com [mailto:asterisk-biz-bounces@lists.digium.com] On Behalf Of Carlos Rojas
Sent: May-28-09 10:23 AM
To: asterisk-biz@lists.digium.com
Subject: [asterisk-biz] Carrier to spain

 

Hello,

we are, a callcenter, and we need termination to spain, with callerid in spain, bill by sec.

Best regards


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Re: [asterisk-biz] Carrier to spain

Why LOL?  We can offer termination to Spain (and most of the rest of the world), with CLI, billed 1/1.
 
--Dave
----- Original Message -----
Sent: Thursday, May 28, 2009 4:22 PM
Subject: Re: [asterisk-biz] Carrier to spain

1/1 ? lol

 

From: asterisk-biz-bounces@lists.digium.com [mailto:asterisk-biz-bounces@lists.digium.com] On Behalf Of Carlos Rojas
Sent: May-28-09 10:23 AM
To: asterisk-biz@lists.digium.com
Subject: [asterisk-biz] Carrier to spain

 

Hello,

we are, a callcenter, and we need termination to spain, with callerid in spain, bill by sec.

Best regards


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Re: [asterisk-biz] Carrier to spain

1/1 ? lol

 

From: asterisk-biz-bounces@lists.digium.com [mailto:asterisk-biz-bounces@lists.digium.com] On Behalf Of Carlos Rojas
Sent: May-28-09 10:23 AM
To: asterisk-biz@lists.digium.com
Subject: [asterisk-biz] Carrier to spain

 

Hello,

we are, a callcenter, and we need termination to spain, with callerid in spain, bill by sec.

Best regards

[asterisk-biz] Carrier to spain

Hello,

we are, a callcenter, and we need termination to spain, with callerid in spain, bill by sec.

Best regards

[asterisk-biz] Free Toll Free Calling

Carrierx.us a Super Technologies Inc service now offers Free calling to any us and Canada
toll free number.

You can make this call from your asterisk server or via the web on

http://carrierx.us/tollfree/

Rehan


Rehan Ahmed AllahWala
President & CEO - Super Technologies Inc.

http://www.supertec.com/ - Internet Telephony Solutions

Don't Remember Me ? Visit http://www.Rehan.com

~~~~~~~~~~~~~~~~~~~
"First they ignore you, then they laugh at you, then they fight you, then you win."
By Gandhi.


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Wednesday, May 27, 2009

Re: [asterisk-biz] Colorado XO Customers

What "circuit?"

* Direct Internet Access (IP)?
- What kind?

* ISDN PRI?
- Origination? Termination? Both?

* Other?
- What?

-- alex

Scott Lewis wrote:

> Sorry if this is a bit off topic but you guys are my best list. I am
> looking for anyone with an XO circuit in Colorado's 303 area code. I
> have a pretty simple request that can make you some money. Please email
> or call me and I can go further into details. I assure you that this
> isn't a scam, I really just need a little help and I am willing to pay.
>
>
>
> Thank you,
>
> -Scott
>
> scottlewis01 (at) Gmail (dot) com
>
> 720.982.6670
>
>
> ------------------------------------------------------------------------
>
> _______________________________________________
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>
> asterisk-biz mailing list
> To UNSUBSCRIBE or update options visit:
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--
Alex Balashov
Evariste Systems
Web : http://www.evaristesys.com/
Tel : (+1) (678) 954-0670
Direct : (+1) (678) 954-0671
Mobile : (+1) (678) 237-1775

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[asterisk-biz] Colorado XO Customers

Sorry if this is a bit off topic but you guys are my best list.  I am looking for anyone with an XO circuit in Colorado’s 303 area code.  I have a pretty simple request that can make you some money.  Please email or call me and I can go further into details.  I assure you that this isn’t a scam, I really just need a little help and I am willing to pay.

 

Thank you,

-Scott

scottlewis01 (at) Gmail (dot) com

720.982.6670

Tuesday, May 26, 2009

[asterisk-biz] Looking for Philippines DIDs

Flat rate monthly DIDs, with two channels for major cities. We will need one for testing and once satisfied put an order for 30-40. Thanks.

Re: [asterisk-biz] UK SIP trunking

Hi,

This is standard and very simple service to set by Operators in UK who are
trustworthy not to abuse "CLI Manipulation".

It is normal for these services in application like "Virtual office" both
voice and fax where return path lands into extension of caller in
"Asterisk world"

Provider will need to convince proper Carriers that they do not wish to
run " Missing Calls Application" where gullible mobile callers return
calls to missed calls which generate revenue to the called party.

Cheers,

Mo


http://www.telpoint.co.uk

> We are looking into the possibilities of hooking our asterisk system
> into a SIP trunk for inbound and outbound calls in the UK.
>
> One of the requirements would be to set our caller id to any number used
> by us (including 0845 / 0800)
>
> We run up to 50 simultaneous external calls, and are currently using 1.4
> of asterisk.
>
> Does anyone know of such a provider in the UK ? Has anyone had any
> dealings with Spitfire
> (http://www.spitfire.co.uk/SIP_Trunking_tel.shtml?headerbar=1)
>
> Thanks!
>
> Julian
>
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>


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[asterisk-biz] Large Inventory Refurbished VoIP Hardware: Audiocodes, Aastra, Cisco, Linksys, Polycom, More

I have the following inventory of professionally refurbished VoIP hardware available for immediate sale.  I have quantities ranging from 1-75 on all of the items below.  If you see anything of interest please shoot me an email.  Equipment is all cleaned, tested, reset to factory default with the very latest firmware installed.  Individually bulk boxed with cables and accessories.  6 month warranty on all.  Very competitive pricing.

 

02-108002 Aastra  480i (VSRF)

Aastra 57i A1757-0131-10-01 (VSRF)

Aastra 57i-CT (VSRF)

Aastra 9112i (VSRF)

Aastra 9133i (VSRF)

Astra 560M A1760-0000-10-55 (VSRF)

Audiocodes MP104-FXS (VSRF)

Audiocodes MP114-FXO (VSRF)

Audiocodes MP118-FXO (VSRF)

Cisco CP-7906G (VSRF)

Cisco CP-7912G (VSRF)

Cisco CP-7914 (VSRF)

Cisco CP-7931G (VSRF)

Cisco CP-7936 (VSRF)

Cisco CP-7941G (VSRF)

Cisco CP-7970G (VSRF)

Cisco CP-7971G-GE (VSRF)

Cisco CP-7975G (VSRF)

Cisco CP-DOUBLFOOTSTAND (VSRF)

Cisco Systems CP-7940G (VSRF)

Cisco Systems CP-7960 (VSRF)

Cisco Systems CP-7960G (VSRF)

Cisco Systems CP-7961G (VSRF)

Grandstream BT102 (VSRF)

Grandstream BT201 (VSRF)

Grandstream GXP2000 (VSRF)

Grandstream GXW4108 (VSRF)

Hitachi-Cable IP3000 (VSRF)

Linksys  SD205 (VSRF)

Linksys  SPA8000-G1 (VSRF)

Linksys  SPA9000-VSRF (VSRF)

Linksys SPA-2102 (VSRF)

Linksys SPA922 (VSRF)

Linksys SPA932 (VSRF)

Linksys SPA941 (VSRF)

Linksys SPA942 (VSRF)

Linksys WIP330-NA (VSRF)

Linksys WRTP54G-V (VSRF)

Linksys WVC54GC (VSRF)

Mediatrix 2102 (VSRF)

Polycom IP301 2200-11331-001 (VSRF)

Polycom IP320 2200-12320-025 (VSRF)

Polycom IP330 2200-12330-025 (VSRF)

Polycom IP4000 2200-06640-001 (VSRF)

Polycom IP430 2200-12430-001 (VSRF)

Polycom IP501 2200-11531-001 (VSRF)

Polycom IP501 POE 2200-11531-025 (VSRF)

Polycom IP560 (VSRF)

Polycom IP6000 2200-15660-001 (VSRF)

Polycom IP601 2200-11631-001 (VSRF)

Polycom IP601 SIDECAR 2200-11700-025 (VSRF)

QuickPhones QA-342 (VSRF)

RedFone FoneBridge 2 Single (VSRF)

Snom 300 (VSRF)

Snom 360 (VSRF)

UTStarcom WiFi Phone F3000-VSRF (VSRF)

 

 

Cory J. Andrews

Director New Market Initiatives

 

Sayers Media Group

VoIP Supply, LLC

454 Sonwil Drive

Buffalo, NY 14225

716-250-3402 OFFICE

716-630-1548 FAX

716-601-4474 MOBILE

candrews@sayersmedia.com

 

 

Have I exceeded your expectations?  Please share your experience with my boss,  Benjamin P. Sayers, CEO

 

NOTICE: The information contained in this email and any document attached hereto is intended only for the named recipient(s). It is the property of the VoIP Supply, LLC and shall not be used, disclosed or reproduced without the express written consent of VoIP Supply, LLC. If you are not the intended recipient, nor the employee or agent responsible for delivering this message in confidence to the intended recipient(s), you are hereby notified that you have received this transmittal in error, and any review, dissemination, distribution or copying of this transmittal or its attachments is strictly prohibited. If you have received this transmittal and/or attachments in error, please notify me immediately by reply e-mail or telephone and then delete this message, including any attachments. Our mailing address is 454 Sonwil Drive, Buffalo, NY 14225 USA.

 

 

[asterisk-biz] RĂ©ponse en cas d'absence

    Heya,how are you doing recently ? I would like to introduce you a very good company which i knew.Their website is  www.bnwtorder.com  .They can offer you all kinds of electronical products which you need like laptops ,gps ,TV LCD,cell phones,ps3,MP3/4,motorcycles  etc........Please take some time to have a check ,there must be somethings you 'd like to purchase .
Their contact email:
bnwtorder@bnwtorder.com
.  MSN: bnwtorder@hotmail.com 
Hope you have a good mood in shopping from their company !
Regards

[asterisk-biz] Suggest good calling service for London

Hello All,
 
We are setting up call center of 10 agents and expecting its growth till 30 agents. Mainly calling is within UK. Please suggest some good service for UK dialing with London DID.
 
Regards,
Kashif Naeem
Business Development Manager
Hadi Telecom
www.haditelecom.com

Cell: +92 (0)345 4226006
Office: +92 (0)42 5692766
           
Email: kashif@haditelecom.com
MSN: kashif__naeem@hotmail.com
Gmail: meet.kashif@gmail.com
Skype: kashif.naeem

302 Y Commercial Area, 2nd Floor DHA Lahore, Pakistan.

Re: [asterisk-biz] UK SIP trunking

Hi,

I would recommend that you review Magrathea's offering.
http://www.magrathea-telecom.co.uk/


Regards
Magnus

> -----Original Message-----
> From: asterisk-biz-bounces@lists.digium.com [mailto:asterisk-biz-
> bounces@lists.digium.com] On Behalf Of Julian Lyndon-Smith
> Sent: 26 May 2009 15:00
> To: asterisk-biz@lists.digium.com
> Subject: [asterisk-biz] UK SIP trunking
>
> We are looking into the possibilities of hooking our asterisk system
> into a SIP trunk for inbound and outbound calls in the UK.
>
> One of the requirements would be to set our caller id to any number
> used
> by us (including 0845 / 0800)
>
> We run up to 50 simultaneous external calls, and are currently using
> 1.4
> of asterisk.
>
> Does anyone know of such a provider in the UK ? Has anyone had any
> dealings with Spitfire
> (http://www.spitfire.co.uk/SIP_Trunking_tel.shtml?headerbar=1)
>
> Thanks!
>
> Julian
>
> _______________________________________________
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> To UNSUBSCRIBE or update options visit:
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[asterisk-biz] UK SIP trunking

We are looking into the possibilities of hooking our asterisk system
into a SIP trunk for inbound and outbound calls in the UK.

One of the requirements would be to set our caller id to any number used
by us (including 0845 / 0800)

We run up to 50 simultaneous external calls, and are currently using 1.4
of asterisk.

Does anyone know of such a provider in the UK ? Has anyone had any
dealings with Spitfire
(http://www.spitfire.co.uk/SIP_Trunking_tel.shtml?headerbar=1)

Thanks!

Julian

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[asterisk-biz] Web design consultant wanted

Integrics is looking for a web design consultant for a short term
contract, with the possibility of future work. The following skills will
be required at expert level:

- Web site artistic design. This is the primary requirement.
- HTML.
- CSS.

The following skills will be required at a moderate level:

- Perl's HTML::Template library.
- Use of subversion for source control.
- General Linux.

Applicants must have experience in a corporate environment.
Qualifications in art or design are desirable. I'm aware of the low-cost
bidding websites for such services, but am looking for someone at a
higher level than these. Samples of previous work will be required.

If you feel you fit the above requirements, or know someone who does,
please submit a CV, samples of previous work (URLs are fine) along with
details of exactly what your contribution to the sample was, and rates
via email. Daily rates are preferred but I'm open to other options.

--
Alistair Cunningham
+1 888 468 3111
+44 20 799 39 799
http://integrics.com/

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Sunday, May 24, 2009

[asterisk-biz] FS - 2 Sangoma A102 $650 each

Hi everyone,

I have two Sangoma A102 cards for sale. Each for $650. They are used but in good working condition.

Thanks!

Jared

Saturday, May 23, 2009

[asterisk-biz] FS: 9 UNITS New D-link DVG-1402S voip routers (2FXS 4 LAN, factory unlocked).

I have 9 units D-link DVG-1402S for sale.All are brand new , factory
unlocked and factory sealed. $350 CAD shipping included. All units
comes with 110v-240v universal power supply.

http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItem&item=270393860731

http://cgi.ebay.com/ws/eBayISAPI.dll?ViewItem&item=260413770551

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Re: [asterisk-biz] Setting up Asterisk conf room per channel on DID

Sandeep Kanao wrote:
> Hello,
> I am looking for a few pointers to setup asterisk conf room.
Sure thing!
> User A calls a local DID #, User B calls a local DID # and they are in
> the conf room. Our system knows caller IDs for both of these users
> (fixed). Assuing these DID's have 2 channels each, could it be
> possible to use both the channels on these DIDs, so User C and User D
> could be in another conf room with the same DID's but using
> the remaining channel?
Yep. You'd need an AGI that checks a database of knows the Caller
IDs and what conferences they go to. Then from there, the system just
routes incoming calls into the correct conferences. I've helped a
customer do something similar in the past for a group support line.
> Any pointers, any feedback is welcome.
> Thanks
My pleasure. Feel free to ask anything else you need.

--Michel


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[asterisk-biz] Setting up Asterisk conf room per channel on DID

Hello,
I am looking for a few pointers to setup asterisk conf room.
User A calls a local DID #, User B calls a local DID # and they are in the conf room. Our system knows caller IDs for both of these users (fixed). Assuing these DID's have 2 channels each, could it be possible to use both the channels on these DIDs, so User C and User D could be in another conf room with the same DID's but using the remaining channel?
Any pointers, any feedback is welcome.
Thanks

[asterisk-biz] hey

Dear friend,
Sorry to disturb you because of this message! I would like to
introduce a good company who trades mainly in electornic products.
Our company mainly in electronic products: laptops, cameras, televisions,
mobile phones, radio stations, and so on. It is facing to wholesalers,retailers,
and personal customer all over the world. Original  Products + Best Quality + Brand
New + Warranty + Quick Shipping + 100% Secure = 100% satisfied.
For more, please visit (www.)Vincentwm.com
 


MSN NZ Travel Find a way to cure that travel bug

[asterisk-biz] encrypted pstn calls

I have written some windows mobile software (used on my HTC phone) which
will encrypt phone calls (the underlying mobile network should not
matter, but its known to work with GSM).

I do not know if anyone is interested in this, or more specifically in
supporting additional platforms. For example, an iphone port, an
android port, a server side module, or something else

The software lets you make gsm<->gsm calls and send text messages
encrypted. In addition if I write server side software, you can do this
to gsm<->[asterisk|freeswitch|maybe something else]. This way a call
to/from a mobile would be encrypted, and could be encrypted via SRTP/TLS
to the handset itself.

The use of crypto is optional on a per call basis, which means that you
can still send/receive calls from unencrypted phones.

This might be ideal for those who need HIPPA compliance for medical
information, or for people in the financial community, or just for
people who want to keep what is said private from prying ears (its not
that difficult to monitor GSM calls, there is even an active multi-year
old project using gnu radio and other tools to do just that, A5 the GSM
cipher has many weaknesses, and there are even rainbow tables making it
as simple as a lookup for the key).

If anyone is interested in sponsoring this software email me privately
and we can discuss this further. I am open to licensing this for resale
as well.


--
Trixter http://www.0xdecafbad.com Bret McDanel
pgp key: http://pgp.mit.edu:11371/pks/lookup?op=get&search=0x8AE5C721

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Re: [asterisk-biz] Toll Free Toll Free tollfreetollfree.com SIP gateway now supports ZRTP in addition to SIP/TLS, SIP/TCP, SIP/UDP, G729

On Sat, 2009-05-23 at 09:19 -0400, SIP wrote:
> It's also too bad that once your phone call heads to the PSTN, any form
> of encryption becomes functionally worthless. It's INCREDIBLY easy to
> tap into, trace, and catalogue a PSTN phone call.
>

there is nothing that can be done about that, unless the answering end
supports crypto too. I still think its easier to monitor VoIP calls
since that can be done from almost anywhere on the intarweb (although do
it in the wrong spot and it becomes quite noticable). PSTN *generally*
requires physical access limiting the ones who can do this somewhat.

> Still... is very nice to see someone providing ZRTP services. Would like
> to see more of that in the future.

yeah, and to be crypto agnostic like that, both SRTP/TLS and ZRTP is a
nice bonus. Ensures that more people can use it, although the way that
ZRTP works, it becomes harder to validate the cipher since you cant
compare the codes each end provides. If you use a mechanical voice or a
sip im it becomes trivial to spoof the challenge/response codes, it
relies on humans speaking and listening to share the codes to validate.

Because you cant validate the cipher I cannot say that you can trust
ZRTP in this implementation, but then it was not designed to be on a
server side, this is where TLS can be handy. However SRTP/TLS is
designed to be server side but not as dynamic nor forget everything
about the session making key recovery impossible (the cert is still
there), something ZRTP is designed for.

ZRTP is more of an end to end human to human implementation, so for
things where the server has to be in the middle SRTP/TLS is certainly
superior.

--
Trixter http://www.0xdecafbad.com Bret McDanel
pgp key: http://pgp.mit.edu:11371/pks/lookup?op=get&search=0x8AE5C721

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Re: [asterisk-biz] Toll Free Toll Free tollfreetollfree.com SIP gateway now supports ZRTP in addition to SIP/TLS, SIP/TCP, SIP/UDP, G729

Trixter aka Bret McDanel wrote:
> On Fri, 2009-05-22 at 19:08 -0400, Jason Garland wrote:
>
>> Toll Free Toll Free tollfreetollfree.com SIP gateway now supports ZRTP
>> in addition to SIP/TLS, SIP/TCP, SIP/UDP, G729
>>
>>
>
> too bad TLS requires TCP and asterisk default doesnt do that, although
> rumors are that there have been SRTP/TLS patches for quite some time,
> they just never made it into trunk. Its nice to see that people are
> starting to engage in crypto for calls though.
>
>
>
It's also too bad that once your phone call heads to the PSTN, any form
of encryption becomes functionally worthless. It's INCREDIBLY easy to
tap into, trace, and catalogue a PSTN phone call.

Still... is very nice to see someone providing ZRTP services. Would like
to see more of that in the future.

N.

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Friday, May 22, 2009

Re: [asterisk-biz] Toll Free Toll Free tollfreetollfree.com SIP gateway now supports ZRTP in addition to SIP/TLS, SIP/TCP, SIP/UDP, G729

On Fri, 2009-05-22 at 19:08 -0400, Jason Garland wrote:
> Toll Free Toll Free tollfreetollfree.com SIP gateway now supports ZRTP
> in addition to SIP/TLS, SIP/TCP, SIP/UDP, G729
>

too bad TLS requires TCP and asterisk default doesnt do that, although
rumors are that there have been SRTP/TLS patches for quite some time,
they just never made it into trunk. Its nice to see that people are
starting to engage in crypto for calls though.


--
Trixter http://www.0xdecafbad.com Bret McDanel
pgp key: http://pgp.mit.edu:11371/pks/lookup?op=get&search=0x8AE5C721

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[asterisk-biz] Toll Free Toll Free tollfreetollfree.com SIP gateway now supports ZRTP in addition to SIP/TLS, SIP/TCP, SIP/UDP, G729

Toll Free Toll Free tollfreetollfree.com SIP gateway now supports ZRTP
in addition to SIP/TLS, SIP/TCP, SIP/UDP, G729

Send your Toll Free SIP calls to tollfreetollfree.com

For ZRTP calls send them to zrtp.tollfreetollfree.com

For more information on zrtp see: http://zfoneproject.com/

If your software doesn't support ZRTP you can install zfone, and it
will actively convert your RTP stream into an encrypted SRTP
stream(Secure encrypted phone calls)

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Re: [asterisk-biz] 1-800 port

On 5/22/2009 19:15, Nitzan Kon wrote:
> If your number is with J2 you can pretty much forget about
> it. See: http://www.dslreports.com/forum/r22154400-Avoid-Onebox-and-j2-at-all-costs
>
> It is sad that they can get away with this crap.

This pretty much explains WHY my LoA has been rejected with a lame
excuse... And maybe why they are sending me to talk to another provider
back and forth.

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Re: [asterisk-biz] 1-800 port

Please send me an email at lpritchard@teliax.com so that I can give you additional information.
 
Kind regards,
 
Lynda


 
On Fri, May 22, 2009 at 3:57 PM, Hermann Wecke <hermann@wecke.com> wrote:
On 5/22/2009 16:02, Lynda Pritchard wrote:
> Did you buy the number from Asterlink? If so, it may be listed under
> their name.

Thanks. The original number is from AT&T, ported 4 years ago into Nufone
and them ported into Asterlink - when NuFone went out of business for
the first time.

> It is still registered under the resporg jft01. Which is a company
> called Timeshift. I am listing the contact information so that you
> can call and find out how to get the number ported.
>
> First Name :  Susan
> Last Name:    Isherwood
> Phone:        323-860-9227
>
> I hope this helps........

The number listed for resporg jft01 is ringing but not answering. Susan
used to work for J2 Global Communications. This number is like a "fools
number", used to receive calls that will never be answered.

I was able to call J2 main office and I was then told to talk to
Airspring @ 888-899-2789, who is now the company responsible for my
number. After calling them, they told me they are not the provider, and
transfered me back to J2.

I guess my number is M.I.A....


> On Fri, May 22, 2009 at 12:25 PM, Hermann Wecke <hermann@wecke.com
> <mailto:hermann@wecke.com>> wrote:
>
>     On 4/29/2009 15:38, Hermann Wecke wrote:
>      > I need to transfer my toll free number from now defunct Asterlink.
>
>     As Asterlink is now dead (telephone 877-278-7565 is in auto-answer, no
>     email reply), I tried to move my number away from them.
>
>     The request has been rejected with an "invalid signature" excuse.
>
>     How can I retrieve my number? As the company is dead/no longer in
>     business, who should I contact?


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Re: [asterisk-biz] 1-800 port

If your number is with J2 you can pretty much forget about
it. See: http://www.dslreports.com/forum/r22154400-Avoid-Onebox-and-j2-at-all-costs

It is sad that they can get away with this crap.


--- On Fri, 5/22/09, Hermann Wecke <hermann@wecke.com> wrote:

> From: Hermann Wecke <hermann@wecke.com>
> Subject: Re: [asterisk-biz] 1-800 port
> To: "Commercial and Business-Oriented Asterisk Discussion" <asterisk-biz@lists.digium.com>
> Date: Friday, May 22, 2009, 5:57 PM
> On 5/22/2009 16:02, Lynda Pritchard
> wrote:
> > Did you buy the number from Asterlink? If so, it may
> be listed under
> > their name.
>
> Thanks. The original number is from AT&T, ported 4
> years ago into Nufone
> and them ported into Asterlink - when NuFone went out of
> business for
> the first time.
>
> > It is still registered under the resporg jft01. Which
> is a company
> > called Timeshift. I am listing the contact information
> so that you
> > can call and find out how to get the number ported.
> >
> > First Name :     Susan
> > Last Name:     Isherwood
> > Phone:     323-860-9227
> >
> > I hope this helps........
>
> The number listed for resporg jft01 is ringing but not
> answering. Susan
> used to work for J2 Global Communications. This number is
> like a "fools
> number", used to receive calls that will never be
> answered.
>
> I was able to call J2 main office and I was then told to
> talk to
> Airspring @ 888-899-2789, who is now the company
> responsible for my
> number. After calling them, they told me they are not the
> provider, and
> transfered me back to J2.
>
> I guess my number is M.I.A....
>
>
> > On Fri, May 22, 2009 at 12:25 PM, Hermann Wecke <hermann@wecke.com
>
> > <mailto:hermann@wecke.com>>
> wrote:
> >
> >     On 4/29/2009 15:38, Hermann
> Wecke wrote:
> >      > I need to transfer my toll
> free number from now defunct Asterlink.
> >
> >     As Asterlink is now dead
> (telephone 877-278-7565 is in auto-answer, no
> >     email reply), I tried to move
> my number away from them.
> >
> >     The request has been rejected
> with an "invalid signature" excuse.
> >
> >     How can I retrieve my number?
> As the company is dead/no longer in
> >     business, who should I
> contact?
>
>
> _______________________________________________
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>
> asterisk-biz mailing list
> To UNSUBSCRIBE or update options visit:
>    http://lists.digium.com/mailman/listinfo/asterisk-biz
>

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[asterisk-biz] [VIDEO] High Volume US Traffic? Claim DIP CNAM Compensation

[VIDEO] High Volume US SIP VoIP Traffic? Claim DIP CNAM Compensation
http://www.youtube.com/watch?hl=en&v=y3v-iwlgszU

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Re: [asterisk-biz] 1-800 port

On 5/22/2009 16:02, Lynda Pritchard wrote:
> Did you buy the number from Asterlink? If so, it may be listed under
> their name.

Thanks. The original number is from AT&T, ported 4 years ago into Nufone
and them ported into Asterlink - when NuFone went out of business for
the first time.

> It is still registered under the resporg jft01. Which is a company
> called Timeshift. I am listing the contact information so that you
> can call and find out how to get the number ported.
>
> First Name : Susan
> Last Name: Isherwood
> Phone: 323-860-9227
>
> I hope this helps........

The number listed for resporg jft01 is ringing but not answering. Susan
used to work for J2 Global Communications. This number is like a "fools
number", used to receive calls that will never be answered.

I was able to call J2 main office and I was then told to talk to
Airspring @ 888-899-2789, who is now the company responsible for my
number. After calling them, they told me they are not the provider, and
transfered me back to J2.

I guess my number is M.I.A....


> On Fri, May 22, 2009 at 12:25 PM, Hermann Wecke <hermann@wecke.com
> <mailto:hermann@wecke.com>> wrote:
>
> On 4/29/2009 15:38, Hermann Wecke wrote:
> > I need to transfer my toll free number from now defunct Asterlink.
>
> As Asterlink is now dead (telephone 877-278-7565 is in auto-answer, no
> email reply), I tried to move my number away from them.
>
> The request has been rejected with an "invalid signature" excuse.
>
> How can I retrieve my number? As the company is dead/no longer in
> business, who should I contact?


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Re: [asterisk-biz] 1-800 port

I think you could/should of sent private information on a legit client via private email ;)

 

 

From: asterisk-biz-bounces@lists.digium.com [mailto:asterisk-biz-bounces@lists.digium.com] On Behalf Of Lynda Pritchard
Sent: May-22-09 3:02 PM
To: Commercial and Business-Oriented Asterisk Discussion
Subject: Re: [asterisk-biz] 1-800 port

 

Did you buy the number from Asterlink? If so, it may be listed under their name. It is still registered under the resporg jft01. Which is a company called Timeshift. I am listing the contact information so that you can call and find out how to get the number ported.

 

First Name :

Susan

Last Name:

Isherwood

City:

Hollywood

State/Province:

California

Zip:

90028

Country:

USA

Phone:

323-860-9227

I hope this helps........

 

Kind regards,

 

Lynda

 

On Fri, May 22, 2009 at 12:25 PM, Hermann Wecke <hermann@wecke.com> wrote:

On 4/29/2009 15:38, Hermann Wecke wrote:
> I need to transfer my toll free number from now defunct Asterlink.

As Asterlink is now dead (telephone 877-278-7565 is in auto-answer, no
email reply), I tried to move my number away from them.

The request has been rejected with an "invalid signature" excuse.

How can I retrieve my number? As the company is dead/no longer in
business, who should I contact?

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  http://lists.digium.com/mailman/listinfo/asterisk-biz