[asterisk-biz] Egypt

[Matt Riddell (lists)]

Hi all,

I’m looking to speak with someone who has an understanding of telephony 
infrastructure in Egypt. 

This is for a potential project for COVID 19. 

If you can help, please email me directly. 

Sent from my iPhone

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[asterisk-biz] Asterisk Usage Survey

[Matt Fredrickson]

Hey All,

For those of you that do not know me, my name is Matthew Fredrickson
and I'm the project lead for the Asterisk project.  First off, I
wanted to thank all of you that contribute in various ways to the
project - whether it be at a developmental level, answering questions
on forums and mailing lists, contributing documentation, or just
generally advocating for it within your sphere of influence.  It takes
so many people's efforts to make the project what it is and to sustain
such a large and vibrant user and developer community.

We created a general survey inquiring how people utilize Asterisk.  It
should only take about 10-15 minutes, but would help us understand
better how our users are utilizing Asterisk and help us to understand
if there are important areas of Asterisk that we underemphasize from a
development perspective.  If you don't mind filling it out, it would
be greatly appreciated.

Thanks *so* much again for your time, and best wishes to each of you
in your efforts.

https://goo.gl/forms/xL1VUHRsf95saly13

-- 
Matthew Fredrickson
Digium - A Sangoma Company | Asterisk Project Lead
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA

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[asterisk-biz] Asterisk Training Manager

[Matt Fredrickson]

Hey All,

We're in the market trying to find someone to manage the Asterisk
training program at Sangoma/Digium.  I thought it wouldn't be
inappropriate to post something here.

You can find information about the position at:
https://www.sangoma.com/company/careers/

You may now return to your regularly scheduled programming.

Best wishes!

-- 
Matthew Fredrickson
Digium - A Sangoma Company | Asterisk Project Lead
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA

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[asterisk-biz] Sangoma Announcement

[Matt Fredrickson]

Greetings!

For those of you who have not noticed, it was just announced that
Sangoma is intending to acquire Digium.  Since discussion about this
will likely come up, I figured that it would be good to post something
here first.

For questions about the acquisition and how it impacts Asterisk and
FreePBX, feel free to see the FAQ on the Asterisk wiki at [1].  For
other questions (particularly those related to products and respective
businesses), feel free to check Digium or Sangoma’s website or contact
Sangoma or Digium directly.

https://wiki.asterisk.org/wiki/display/AST/Sangoma+and+Digium+Join+Together+FAQ

I'm personally looking forward to seeing what all the great talent at
both companies can do working together.

Best wishes!

-- 
Matthew Fredrickson
Digium, Inc. | Asterisk Project Lead
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA

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[asterisk-biz] Open Position on Asterisk Team

[Matt Fredrickson]

Hey All,

I remember back in 2001 when Mark Spencer hired me at Digium, he asked
me, "Isn't it cool that you actually get paid to work on open source
software?"  Back then, it was a lot harder to find employment doing
open source software development but I still marvel that 16 years down
the road I'm still able to do so and that I get to invite someone else
to as well.

There is an open position on Digium's Asterisk development team for an
Asterisk Developer.  This is primarily a C and Python development
position in which you get the opportunity to work on the engine that
drives a large portion of the open source telephony world.  You get to
make an impact on a codebase that is used by thousands of people all
over the world.  How awesome of that?

You can find more information at the link below:
https://recruitingapp-7001.umantis.com/Vacancies/488/Description/2

If you have any questions, feel free to email me.

-- 
Matthew Fredrickson
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA

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Re: [asterisk-biz] Kamailio World Conference 2017, May 8-10, in Berlin, Germany

[Matt Fredrickson]

I know this is a bit late to the party, but I went to KamailioWorld
last year and can't understate what a high quality telecom conference
it is (particularly if you're focused on the carrier/infrastructure
side of things).  Also, I'll be talking there as well, so this is a
shameless self promotion.

Hope to see many of you there!

Matthew Fredrickson

On Mon, Mar 20, 2017 at 6:18 AM, Daniel-Constantin Mierla
 wrote:
> I would like to inform that the next Kamailio World Conference & Exhibition
> is taking place again in the city center of Berlin, Germany, during May
> 8-10, 2017. More details can be found at:
>
>   - https://www.kamailioworld.com
>
> It is two days and a half event with presentations and workshops about Real
> Time Communications, spanning from classic SIP/VoIP to WebRTC and IoT.
> Besides Kamailio, the Asterisk project has again a consistent presence at
> the event. The details for most of the speakers and presentations are
> available at:
>
>   - https://www.kamailioworld.com/k05/speakers/
>   - https://www.kamailioworld.com/k05/schedule/
>
> The registration is open, the venue has a limited number of seats, hurry up
> if you plan to attend the event:
>
>   - https://www.kamailioworld.com/k05/registration/
>
> All the best!
>
> --
> Daniel-Constantin Mierla - http://www.asipto.com
> http://twitter.com/#!/miconda - http://www.linkedin.com/in/miconda
>
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-- 
Matthew Fredrickson
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA

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Re: [asterisk-biz] Digium Announced the Call for Speakers is Now Open for AstriCon 2017

[Matt Fredrickson]

Good catch, and thanks for posting that Eric!

Matthew Fredrickson

On Wed, Feb 15, 2017 at 2:55 AM, Eric Klein
 wrote:
> Sharing this as it apparently has not made it to the lists yet. Last week
> Digium Announced the Call for Speakers is Now Open for AstriCon 2017
>
> The 14th annual Asterisk user conference and expo to be held Oct. 3-5 at the
> Omni Orlando Resort at ChampionsGate in Orlando, Fla.
>
> Digium blog post:
> http://blogs.digium.com/2017/02/02/astricon-2017-october-orlando/
> Announcement PR:
> http://www.prnewswire.com/news-releases/digium-announces-call-for-speakers-now-open-for-astricon-2017-300401428.html
>
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-- 
Matthew Fredrickson
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA

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[asterisk-biz] Customer needs US DIDs

[Matt Riddell]

1. Please reply off list

2. Customer needs around 400 DIDs sent to a SIP address.

3. Inbound volume to these DIDs will be high but they only really need 30 
concurrent channels

4. Load will be spread over around 40 DIDs per block with the block changing 
every few days

5. Cost in not the most important factor.

6. Email me off list for further details.

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Re: [asterisk-biz] IP KEY SYSTEMS Launch at ITEXPO

[Matt Brei]

Nice presentation but you may consider speeding up the text a little bit.
It feels like it drags on.  Some interface screenshots of the setup and in
production would be nice too.

-- 
Matt Brei
Network Security/IP Telephony Engineer
mattb...@gmail.com

On Tue, Jan 26, 2016 at 10:01 PM, Stephen Wingfield <st...@bicom.us> wrote:

> If you been looking for a solution to provide to traditional users of Key
> Systems, please see:
>
>
> https://drive.google.com/file/d/0BwuXJaX8xMozdlkyZmR0UlNvYzg/view?usp=sharing
>
> If you are are ITEXPO, please call, booth 323.
>
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Re: [asterisk-biz] Opensips-solutions

[Matt Riddell]

> On 22Nov, 2015, at 10:40, Talking Voice <talkingvo...@gmail.com> wrote:
> 
> Is anyone on this list using this product 
> Opensips-solutions(http://www.opensips-solutions.com/ 
> <http://www.opensips-solutions.com/>) ?
> I am trying to obtain information from them but so far not been successful. I 
> have made several attempts from their website.

I know this is a bit late but did you manage to get hold of them ok?

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Re: [asterisk-biz] Free US DIDs to give-away

[Matt Brei]

May I have a few of these if there's still some left please?

Thank You

-- 
Matt Brei
Network Security/IP Telephony Engineer
mattb...@gmail.com
Phone: 708-737-4930

On Thu, Oct 29, 2015 at 1:56 PM, Markus <unive...@truemetal.org> wrote:

> Hi list,
>
> we have a couple hundreds of US DIDs in NY and LA to give-away for free
> due to an over-commitment. The available capacity is about 800 concurrent
> calls.
>
> Regards
> Markus
>
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Re: [asterisk-biz] Hosted TTS

[Matt Riddell]

 On 2Jul, 2015, at 06:22, David Knell david.kn...@telng.com wrote:
 
 Hi -
 
 We've been putting together a hosted speech service, and we're ready to make
 the first element available.  It's a hosted TTS service for Asterisk users, 
 provided
 via an AGI script.


Pricing?

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Re: [asterisk-biz] US$500 Bounty to fix bug 20402

[Matt Riddell]

On 15/09/2012, at 2:00 AM, David Knell d...@3c.co.uk wrote:
 Hi Matt -
  
 I'll take a look at this if you've no takers so far..
  

Already sorted - looks like a documentation change.

For the list - agents.conf defaults to using * to disconnect a call (as does 
features.conf).

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Re: [asterisk-biz] US$500 Bounty to fix bug 20402

[Matt Riddell (lists)]

Yeah will do. 

Sent from my iPhone

On 15/09/2012, at 5:46 PM, Paul Belanger paul.belan...@polybeacon.com wrote:

 On 12-09-15 11:51 AM, Matt Riddell wrote:
 On 15/09/2012, at 2:00 AM, David Knell d...@3c.co.uk wrote:
 Hi Matt -
 
 I'll take a look at this if you've no takers so far..
 
 Already sorted - looks like a documentation change.
 
 For the list - agents.conf defaults to using * to disconnect a call (as does 
 features.conf).
 Make sure you get a patch for the updated documentation in the issue, we can 
 then get it committed into subversion.
 
 -- 
 Paul Belanger | PolyBeacon, Inc.
 Jabber: paul.belan...@polybeacon.com | IRC: pabelanger (Freenode)
 Github: https://github.com/pabelanger | Twitter: 
 https://twitter.com/pabelanger
 
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Re: [asterisk-biz] Wi-fi SIP handset

[Matt Darnell]

On Sat, Oct 1, 2011 at 1:41 AM, toto tata toto_je...@yahoo.com wrote:

 Hi Matt,

 Please, can you post a link for the iPod Touch you are talking about? I'd
 like to see the picture and full specifications.

 Looking forward to hearing from you.
   Z T
  http://www.earnmailer.com/link/52616ecd7931085c
 http://www.earnmailer.com/alert/confirm/52616ecd7931085c


http://store.apple.com/us/browse/home/shop_ipod/family/ipod_touch?mco=MTM3NTEyOTc
Standard Apple iPod.

http://lifehacker.com/5636976/how-to-turn-your-ipod-touch-into-an-iphone-4g-edition
http://technina.com/voip/sip-client-for-iphone-ipod-to-make-cheap-voip-calls/


-Matt
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Re: [asterisk-biz] Wi-fi SIP handset

[Matt Darnell]

On Thu, Sep 29, 2011 at 2:01 PM, Alex Balashov
abalas...@evaristesys.com wrote:
 Any recommendations on a decent, but affordable wi-fi SIP handset?

 I've only seen some of the earlier-generation stuff from a few years ago
 from Linksys and was not impressed;  didn't seem very mature.  However, I'm
 sure they've evolved.

 Decent battery life (3-4 hours) would be one desirable criterion, but I
 suppose decent sound and microphone quality (including good sensitivity)
 would have to top the list.

 If anyone has any good experiences, please share!

Matt

It is hard to beat an iPod touch for $229.  Lots of free SIP phones for IOS.

-Matt

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[asterisk-biz] Inbound DDI Unlimited

[Matt Riddell]

Hi all,

Looking for a United Status number or numbers with unlimited or per 
channel prices for inbound calls - can you send me some quotes - we'd be 
looking at a reasonably high number of inbound channels (i.e. 50-100).


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Matt Riddell
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Re: [asterisk-biz] Rates Importer Tool

[Matt Riddell]

On 10/05/11 6:28 AM, A E [Gmail] wrote:

Hi All,

new to the list. Wondering if anyone has / knows of, a good rate
importer tool that can be used to standardize and normalize the
ratesheets / rate decks etc. obtained from various carriers so they can
be analysed and imported into a DB or be saved as a CSV or something?


We've done it, but just for our ITSP platform.

Even then, it still can require tweaking.

We provide an Excel import field.

The system imports the Excel file and shows the fields on screen for the 
first few rows.


It then asks you what field contains: Dest Name, Dest Number and Rate.

Once it's done that it asks you for any tweaking of the numbers required 
from E164 formatting (i.e. some providers expect you to dial 00XX where 
XX is the international code, others require 011XX etc etc.


It then asks you for an across the board markup on their rate and what 
the currency is.


There is an option to update exchange rates.

It's a pretty complicated system but works.

We maintain buy and sell rates and there is an option to convert between 
them when you import new buy rates - although the buy rate might change 
because of route issues, the sell rate always stays the same.  The sell 
rate is different for different rate plans.


I'm not sure how useful our code would be for anyone else as it is just 
formatting the data for use by our LCR + Weighting AGI scripts, but I'd 
be happy to provide anyone else with assistance if they're looking at 
writing one.


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Matt Riddell
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Re: [asterisk-biz] Rates Importer Tool

[Matt Riddell]

On 10/05/11 12:41 PM, Alex Balashov wrote:


At the risk of impetuousness, I think this is a somewhat misguided
endeavour, simply because the code/templates/scripts you're asking for
are inextricably found up in the specificity of a given ITSP
environment, vendor blend, billing engine, etc.


Yeah that's entirely correct in our case - made somewhat worse (for 
others) by the fact that we rely heavily on our in house function 
libraries for doing pretty much everything :-)


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Matt Riddell
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Re: [asterisk-biz] wholesale VoIP providers using Asterisk

[Matt Riddell]

On 5/05/11 8:17 AM, m...@leetop.net wrote:

So what ARE the reasons for the death throes then? And is there another
marketplace to find customers? In particular for relatively high-cost
(and therefore high-margin) routes outside the competitive US/EU locations?


Anybody who has ever posted a product here or information about their 
business has been blasted pretty harshly whether it be for their 
English, their newly created website, broken links etc.


The problem stems from the fact that everyone on this list has their own 
products and feel the need to lash out at other's products when they're 
posted.  People therefore scan through everything and every link posted 
trying to find something wrong.


Once they do they announce that fact to the world and ridicule the 
company making the announcement - making them feel more secure in their 
own product.


It's something I don't have a solution for.

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[asterisk-biz] Asterisk Support in Alicante, Spain

[Matt Riddell]

Hi all,

Just wondering if there is someone who does contract Asterisk support 
around the Alicante region of Spain?


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Re: [asterisk-biz] need ITSP recommendation

[Matt Purdy]

Hi Robert - We'd be happy to help you out with this. 

 

We have the ability to support your proposed call volume, (as well as
growth), a fair rate for orig/term, and reasonable porting charges for your
DIDs. 

 

Can we connect directly and discuss the details? 

 

Thanks in advance, and I hope to talk to you soon!

 

Cheers, 

-Matt

 

Matt Purdy

General Manager

Planet Telecom, a registered trademark of Tasion Communictions Inc.

E:m...@planettelecom.com

O:   780-409-9000 EXT 40

TF: 1-866-296-7771 EXT 40

M: 780-231-7343

PT-logo(small 160 x 49)

 

From: asterisk-biz-boun...@lists.digium.com
[mailto:asterisk-biz-boun...@lists.digium.com] On Behalf Of Robert Rothberg
Sent: Friday, April 08, 2011 11:44 AM
To: Commercial and Business-Oriented Asterisk Discussion
Subject: [asterisk-biz] need ITSP recommendation

 

Every time I install a new PBX for a customer, I end up having to search for
an appropriate iSTP.  For one reason or another, the one's I've used in the
past always have issues.  Most recently I've used Telasip, but for two days
now I have been unable to reach anyone via phone or email.  If you call the
main number and select sales, the voicemail recording sounds like it was
done by some dude that's half asleep with a serious hangover.

Can someone recommend an ITSP for use with my Lylix hosted Asterisk that
provides:
Fixed rate cost for bulk minutes in the US/Canada (I'd need somewhere around
7000)
Supports 8 concurrent calls (with room to grow)
Can port 100 DIDs without charging me an arm and a leg (of those 100 only 20
are currently active, but the customer needs to be able to add more on
demand I'm imagining this means I'll simply have to have all of them
active and just send the non-used ones to nowhere)
Therefore, won't charge me an arm and a leg for 100 DIDs

And is reliable, has fault tolerant gateways in place and has been around a
while and will likely stick around for a long time.

Thanks!

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Re: [asterisk-biz] Asterisk pop-up developer needed - ASAP requirement

[Matt Riddell]

On 11/01/11 10:32 AM, Bruce B wrote:

Hi Everyone,

We are looking for a programmer who is experienced in Microsoft Visual
Studio to customize the code for an Asterisk Pop-up program that already
exists: Outcall

Outcall is an opensource project that provides with pop-up and call-out
through Outlook using Asterisk AMI.
http://code.google.com/p/outcall/

Scope of the project will be detailed and e-mailed to you in you send me
a private e-mail. In nutshell we want it re-themed and we want a few
bugs taken care of.


Have you tried talking to the original developer?

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[asterisk-biz] Quote to add voicemail forwarding to more than one user

[Matt Darnell]

Aloha,

I would like to get a quote on adding the following functionality:

1.  Forward a voicemail that has been left for a user to more than one
mailbox in one step – keep entering mailboxes until all have entered.
The user can record a message that will be pre-pended to the forwarded
message.
2.  Record a message and send it to more than one mailbox in one step –
keep entering mailboxes until all have entered

The code will be donated back to the community and released as GPL.

The code will need to be back-ported to 1.4, no need for 1.6.X

Please let me know.

Happy Holidays,
Matt

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Re: [asterisk-biz] Asterisk Exchange - a waste of money?

[Matt Riddell]

I think a lot of the problems could be solved by having a different 
pricing structure (i.e. CPM etc).

Any chance this could be considered?

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Re: [asterisk-biz] Asterisk cluster / failover program - Lite version announcement

[Matt Riddell]

On 22/11/10 4:58 AM, li...@contacttel.com wrote:
 Sorry you can't put Asterisk and failover in same sentence.

 Any one who Thinks and sells that, is just plain defrauding clients,

 Same goes for those 99.999%uptime guys... that's like under 25 secs downtime
 in 30 days...

 Just no way,

I take it you haven't been following Asterisk SCF then?

Or redfone?

Or the product mentioned here?

Or our ITSP platform?

Or Enswitch's platform?

There are many ways to provide 5 nines - personally we connect customers 
to multiple servers at the same time and a failure of one doesn't impact 
services.  We've had a simultaneous failure of 3 servers without losing 
the ability to provide voice services.

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Re: [asterisk-biz] Asterisk cluster / failover program - Lite version announcement

[Matt Riddell]

On 25/11/10 11:32 AM, John Ervin wrote:
 When I was in the Computer field, 99.999% uptime didn't count planned
 downtime for maintenance and such..

This computer field?

http://tinyurl.com/27yzfwo

:)

I've always considered uptime to be uptime - customer doesn't really 
care why a machine is down.

BTW: what field are you in now?

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Re: [asterisk-biz] Test numbers Worldwide

[Matt Riddell]

On 27/10/10 8:58 PM, Sevana Oy wrote:
 Thank you very much for this number! The problem is that our customer
 would like to have such numbers globally... basically all around the Globe.

For New Zealand you can call +64 800 000 000 for the talking clock - 
depending on how you terminate calls to New Zealand (it's a toll free 
national number).

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Re: [asterisk-biz] cheapest place to buy codec license

[Matt Riddell]

On 1/10/10 8:05 AM, John Todd wrote:

 On Sep 29, 2010, at 10:36 AM, Kanishka Somaratne wrote:

 what is the cheapest place to buy asterisk G729 and G723 license ?

 At this time, Digium is the only authorized vendor of G.729 licenses
 for Asterisk, and there is no commercially authorized G.723 transcode
 software that works with Asterisk to my knowledge.

There is however hardware:

http://www.digium.com/en/products/voice/tc400b.php

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[asterisk-biz] Asterisk News Accepting Submissions

[Matt Riddell]

Hi all,

Sorry for the crosspost but I assume this may be of interest to both 
businesses and users.

The Daily Asterisk News (running since 2004) is now accepting article 
submissions.

Basically I've created a submission form where you specify whether your 
post is commercial or non commercial and I'll be reviewing each article 
to check what it falls under.  You'll be able to specify whether you 
want to hide commercial posts or not.

What we're looking for is anything cool you're doing with Asterisk or 
any products you've created that work with Asterisk.

If you have any ideas or suggestions, feel free to mail me on them.

Oh, and we've moved the Daily Asterisk News web server to Dallas, TX, so 
it should be a bit quicker for those of you in the states - well, 
anywhere except New Zealand really :)

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Re: [asterisk-biz] QueueMetrics support SLA - Are they usually available for support or are they a really small one man run company?

[Matt Riddell]

On 8/09/10 2:32 PM, bruce bruce wrote:
 Thanks Matt. Now, I really want to live in Italy :-)

I know!  Tell me about it :D

 I have already received feedback on the forum just like Marco promised.
 One of his posts seems to have rectified 90% of my issue already.

Cool.

 Thanks for the feedback.

No problems :)

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Re: [asterisk-biz] QueueMetrics support SLA - Are they usually available for support or are they a really small one man run company?

[Matt Riddell]

On 6/09/10 1:44 AM, bruce bruce wrote:
 Thanks for the input guys.

 However, as a service provider I will become very careful in my dealings
 with a company which has a forum up and doesn't post a single response
 to any of it's users in 10 Days (last post responded August 27th). Who
 doesn't go on Holidays? But we all make sure that there are arrangements
 in place.

:) Most of Italy etc pretty much doesn't exist during August and slowly 
starts again in September.

We have a similar time period here in New Zealand over January (although 
not quite as intense).

It's quite culturally different in Italy than in most places in that 
everyone takes their holidays at the same time.

Once you learn that this is the case you begin to understand what a 
sacrifice people are making to be there answering posts and emails at all :)

There are always cellphones for urgent support requests - although 
expect a wait of a few minutes while the sand is dusted from the laptop 
and a UMTS connection is brought online :)

Lenz (and QueueMetrics) have been around for ever (well not literally, 
but for a long time) and I don't think they're going anywhere.  If 
you're needing some reassurance that the company you're looking to deal 
with is not filled with 'fly by nighters', I can offer my assurance to you.

I have no business relationship with QueueMetrics other than to 
recommend their products.

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Re: [asterisk-biz] Allison Smith Hilarity

[Matt]

Yes

On Mon, Aug 16, 2010 at 6:11 AM, Thomas Kenyon dig...@sanguinarius.co.ukwrote:

 On 10/8/10 22:58, Randy R wrote:
  On Tue, Aug 10, 2010 at 2:40 PM, Andrew Lathamlath...@gmail.com
  wrote:
  To punish telemarketers of course!
 
  Precisely!
 
 Is it just me, or is Alison slowly turning into the ATT voice synthesizer?

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Re: [asterisk-biz] asterisk-biz Digest, Vol 73, Issue 13

[Matt Signorello]

Hello,

Is this dialer traffic?

What is your target rate?

-Matt

On Tue, 2010-08-10 at 12:00 -0500, asterisk-biz-requ...@lists.digium.com
wrote:
 Send asterisk-biz mailing list submissions to
   asterisk-biz@lists.digium.com
 
 To subscribe or unsubscribe via the World Wide Web, visit
   http://lists.digium.com/mailman/listinfo/asterisk-biz
 or, via email, send a message with subject or body 'help' to
   asterisk-biz-requ...@lists.digium.com
 
 You can reach the person managing the list at
   asterisk-biz-ow...@lists.digium.com
 
 When replying, please edit your Subject line so it is more specific
 than Re: Contents of asterisk-biz digest...
 
 
 Today's Topics:
 
1. Voip minutes (amit mehta)
 
 
 --
 
 Message: 1
 Date: Mon, 9 Aug 2010 14:41:53 -0800
 From: amit mehta amit.magn...@gmail.com
 Subject: [asterisk-biz] Voip minutes
 To: Commercial and Business-Oriented Asterisk Discussion
   asterisk-biz@lists.digium.com
 Message-ID:
   aanlktinttvp1mr3yjhebt19ronwukxusmb9l0an1q...@mail.gmail.com
 Content-Type: text/plain; charset=iso-8859-1
 
 Hi,
 
 I am looking for voip minutes for USA.(dial from India)
 
 Large volumes available.
 
 Regards,
 Amit Mehta
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Re: [asterisk-biz] Billing/Provisioning Software

[Matt]

Has anyone integrated CDR reporting into vTiger or Sugar?

On Tue, Jun 22, 2010 at 9:40 AM, Matt mhop...@gmail.com wrote:

 Is anyone aware of a billing/provisioning software that is of a fairly
 mature state that can be used with Asterisk, provide billing support, and
 perhaps interface with a GSM cellular network?

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Re: [asterisk-biz] Is DUNDi done-y?

[Matt Riddell]

On 27/05/10 2:38 PM, Jason Garland wrote:
 Is anyone using DUNDi?

Yeah, mostly internally though - for our ITSP platforms.

Basically we create a cluster of Asterisk machines which speak to each 
other via DUNDi.  If you call  on any of the Asterisk servers and 
they don't have that peer locally they'll ask the others where the peer 
is and route accordingly.

We don't really use it for peering externally - think the GPA made that 
difficult.

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[asterisk-biz] VoIPJet

[Matt Riddell]

Hi all,

Just wondering if anyone else is having problems with VoIPJet?

A couple of weeks ago without any changes, calls via them started 
failing with Circuit Busy responses (even to DDI numbers we own worldwide).

We have a reasonable amount of credit with them, and I've sent emails to 
support, fastsupport and to Jon himself without any reply.

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Re: [asterisk-biz] VoIPJet

[Matt Riddell]

On 5/05/10 12:46 PM, Brooks Bridges wrote:
 Good luck, I've had credit on file with them for years, and they refuse
 to answer any emails about reinstating my account so I can use it.

 Guess they're in the business of stealing their long time customer's
 money now.

Well, it wouldn't be too difficult to get the money back, I just reverse 
the charge.

I was more worried about the fact the calls weren't going out - I'm not 
the type to normally need support, so lack of support has never really 
bothered me :)

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Re: [asterisk-biz] Apple SIP application

[Matt Riddell]

On 21/02/10 1:20 AM, Rehan Allah Wala wrote:
 http://www.vnet-corp.com/iphone.htm#

What's wrong with fring?

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[asterisk-biz] Free iPhone Asterisk Function and Application Reference

[Matt Riddell]

Hi all,

I've uploaded a free app for the iPhone called AsteriskRef to the Apple 
AppStore.

This allows you to lookup applications and functions using your iPhone 
or iPod touch so you don't have to jump out of extensions.conf or open 
another terminal tab.

It currently supports applications and functions from Asterisk 1.4, but 
I'm adding 1.6 and trunk at the moment.

It currently requires OS3.1.3, but I've got another version under review 
at the moment which will run on 3.0.

Hope you like it, let me know if you have any questions.

More info here:

http://www.venturevoip.com/news.php?rssid=2353

-- 
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Matt Riddell
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[asterisk-biz] Permanent Queue Members - and redirect to Asterisk Users

[Matt Riddell]

On 26/01/10 9:36 AM, John Rousseau wrote:
 In order to get advice about programming my system
 For example,
 I need to add a permanent member to a Queue.
 What forum would you recommend?

Asterisk Users List.

Also, when creating an email you should really create a new one rather 
than replying to an existing email - a lot of people use threading to 
read their emails.

This means that a conversation is combined together - if someone is 
interested in the conversation they can read it - if not, they can 
delete all emails in the thread.

So, you're likely to get your emails read by more people if you click 
New Email (or whatever the link is) and put the destination email 
address in.

By the way, just add a queue member to queues.conf to make them permanent.

I.E. member = SIP/1234

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[asterisk-biz] cuba routes needed

[Matt - potatoboy.com]

anyone got Cuba competitive routes? Please contact me off list,
thanks

Kind Regards

Matt
sa...@potatoboy.com
http://www.PotatoBoy.COM
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Re: [asterisk-biz] Remote SIP monitor

[Matt Riddell]

On 6/01/10 3:29 PM, Tim Nelson wrote:
 Are these modules offered with or without charge? I'd be very interested
 in them...

Heh, they're just scripts :) copied and pasted bits and pieces from 
other scripts.

Drop me a line offline if you're keen - otherwise I could put them in an 
article or something.

-- 
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Re: [asterisk-biz] Remote SIP monitor

[Matt Riddell]

On 6/01/10 9:14 PM, Olle E. Johansson wrote:
 I would like feedback on what's missing in the AMI in order to monitor an 
 Asterisk platform. I've added a lot of events and some actions to AMI in 
 order to enhance monitoring, but new ideas are always welcome.

RTCP :)

That's probably it.

I'd kinda like to be able to get memory, cpu, disk info, via the manager 
because all up that's what I care about pretty much on most of the 
nodes, but understand that they don't really have much to do with Asterisk.

Reverse Triggers would be nice :D

Like, connect to port x on addr y if you get a certain event and send 
some info.

That way you could have central daemons that log from end nodes.

You can always connect back to the Manager and parse the info, but with 
a lot of machines the load can get kinda high.

If Asterisk were able to make an outbound connection you'd only see 
traffic when something was up.

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Re: [asterisk-biz] Remote SIP monitor

[Matt Riddell]

On 6/01/10 1:28 PM, Erik Lagerway wrote:
 Do any of you know of a simple remote SIP monitor service or
 application/software? Looking for something I can point towards my
 various SIP servers to monitor registrations, ongoing call connectivity
 and email/SMS me if there is a problem etc.

 I don't want to install appliances or dedicate servers in the network
 for this, any ideas?

Munin, Nagios, MRTG etc.

I personally used to use MRTG, but am now using Munin.

If you do decide to go down that path and need modules for Asterisk, let 
me know I'm currently running:

1. Asterisk SIP Peers (online/offline/lagged)
2. Asterisk IAX2 Peers (online/offline/lagged)
3. Asterisk Concurrent Channels
4. Num calls in last hour (answer/no answer/busy etc) - just from MySQL 
CDR disposition.

-- 
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Matt Riddell
Managing Director
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Re: [asterisk-biz] hongkong and china termination

[Matt - potatoboy.com]

We have superior quality china route.

Please contact us off list.


Matt
- Original Message - 
From: Kelvin Chan kelv...@positronics.com
To: asterisk-biz@lists.digium.com
Sent: Wednesday, December 09, 2009 8:33 AM
Subject: [asterisk-biz] hongkong and china termination


 Hi guys,

 I'm looking for hongkong and china termination by SIP.

 Please contact off list.

 Cheers,

 -- 
 Kelvin Chan   | Positronics Ent.
 Product Development   |



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Re: [asterisk-biz] Protect Source Code based software

[Matt Riddell]

On 5/12/09 4:04 AM, Ignacio Ramos wrote:
 Hello frieds, we have finished the development of a high-enterprise
 multi-tenant call center and we are about to install it the client's
 NOC. How ever, we are a little worried about the client getting our
 software and reselling it, since we will install all the
 Asterisk+PHP+Ajax source code in their servers.

 So I would like to know, how do you protect your OpenSource-based
 software from being copypaste? Do you install only the executable
 files? what program do you use for that?

There are a few ways:

1. Zend (already mentioned)

2. Phone Home licensing (be sure you're providing a 
salt+nonce+geographically redundant servers).

3. Compile with extra data to specify client (only really applicable for 
a C app).

I note you're using Asterisk, PHP and Ajax.

Asterisk you'd need a business license from Digium to encrypt - 
otherwise you'd be in breach of the GPL.

PHP - Zend etc

Ajax - Good luck - obfuscation is probably the best.

We provide a mix of options.  Zend framework allows you to get the 
machine ID - you can license a product to just that CPU.

We also provide most of our code in Open Source (BSD) format and 
encourage copying.  The front end of the software then talks to the back 
end of the software (written in ANSI C) which connects to our licensing 
servers to update itself etc.

It's a really messy field, and you'll need to be 100% sure that you are 
not using _ANY_ software written by a third party - unless you're 
releasing as Open Source (in which case you need to be sure your 
licenses match).

Ohloh.net can be good for checking on this.

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Re: [asterisk-biz] auto-answering an originated call, dialplan / manager interface problem

[Matt Riddell]

Instead of sending to a channel, use Local/ex...@context/n

In the context set your variables before making the call.

We do this on a virtual receptionist with touch screen.

You touch the name of the person (or group) you want to speak to, it 
makes a call to a Linksys (which autoanswers) then sends the other leg 
to the person they are trying to contact.

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Re: [asterisk-biz] G.729 Transcoding on a Cell Processor

[Matt Riddell]

On 22/11/09 9:23 AM, Peter Briffett wrote:
 No problem. Buy the card from wherever you want - you can get the software
 from us. Our floating license model means you can 'float' licenses across
 servers for load balancing - you don't need extra hardware for redundancy.

 We've tried to get G.729 transcoding down to a price point that makes sense
 to everyone - I can't see how you can get 425 channels of transcoding on a
 single PCI card any cheaper from anyone else. If you can, please let me know
 and I will more than happily re-evaluate our commercial model.

What's the deal with the license and a GPL'd version of Asterisk?

Or does this have to be used with the Asterisk Business Edition?

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Re: [asterisk-biz] Routes Needed

[Matt - potatoboy.com]

We have direct but you must prepay with bank transfer.

Best Regards

Matt
Account Manager
http://www.PotatoBoy.COM
a FCC Licensed Carrier with:
Premium Plus A-Z featured by 100+ direct inter-connect with 2 sec
PDD, 65%ASR
Premium A-Z featured by LCR among US Tier-1 carriers
Economy A-Z featured by LCR for lowest rate possible

  - Original Message - 
  From: Steven Lauber
  To: asterisk-biz@lists.digium.com
  Sent: Saturday, November 21, 2009 1:34 PM
  Subject: [asterisk-biz] Routes Needed


  Hello


  We need the following routes with wholesale quality. (No FAS
for sure , minimum 2.5 minutes ACD)
  We have live bulk traffic to these routes and need more
capacity.
  Real directs preferred !


  Afghanistan Mobiles
  Albania Proper
  Albania Tirana
  Albania Mobile Vodafone
  Albania Mobile AMC
  Guatemala Mobile Comcell
  Iran Mobile
  Pakistan Proper
  Pakistan Mobile
  Sudan Mobiles
  Tunisia Mobiles
  Turkey Mobiles
  Zimbabwe Mobiles


  We also are building traffic to most of the African mobile
destinations . So please offer us any of your direct African
destinations.


  Please contact me offlist if interested or send an email to
sa...@rivatel.com


  Thanks


  Steven


  Riva Communications
  www.rivatel.com


-
-


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Re: [asterisk-biz] Best/Cheapest DID Provider with 10 or more channels?

[Matt Riddell]

On 10/11/09 2:40 PM, Alex Balashov wrote:
 Yaro Donchenko wrote:

 Before you sing up with any one don't forget to ask them if you will be
 able to port the numbers out if there is any problems.

 In theory, the underlying carriers (UCs) whose facilities those
 providers use are required to allow the port-out, and so is the ITSP.

 Recent FCC rulings clarified that numbers cannot be held hostage to
 billing disputes with the ITSP, and that the end-user is the logical
 owner of that number even if purchasing from a reseller/ITSP and not
 directly from a carrier.

 This is all in theory.  The point is that if you ask if you can port
 out, you'll be told yes.  In practice, it may be a different story.

So legally you could get 10 numbers, never pay a cent, then port to 
someone else and do the same?  Sounds rather dangerous.

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Re: [asterisk-biz] Best/Cheapest DID Provider with 10 or more channels?

[Matt Riddell]

On 10/11/09 3:12 PM, Mitul Limbani wrote:
 Matt,

 There is porting fee per number n time delay, so practically what you
 saying is cumbersome :) (to jump around between telcos with porting
 request)

You'd hope so!

I guess the law makes sense if you have a dodgy provider or something - 
it would suck big time if your main business number was tied up because 
of a dispute that could take months to go through the courts.

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Re: [asterisk-biz] Capitalisation in English writing from Indian subcontinent

[Matt Riddell]

On 9/10/09 3:35 AM, Peter Beckman wrote:
As Alex Said, If The Capitalization Of Words Was Consistent, he might Be
able to understand.  But this is One of Those things that the
capitalization makes No sense, and seems almost random and without
understanding of the Purpose of capitalized letters.

So, it happens for emphasis.  I think that's pretty much agreed.

Consistency then depends on a common train of thought.  A document 
created in one pass without interruption.  With interruption or multiple 
attempts at a document the fact someone has provided emphasis for a word 
earlier does not necessarily mean they will provide emphasis later.

I think you'd find that if they went through the document again, they 
would likely capitalize each occurrence of the word they wanted to 
emphasize.

Maybe the main reason is that we tell people not to use HTML email so 
they can't use bold and get used to capitalization as a replacement. :)

-- 
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Re: [asterisk-biz] need help to develop IM application with the help of asterisk

[Matt Riddell]

On 9/10/09 3:24 AM, Kevin Smith wrote:

 Hi N,

 I am trying to make a separate application for IM and then I will
 integrate it with my developed softclient.

What we've done in the past for this is a few steps:

1. Create a softphone (lots of libraries around)
2. Sign up to some SMS gateway (Clickatel etc)
3. Create a PHP (or whatever) backend page to authenticate/send/receive 
messages
4. Poll a received messages page periodically from the sofphone

So, send a message from softphone to an account on another softphone and 
you store it in the DB until the destination polls for it at which time 
you display it.

If the destination is not the account of another softphone user then:

1. Check user's credit
2. Send SMS via gateway
3. Provide response codes (sent/not sent etc) and charge customers account

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Re: [asterisk-biz] Capitalisation in English writing from Indian subcontinent

[Matt Riddell]

On 9/10/09 10:25 AM, C F wrote:
 On Thu, Oct 8, 2009 at 6:29 AM, Alex Balashovabalas...@evaristesys.com  
 wrote:
 Kevin Smith wrote:

 Hello Alex Sir,

 I have given your url for your help. if you will see on home page errors
 will visible easily to you.

 http://www.evaristesys.com/

 See homepage content only.

 I fear that what is obvious (visible easily) to one person is not to
 another.

 For some reason I fear that HIS spelling is so bad that he is
 convinced that the dictionary is wrong.

Yeah, according to my email spell checker, there are no real spelling 
errors.

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Matt Riddell
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Re: [asterisk-biz] NGTLive Telecom solution provider

[Matt Riddell]

On 8/10/09 12:22 AM, Michael Jerris wrote:
 I take this liberty to mention that your website is horrible and
 provides no information at all about your products and it was probably
 quite a waste of time to post it here.

Tough crowd in here these days! :)

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[asterisk-biz] Creating a monster...

[Matt King]

Hi there,

I've got a requirement to build a 16-port E1 ISDN Asterisk machine (so 
that's a whopping 480 simultaneous calls).  I'm thinking of using 4x 
TE405P or 4x TE410P to provide the 16 E1 ports required.

There is no VoIP requirement at this customer.

Does anyone out there have any experience you can share with building 
and running a system like this?

If so, please get in touch.

Thanks,

Matt.

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Re: [asterisk-biz] Creating a monster...

[Matt Florell]

On 10/2/09, Steve Edwards asterisk@sedwards.com wrote:
 On Fri, 2 Oct 2009, Matt King wrote:

   I've got a requirement to build a 16-port E1 ISDN Asterisk machine (so
   that's a whopping 480 simultaneous calls).  I'm thinking of using 4x
   TE405P or 4x TE410P to provide the 16 E1 ports required.
  
   There is no VoIP requirement at this customer.
  
   Does anyone out there have any experience you can share with building
   and running a system like this?


 I had a similar requirement about 5 years ago for a 24x7x365 adult chat
  site, except it was for T1s instead of E1s. Back then, the interrupt
  activity from the 4 cards was strongly advised against. I don't think that
  is an issue any more.

  When we thought about it, 4 1u pizza boxes with 1 quad card each made more
  sense. The dialplan on these telco servers just answered the call and
  dialed the application server.

  If a telco server crashed, we only lost 25% of our capacity. Likewise
  (because we were running at less than 75% of capacity), we could do a
  rolling upgrade of the telco servers without any downtime. When we
  wanted to upgrade the application server, we changed the telco server
  dialplan to dial the backup application server.

  You may want to take a look at external E1/T1 devices like red-fone.com.

We have also built a 16 x T1 server, except we used two Sangoma
octal(8) port cards.

It is doable, but to reiterate what Steve already said, you would be
better off breaking it out into separate servers.

MATT---

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Re: [asterisk-biz] Direct India with CLI and low PDD

[Matt Riddell]

On 21/09/09 8:58 AM, Michael Schwartzman wrote:
 My friend, this is over multiple STM-1 from 60 Hudson NYC over Tier 1
 carrier to India,

Cool, thanks for the info - will contact you offlist.

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Re: [asterisk-biz] Direct India with CLI and low PDD

[Matt Riddell]

On 21/09/09 9:04 AM, Alex Balashov wrote:
 It must be a very sad carrier.

:) I corrected it in my reply

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[asterisk-biz] T38 Termination

[Matt Riddell]

Hi all,

Just wanting to know who can do bulk T.38 for:

Australia
US
UK

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Re: [asterisk-biz] Dialer Friendly North American Termination

[Matt Riddell]

On 12/09/09 8:12 AM, ja...@dialertermination.com wrote:
 Hello,

 We have about 40,000 ports available, with rates starting at 0.0072 for
 US-48. We also have Canada and Mexico. Every one of our routes are
 dialer friendly as we accept all traffic.

 If you'd like a rate deck, let me know and we can have one sent today
 along with a test form.

How many calls per second do you accept?

-- 
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Re: [asterisk-biz] Dialer Friendly North American Termination

[Matt Riddell]

On 12/09/09 11:05 AM, Alex Balashov wrote:
 Matt Riddell wrote:

 On 12/09/09 8:12 AM, ja...@dialertermination.com wrote:
 Hello,

 We have about 40,000 ports available, with rates starting at 0.0072 for
 US-48. We also have Canada and Mexico. Every one of our routes are
 dialer friendly as we accept all traffic.

 If you'd like a rate deck, let me know and we can have one sent today
 along with a test form.

 How many calls per second do you accept?

 Accept, or actually handle without falling over?

:) yeah well - seeing as it's to be used for emergency broadcasts and 
flu spread polling, falling over would be kinda bad.

You wouldn't believe how many people think they can do x thousand calls 
per second, but start failing once you send them a few hundred CPS :)

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Re: [asterisk-biz] Dialer Friendly North American Termination

[Matt Riddell]

On 12/09/09 11:47 AM, Alex Balashov wrote:
 Hundreds?  Most high-end, high-capacity carrier-grade big iron can
 easily fall over after a few dozen CPS at most.  Good high-density
 ISDN-VoIP gateway chasses advertise 20 CPS, and we know that's a
 stretch, courtesy of marketing.

 Anybody claiming hundreds and hundreds of CPS is completely full of
 shit.  It's very, very difficult to build architectures that can deal
 with this.  Even if subscriber-facing edges can, centralized billing
 plexi can't, and neither can the upstream supplier equipment.

We're doing 400 CPS on a system at the moment.

Load balanced across Asterisk machines, and load balanced across 
provider's machines.

MySQL is the hardest thing to get working, and requires splitting of 
tables and using InnoDB.

Whole system is split into blocks which each for part of a redundant system.

One of the problems was lack of randomisation of numbers - you can take 
out a telephone exchange pretty easily if too many calls go to one place.

-- 
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Matt Riddell
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Re: [asterisk-biz] Voice Broadcast Software

[Matt Riddell]

On 10/09/09 3:25 AM, Ignacio Ortega A. wrote:
 Dear all

 Can anyone recommend a good voice broadcast software?

 we need the following

 Brand name customizable
 Multi user / Campaign
 Press 1 Campaign  (message press digit transfer)
 simple to load leads, load recordings, ,make the call flowchart and easy
 to pull out reports
 other options will be taken in consideration (text to speech,
 voice recognition, among others)

We provide the SmoothTorque message broadcaster which should fulfill 
your needs.

We also now provide a leasing option.

Main page here:

http://www.venturevoip.com/st.php

Have a look at some of the tutorial videos:

http://www.venturevoip.com/st_tut.php

Free demo download here:

http://www.venturevoip.com/st_demo.php

Mail me off list if you have any further questions.

-- 
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Matt Riddell
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Re: [asterisk-biz] Hacker's attack on Asterisk

[Matt Riddell]

On 8/09/09 3:19 PM, Faiz Rehman wrote:

 I was an Extension of 11 digits like1403XXX with the same password, I 
 guess it was a brute force attack.

Seems somewhat unlikely that they would try that many combination on one 
box.

You sure guest account is not allowed to make calls?

-- 
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Re: [asterisk-biz] Hacker's attack on Asterisk

[Matt Riddell]

On 8/09/09 4:09 PM, Alex Balashov wrote:
 Never, ever accept anonymous/guest calls.  For any reason.  Ever.
 Doesn't matter what the reason is.  Just don't.

Honeypot?

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Re: [asterisk-biz] Need USA virtual PRIs for www.VisionVoIP.com

[Matt Riddell]

On 1/09/09 6:53 PM, Gavin Henry wrote:
 Did you see the IAX load balancer that Gradwell.com contributed that
 they wrote and use?

Yeah, I played with it for a bit - don't remember now, but there was a 
feature of IAX that it didn't support - it was a while ago now though, 
so maybe it has been upgraded since then - I really should check it out 
again.

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Re: [asterisk-biz] Need USA virtual PRIs for www.VisionVoIP.com

[Matt Riddell]

On 1/09/09 7:09 PM, Pascal Bruno wrote:
 Lol, now you want to tell people how to use their devices lol, you
 funny. I use the i am very creative in the way i can use my stuff.
 Once again there is no need to feel offended, you talking like you are
 mad. You have a problem with my ipod? You think i should get a zune
 right? And keep it just for music? And the computer i should uninstall
 all media player and save it for internet. I think you should save
 your mouth for creative and constructive comments me. Lol...

To be fair, Alex does usually know what he's talking about.

I think this thread is starting to get a bit far away from Commercial 
and Business-Oriented Asterisk Discussion.

:)

Anyway, what do you need music for? :D

-- 
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Re: [asterisk-biz] Need USA virtual PRIs for www.VisionVoIP.com

[Matt Riddell]

On 31/08/09 9:29 PM, Zeeshan Zakaria wrote:
 So Alex, in other words you are saying that IAX providers are not
 good, i.e. bad providers. Is this a fact or just your personal
 opinion? Do you have any reference(s) to support it? My experience shows
 that IAX is a better choice, plus various IAX vs SIP comparison
 documents I have gone through over time show IAX a better option. Maybe
 you are referring to some protocol other than SIP, but didn't mention it.

What he means is that you're not likely to be able to buy IAX 
termination off someone like verizon or global crossing or whatever.

There's currently no decent IAX load balancer, so you have to do it 
other ways - for SIP there are a few choices.

We work around it by having lots of boxes and round robin DNS, but it's 
pretty rare - most people just do SIP.

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Re: [asterisk-biz] Need USA virtual PRIs for www.VisionVoIP.com

[Matt Riddell]

On 1/09/09 4:55 AM, Alex Balashov wrote:
 Trixter aka Bret McDanel wrote:
 didnt they allow the separation of media and signaling in later
 versions?  Of course this would mean that you are at the whim of the
 users version of iax if they did, and if the user restricts the ability
 I haven't heard anything about that.

Yep, you can do media only transfer of IAX2 as well as the standard 
entire transfer.

It was added in a version of 1.4, but you'll need to be talking to other 
1.4 boxes.

IIRC the option is transfer=mediaonly or some such.

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Re: [asterisk-biz] North America Internet bandwidth

[Matt Riddell]

On 1/09/09 12:22 PM, Alex Balashov wrote:
 Jack Chen wrote:

 I would like to know if there is international internet bandwidth in N/A.
 If I am ordering a 100M dedicated bandwidth, will this link include
 internatinoal bandwidth and how much?

 Internet bandwidth is Internet bandwidth.  It is not topologically
 organised along national boundaries.

Not quite correct internationally.

Here in New Zealand we pay for national and international bandwidth.

So, we have to purchase a package for national bandwidth (i.e. 10Mbit 
guaranteed - no cap) and a package for international bandwidth (i.e. 
5Mbit and 100GB cap).

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Re: [asterisk-biz] North America Internet bandwidth

[Matt Riddell]

On 1/09/09 1:23 PM, Alex Balashov wrote:
 The question was about North America.  My answer was implicitly scoped
 to North America.  I should have been clearer about that.

Sweet :)

 Other countries - particularly smaller and topographically isolated ones
 such as NZ - have a myriad of other arrangements.

Who you calling small? :)

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[asterisk-biz] Looking for new resellers in Middle East and Africa

[Matt Riddell]

Hi,

We're looking for new resellers for SmoothTorque.

Mainly in the Middle East and Africa, although we currently only have 
exclusivity deals in place in the UK.

If you're interested in working with us, please send me a mail offlist.

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[asterisk-biz] Interview with John Todd

[Matt Riddell]

Hi,

We've just completed an interview with John Todd - the Open Source 
Community Director for Asterisk.

Some interesting comments in it - hope you enjoy it:

http://www.venturevoip.com/news.php?rssid=2200

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[asterisk-biz] One week 50% off SmoothTorque

[Matt Riddell]

Hi all,

We're running a 50% off the cost of SmoothTorque (bringing the price 
from US$4000 to US$2000) for one week for new installations when you 
mention this mail to the Asterisk Business List.

If you're interested, please send me an email off list and we can get 
you set up and going.

Lease pricing has not changed - if you're interested in leasing 
SmoothTorque from us or would like to use someone else's hosted 
solution, just let us know.

The above price is for one server, although we will provide additional 
discounts over and above the normal discounts for multiple server instances.

Why are we doing this?  We need to raise some additional capital for 
some new projects which we are undertaking.

A description of SmoothTorque is available on our website:

http://www.venturevoip.com

SmoothTorque is traditionally used for:

* Emergency Warning Telephone Broadcasts
* Automated disease spread polling
* Political Polling
* Predictive Dialing for call centres
* Message Broadcasting
* Providing a hosted service

If you require any further information or would like to see a 
demonstration please don't hesitate to contact me off list, or send an 
email to sa...@venturevoip.com

Don't forget that in order to get the reduced pricing you will need to 
mention this email.

This offer expires one week from now on Saturday August the 29th.

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[asterisk-biz] 39 Free Softphones

[Matt Riddell]

I've written another round up article on soft phones that you can use 
with Asterisk.

I've tried to omit any that required you to sign up before downloading, 
or had time limits to usage etc.

There are a couple in there that actually have their own servers but may 
be interesting nonetheless.

I also noticed while writing it up there's a version of Firefly for Mac 
(beta) now.

Anyway, article complete with screenshots is here:

http://www.venturevoip.com/news.php?rssid=2188

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Re: [asterisk-biz] 30 Great free Asterisk applications

[Matt Riddell]

On 11/08/09 2:22 AM, Tim Booth wrote:
 where is this list?

http://www.venturevoip.com/news.php?rssid=2184

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Re: [asterisk-biz] 30 Great free Asterisk applications

[Matt Riddell]

On 10/08/09 2:55 AM, Alexander Argov wrote:
 Matt

 We moved the CRI to a free download zone.

Great! I've added it to the list (now 35 Great free Asterisk applications).

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Re: [asterisk-biz] SMS gateway API.

[Matt Riddell]

On 8/08/09 1:04 PM, Alex Balashov wrote:
 Hi everyone,

 Sorry if this is a little off-topic, but I thought I would tap the
 enormous collective wealth of knowledge here.

 I come from the fixed-line world, so I don't know terribly much about SMS
 or mobile anything.

 I have a situation where I need an endpoint to receive a fairly large
 amount of SMS messages in something close to real-time and then be able to
 automatically do something with them as part of a backoffice process, and
 need to set up something rather quickly.

Is this not the exact same message as the Voice Ops list one?

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Re: [asterisk-biz] Reseller's Platform

[Matt Riddell]

On 8/08/09 3:14 AM, Barton Fisher wrote:
 I'm looking for a someone that can provide a hosted asterisk reselling
 platform that is not locked to to their rates and services. For example
 I can use my provider for calls. I'm mainly looking for customer sign-up
 and auto billing for out and inbound usage.

 If you have someone you like, please let me know.

A little biased :-)

http://www.venturevoip.com/exchange.php

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Re: [asterisk-biz] Looking for a SIP Termination Provider

[Matt Riddell]

On 5/08/09 10:11 PM, Faiz Rehman wrote:
 We've been having problem in finding a really reliable company to
 provide us with SIP Termination. So far we've used VoiceTrading (Premium
 routes)
 and have problems with calls not being connected, messages not being
 played, false answer, etc etc.
 Any help would be greatly appreciated.

As others have said, it really depends on what you're after.

We can provide New Zealand routes.

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[asterisk-biz] New VentureVoIP Site and Free SmoothTorque Demo

[Matt Riddell]

Hi,

We've finally launched our new website and have included the option for 
a 7 day trial of SmoothTorque including tutorial videos on how to use it 
- drop me a note if you have comments.

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Re: [asterisk-biz] Call Termination Beta Testers Needed (Dialer OK)

[Matt Riddell]

On 5/06/2009 5:44 a.m., James Shanks wrote:
 Greetings!

 This is an official Beta Testing Invitation from PBX Techs LLC:

 ** We're currently seeking companies to beta test our new termination
 platform.  Dialer traffic is okay (subject to verification).

How many calls per second can you do?

-- 
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Re: [asterisk-biz] Is NuFone dead?

[Matt Riddell]

On 1/05/2009 10:09 a.m., Trevor Hammonds wrote:
 I have been unable to reach NuFone.net since earlier today.  There
 website is also unreachable.  I was wondering if anyone else is
 experiencing this, and also if anyone knows of the viability of the company.

Looks like their DNS entries are gone?  Maybe they forgot to renew their 
domain name?

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[asterisk-biz] Asterisk/Database Admin

[Matt Riddell]

Hi,

Please send your details and CV if you are interested in a job doing 
Asterisk and Database administration in the greater London area.

This is a full time job.

-- 
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Matt Riddell
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Re: [asterisk-biz] OT: Co-location vs Build Out Insight

[Matt Gibson]

 Its interesting that most people prefer colo. I dont know why but I'm
afraid of trusting other 
 people with my operations or data. The thought of the colo employee,
sending out millions of 
 voip calls from my box perhapse. How did you guys over come this fear? or
am I just being too 
 paranoid? 


I don’t think the colo would be in business very long if they did that or
anything similar. That’s what contracts are for, along with research into
the best places to host in the area you require.


 


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Re: [asterisk-biz] Africa and middle east

[Matt - potatoboy.com]

some of our white routes for African and middle east countries
have cli. such as some Egypt prefix, Kuwait, etc. please contact
me off list for your specific requirement.

thanks

Matt Liu
Account Manager
sa...@potatoboy.com
http://www.PotatoBoy.COM
Click-2-Call:
http://panel.potatoboy.com/clickcall.php?n=206260003p=%2FYtYKCiPLIM%3D;
SIP: 206260...@us3.potatoboy.com
a FCC Licensed Carrier
Premium A-Z featured by 100+ direct inter-connect with 2 sec PDD,
65% ASR, no capacity limit

  - Original Message - 
  From: ContactTel Business
  To: 'Commercial and Business-Oriented Asterisk Discussion'
  Sent: Monday, April 06, 2009 12:52 PM
  Subject: [asterisk-biz] Africa and middle east


  Anyone have white routes to Africa/middle east that passes the
callerid number ?



  Need 50 channels



  Thanks



-
-


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[asterisk-biz] Parabel Asteroid

[Matt Riddell]

I was reading through a bug report last night and noticed the following 
product:

http://www.parabel-labs.com/products/asteroid/

Asteroid is a voice channel bank, designed in 1U 19 case. It is 
comprised of four FXS/FXO modules with 8 ports each. The Asteroid can be 
used for attachment of plain telephone subscribers (lines) to the 
traditional PBX or IP PBX Asterisk/CallWeaver. The E1 port can be used 
for bank connecting to the traditional PBX. Alternatively, with usage of 
the Ethernet port and TDMoE protocol, Asteroid can be connected to the 
Asterisk/CallWeaver. In both cases Loop Start signalling is provided for 
PBX.

Anyone used these?  If so, what are they like?  Stable?  Support?

-- 
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[asterisk-biz] ABE concurrent calls or channels

[Matt Florell]

Hello,

I am working on a client proposal and I was just wondering if someone
could clear this up for me, the Digium website claims that:

Asterisk Business Edition supports from 10 to 240 simultaneous calls
per system.
(http://www.digium.com/en/products/software/abe.php?tab=benefits)

Is this really Asterisk calls or is this Asterisk channels?

There is a big difference between the two and I was looking for a
clarification before I put any specifications in this proposal. If a
call is originating from a SIP phone on the server over a SIP trunk
that would be 1 call, but 2 channels. If those two SIP end points went
into a meetme conference that would then be 4 Asterisk channels(you
need pseudo channels for non Zap/DAHDI meetme participant channels).
You can see where the math could be very different depending on
Digium's definition of calls.

Thanks,

MATT---

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Re: [asterisk-biz] Available for Software Development

[Matt Riddell]

On 11/03/2009 9:30 a.m., Dwayne Hubbard wrote:
 I'm available for software development work.  If anyone would like to
 discuss opportunities that they may have, please contact me off list at
 dwayne (dot) hubbard (at) gmail (dot) com.  A brief career description is
 available at http://www.linkedin.com/in/dwaynehubbard

 I have been in telecommunications since 1995 and have worked for great
 companies like Dialogic, Intel, and Digium.  My activity in the Asterisk
 community has mostly involved Asterisk Business Edition, but I have also
 authored chan_dialogic, taskprocessor.c (available in 1.6.1), and the FAX
 modules that are currently in Beta.  I am also one of the co-architects of
 the C++ Communication Services Framework available on sourceforge (
 http://sourceforge.net/projects/commsvcfw/)

How come you left Digium?

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Re: [asterisk-biz] Fraud alert

[Matt Gibson]

Same here, but about 3 months ago. Luckily I was able to stop it after about
30 minutes, but they still got about 100 calls out, I got a lot of calls
back from little old ladies wanting to give me their credit card info, scary
stuff. 


 -Original Message-
 From: asterisk-biz-boun...@lists.digium.com [mailto:asterisk-biz-
 boun...@lists.digium.com] On Behalf Of C. Savinovich
 Sent: Friday, February 27, 2009 4:18 PM
 To: 'Commercial and Business-Oriented Asterisk Discussion'
 Subject: Re: [asterisk-biz] Fraud alert
 
 
   It seems to be the same pattern of people who attacked 3 of my
 servers in
 a 3 week period a couple of weeks ago.  The calls were made mostly to
 area
 codes 252 and 818 and indeed they showed the caller-id of the phones.
 My
 customer claims he received a call from the FBI saying that the calls
 were
 credit card solicitations.  The point is, whoever is doing this, is
 doing
 this massively.
 
 CS
 
 -Original Message-
 From: asterisk-biz-boun...@lists.digium.com
 [mailto:asterisk-biz-boun...@lists.digium.com] On Behalf Of
 voip-aster...@maximumcrm.com
 Sent: Friday, February 27, 2009 4:04 PM
 To: Commercial and Business-Oriented Asterisk Discussion
 Subject: Re: [asterisk-biz] Fraud alert
 
  I'd suggest to everyone to ban that IP, it's been scanning our
 networks
  from time to time, in a sequential manner by IP.
 
  I've had really good luck with this:
 
  http://www.voip-
 info.org/wiki/view/Fail2Ban+(with+iptables)+And+Asterisk
 
  Basically, it automatically blackhols via IPtables any host that
 fails a
  certain number of registration attempts in a given period.
 
 Yeah we're actually rolling it out on all of our production servers,
 it's
 a great application to run.
 
 I'm working on some scripts to propagate the bans to the firewall so
 that
 all of the servers get protected as soon as possible.
 
  [default]
  ; Send any unauthenticated calls to the local FBI office
  context=local-fbi-office
 
  I've got a honeypot server that pretty much accepts any calls that
 come
  through, and plays a Thank you for calling the Telecommunications
 Fraud
  hotline. Please stay online for the next available representative.
 If
 they
  stay online for more than 20 seconds, it connects them to an agent at
 the
  FBI that we have been working with.
 
  I've been meaning to add some code in that pulls out the originating
 IP
  address of the call and tells it to the agent when we call. :)
 
 That would be great to have!
 
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[asterisk-biz] OrderlyStats Server Edition - Now Available

[Matt King]

Hello,

I thought you might like to know that our popular Managed Service for 
Call Centres, OrderlyStats, is now available as a 
download-and-install-yourself Server Edition.

You can get it at:

http://www.orderlyq.com/statistics.html

Just look for the 'Download Now' link.

OrderlyStats is a comprehensive call centre management platform for 
Asterisk, featuring:

- Fully Real-Time (synchronous/message-based) Control Panel showing you 
every event at your call centre as it happens,
- Agent Toolbar giving agents Login, Pause and queue and caller 
information to agents in real-time, as well as screen-pops,
- Automatic monitoring of inbound, outbound and queue calls,
- Automatic daily update emails for your call centre manager,
- Over 450 historical figures, reports, graphs and charts,
- Aggregate reports across multiple queues,
- Multiple users, roles and privileges,
- Workforce management tools,
- Fully automatic self-configuration.

OrderlyStats SE is compatible with all known versions of Asterisk.  
Orderly Software is an official Digium Software Partner.

Enjoy!

Kind regards,

Matt King
Managing Director
Orderly Software Ltd.



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[asterisk-biz] Inbound PSTN - SIP Requirement (Europe-Wide)

[Matt King]

Hi there,

We have a requirement from a customer to provide their callers with a 
range of europe-wide inbound PSTN numbers, for SIP (preferably) or IAX2 
delivery to their premises in London.

We are seeking tender from local SIP termination providers for this 
business.

The country and city codes we need are as follows:

Holland   +31 (0) 207 (Amsterdam)
France+33 (0) 1 70(Paris)
Germany +49 (0) 1801   (Non Geographic)
   +49 (0) 30  (Berlin)
Greece+30 (0) 21 302 (Athens)
Italy+39 06 452   (Rome)
   +39 02 304  (Milan)
Czech Republic+42 02 460 (Prague)
Hungary  +36 (0) 199  (Budapest)
Poland   +48 (0) 222   (Warsaw)
Romania +40 (0) 31(Bacau)
Slovakia+421 (0) 2 33(Bratislava)

If you are also able to port existing numbers to your service that would 
be ideal.

Please get back to me if you are able to provide inbound numbers in any 
of these cities.

Kind regards,

Matt King
Managing Director, Orderly Software Ltd.




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Re: [asterisk-biz] BroadTel iPBX-400 IP-PBX with 4 port FXS or FXO

[Matt Riddell]

On 18/02/2009 8:35 a.m., VoIP Newbie wrote:
 Hi all,
 BroadTel has released iPBX-400 with up to 4 ports of FXS or FXO, 2 Ethernet
 and 1 SD/MMC slot. The product with CPU of 400MHz and 64M RAM and 256M Flash
 supports up to 50 extensions for home soho, small and medium companies. It

Really?  How have you found 50 concurrent calls on a 400Mhz machine with
64Mb of RAM?  What did you use for testing?

 is 100% Asterisk compatible, supporting IAX and SIP.  It weighs less than
 0.5kg in compact size. OEM and ODM orders are welcome. For details, please
 visit BroadTel corporate web site
 http://www.broad-tel.com/products/ippbx.php, or conact us at
 broad...@126.com if you are interested in the product.
 Thanks,
 BroadTel
 P.S. We also make TDM400P with FXS and FXO modules available at good price
 to enhance competitive edge of your business.

Do you find that this works?  I.E. 4 FXO modules with software echo can
on a 400Mhz machine?

-- 
Kind Regards,

Matt Riddell
Director
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Re: [asterisk-biz] PBX got Hacked

[Matt Riddell]

On 8/02/2009 2:31 p.m., VIP Carrier wrote:
 Here is an IP witch they have used to access a system *116.122.36.95*

Did you talk to the contact?

inetnum:  116.120.0.0 - 116.127.255.255
netname:  HANANET
descr:Hanaro Telecom
descr:Shindongah Bldg, 43, Taepyeongno2-ga,
descr:Jung-gu, Seoul
descr:***
descr:Allocated to KRNIC Member.
descr:If you would like to find assignment
descr:information in detail please refer to
descr:the KRNIC Whois Database at:
descr:http://whois.nic.or.kr/english/index.htm
descr:***
country:  KR
admin-c:  HL196-AP
tech-c:   JK250-AP
status:   Allocated Portable
remarks:  www.hanaro.com
mnt-by:   MNT-KRNIC-AP
mnt-lower:MNT-KRNIC-AP
changed:  hm-chan...@apnic.net 20070521
source:   APNIC

person:   Han Lee
nic-hdl:  HL196-AP
e-mail:   ip-...@hanaro.com
address:  726-1, Janghang2-dong, Goyang-si, Ilsan-gu, Seoul
phone:+82-2-106
fax-no:   +82-2-6266-6483
country:  KR
changed:  hostmas...@nic.or.kr 20040326
mnt-by:   MNT-KRNIC-AP
source:   APNIC

person:   Jinyoung Kim
nic-hdl:  JK250-AP
e-mail:   ip-...@hanaro.com
address:  726-1, Janghang2-dong, Goyang-si, Ilsan-gu, Seoul
phone:+82-2-106
fax-no:   +82-2-6266-6483
country:  KR
changed:  hostmas...@nic.or.kr 20040326
mnt-by:   MNT-KRNIC-AP
source:   APNIC

inetnum:  116.120.0.0 - 116.127.255.255
netname:  broadNnet-KR
descr:SK Broadband Co Ltd
country:  KR
admin-c:  IM12-KR
tech-c:   IM12-KR
status:   ALLOCATED PORTABLE
mnt-by:   MNT-KRNIC-AP
remarks:  This information has been partially mirrored by APNIC from
remarks:  KRNIC. To obtain more specific information, please use the
remarks:  KRNIC whois server at whois.krnic.net.
changed:  hostmas...@nic.or.kr
source:   KRNIC

-- 
Kind Regards,

Matt Riddell
Director
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Re: [asterisk-biz] Fraud from 12.168.40.206 and 80.232.111.204

[Matt Riddell]

On 17/02/2009 8:32 a.m., Dovid B wrote:
 Hi,
 These two IP's were trying to send me traffic to a client of mine. I sent the 
 calls to my phone. The have an IVR saying that they are a bank and that need 
 to verify my credit card information. Just wanted to pass it along.

:)

Well, did you confirm your credit card details?  They might shut you off
if you don't!

:)

-- 
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Matt Riddell
Director
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Re: [asterisk-biz] Fraud from 12.168.40.206 and 80.232.111.204

[Matt Riddell]

On 17/02/2009 10:27 a.m., David Gibbons wrote:
 If I send my card info to the list, will you forward it to them for me?
 
 I don't want to have it shut off...
 
 :)
 
Yeah no problem, there may be a small charge for my new
Ferrari^H^H^H^H^H^H^H^H^H^H^H processing though.

:)

-- 
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Matt Riddell
Director
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Re: [asterisk-biz] Call Recording and Legal Issues in the US

[Matt Florell]

Hello,

While there is only a requirement of single-party notification that a
conversation is being recorded at the federal level, there are 12
states in the USA that have a requirement that all parties in a phone
call must be notified that a call is recorded. Here is the list of
those states:
- California
- Connecticut
- Florida
- Illinois
- Maryland
- Massachusetts
- Michigan
- Montana
- Nevada
- New Hampshire
- Pennsylvania
- Washington

All of the states above allow for criminal charges, and all but
Montana allow for Civil penalties as well. The penalties vary from
state-to-state, but often depend on what was done with the recording
if it has even been kept.

MATT---



On 1/28/09, SIP s...@arcdiv.com wrote:
 The answer is that, unfortunately for you, you can be sued regardless of
  what sort of waiver your customer signs. The argument being that your
  customer may not know all the applicable laws regarding his rights in
  the situation before he signs the waiver (waivers are not terribly
  effective protection in court).

  Whether or not whomever is suing you will WIN the suit will rely mostly
  on the law, and you may or may not come out okay in a legal position,
  but the ensuing costs of fighting the legal battle CAN lead to a
  bankruptcy situation.  And, of course, if you're seeking funding rounds
  or business loans, no one will want to give money to a company that may
  lose it all in a legal battle.

  Unless you're absolutely certain of the laws of the states in which you
  do business, I would recommend against aiding in the recording of calls
  by customers.

  N.



  Peter Beckman wrote:
   I'm aware that in some states and in some cases there are differing rules
   about who and when a caller or callee can or cannot record a phone call.
  
   As a service provider, I can easily enable my users to record calls on
   demand, for only some of calls, or for all calls.
  
   The question is, do I need to, in our Terms of Service, specifically say
  
The user is responsible for following all applicable laws regarding
call recording.  The user will hold harmless Company and its vendors
partners and subsidiaries for violations of any laws regarding call
recording.
  
   Or would we be covered by a general you can't sue us clause.
  
   Please -- no conjecture, only reply if you actually know from a good legal
   source the answer.
  
   Beckman
   ---
   Peter Beckman  Internet Guy
   beck...@angryox.com http://www.angryox.com/
   ---
  
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Re: [asterisk-biz] Consortium of Sorts for Interconnects and VoIP providers to a Lesser Degree

[Matt Riddell]

On 29/01/2009 12:36 p.m., Steve Totaro wrote:
 To better answer your question, those people will never be accepted or
 weeded out quickly.
 
 I am not sure why the benefits are not obvious to you, but I would
 guess that having a negative viewpoint on the idea precludes you from
 being in the Consortium.
 
 How long have you been doing Asterisk?  How long have you been a
 consultant?  How many call centers have you setup successfully?  How
 many systems in the real world?
 
 Qualify yourself as an expert before shooting down ideas.

I'm kinda keen, and am in somewhat of the same boat as you Steve; I have
jobs but not enough staff or time to accept them and so pass them on to
others.

I also have other jobs where my physical location (New Zealand) makes it
hard for me to travel onsite.

Obviously you'd want to be able to continue operating as your own
company too as we have significant existing arrangements with partners.

Another problem I see with this is the generation of contracts that span
borders.  I'm not sure I'd want to sell a system into New
Zealand/Australia that was governed by US laws (and therefore US patents).

I'm kinda seeing this as a loose consortium - what do you think?

-- 
Kind Regards,

Matt Riddell
Director
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