[asterisk-biz] 407 DID available from www.didx.net

Nauman Ibrahim ni at supertec.com
Fri Apr 9 11:57:05 CDT 2010


Dear Luis,

We can provide you  407 area code. Could you email me at ni at supertec.com ,
so that I could reply you back with the quote.
Regards,

On Fri, Apr 9, 2010 at 9:06 PM, <asterisk-biz-request at lists.digium.com>wrote:

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> Today's Topics:
>
>   1. Dutch language recording (Arkadi Shishlov)
>   2. Orlando DID's (Luis Mata)
>   3. Friday Apr 9th 2010 @ 12 Noon EDT: More Cloud     Telephony (Randy R)
>   4. The hunt for a workable Asterisk GUI (Chris Bagnall)
>   5. Re: The hunt for a workable Asterisk GUI (Stelios Koroneos)
>   6. Re: The hunt for a workable Asterisk GUI (Zeeshan Zakaria)
>   7. Re: Orlando DID's (Rick Orford)
>
>
> ----------------------------------------------------------------------
>
> Message: 1
> Date: Fri, 09 Apr 2010 03:00:08 +0300
> From: Arkadi Shishlov <arkadi.shishlov at gmail.com>
> Subject: [asterisk-biz] Dutch language recording
> To: asterisk-biz at lists.digium.com
> Message-ID: <4BBE6E08.3040505 at gmail.com>
> Content-Type: text/plain; charset=UTF-8
>
> Hi!
> We are interested in a person who speaks Dutch language to perform a ~1h
> recording for IVR menu. Project details here
> http://lists.digium.com/pipermail/asterisk-users/2010-April/247044.html
> Send you inquiries to leo at azuaz.com
> This is price sensitive assignment because the project itself is close to
> non-commercial, but you can get famous. :)
>
>
>
> ------------------------------
>
> Message: 2
> Date: Thu, 8 Apr 2010 23:17:14 -0400 (EDT)
> From: Luis Mata <luism at ocntechnologies.com>
> Subject: [asterisk-biz] Orlando DID's
> To: asterisk-biz at lists.digium.com
> Message-ID:
>        <
> 32522430.35031270783034481.JavaMail.root at zimbra.ocntechnologies.com>
> Content-Type: text/plain; charset=utf-8
>
> Does any one have Orlando FL DID's (area code 407) need info.
>
> When you send info , let me know if you also accept number portability.
>
> thank you
>
> Luis B. Mata
> http://www.ocntechnologies.com
> 954-889-8626
>
>
>
>
> ------------------------------
>
> Message: 3
> Date: Fri, 9 Apr 2010 10:03:47 +0200
> From: Randy R <randulo2008 at gmail.com>
> Subject: [asterisk-biz] Friday Apr 9th 2010 @ 12 Noon EDT: More Cloud
>        Telephony
> To: VOIP Users Conference <VOIP-Users-Conference at googlegroups.com>,
>        Asterisk Users Mailing List - Non-Commercial Discussion
>        <asterisk-users at lists.digium.com>,      Commercial and
> Business-Oriented
>        Asterisk Discussion     <asterisk-biz at lists.digium.com>
> Message-ID:
>        <u2g1860686a1004090103q6c19c7bz78ed74fcd55dfccc at mail.gmail.com>
> Content-Type: text/plain; charset=ISO-8859-1
>
> Today, Chris Matthieu, Founder & CEO of GetVocal,  entered the
> cloud-based communications market in February, with its launch of
> Teleku.
>
> Teleku is a new cloud-based telecom service that allows Web developers
> to build and host phone applications that answer inbound calls and
> initiate outbound calls, interact with Web applications, and
> send/receive SMS text messages!
>
> Chris, who you may have met at Astricon last year, is our guest later
> today. If you're interested in the cloud - and who isns't, even if you
> don'thave immediate plans - join us, first on IRC on Freenode.net
> (channel #vuc) or http://vuc.me/irc for a web-based client.
>
> The VUC takes place at Noon Eastern US Time, but for your time zone,
> look here : http://vuc.me/next
>
> General info on how to connect, etc: http://vuc.me
>
> SIP
>
> sip:200901 at login./zipdx.com is best for g722 wideband-capable phones
> and accepts g711 as well
>
> You can also call sip:7463#22622#1 at proxy.ideasip.com to connect to the
> Talkshoe bridge.
>
> /r
>
>
>
> ------------------------------
>
> Message: 4
> Date: Fri, 9 Apr 2010 11:09:00 +0100
> From: "Chris Bagnall" <lists at minotaur.cc>
> Subject: [asterisk-biz] The hunt for a workable Asterisk GUI
> To: "'Commercial and Business-Oriented Asterisk Discussion'"
>        <asterisk-biz at lists.digium.com>
> Message-ID: <0a3001cad7cc$ae6252e0$0b26f8a0$@cc>
> Content-Type: text/plain;       charset="UTF-8"
>
> Greetings list,
>
> I've traditionally been a proponent of the "manual configuration "
> approach, using .conf files and command lines to give us the greatest
> possible flexibility when writing call flows for customers. I've shied away
> from web interfaces as being overly restrictive and limiting what we can do
> for our customers.
>
> However, as we've grown as a company, taken on more customers (and hence
> more staff), there's become an ever-growing need for certain operations to
> be carried out by admin staff, rather than always having to be passed down
> to technical staff (who often have better things to do). I'm sure it's a
> problem faced by many companies on the list.
>
> So, what to do about it? Obviously there are "user-friendly" interfaces
> like FreePBX available, but they take over the *whole* asterisk config,
> shoehorning the user into their own fairly tight confines. Don't get me
> wrong, FreePBX is great as a company PBX installed on an on-site server, but
> it isn't much good as a VoIP hosting platform.
>
> What I think we're looking for is a fairly simple web interface to
> manipulate the tables used by Realtime. It doesn't have to be friendly. It
> doesn't have to be pretty. It just has to be easy enough for admin staff to
> use (with training, obviously) so that trivial call flow changes such as
> "please forward my calls to this mobile number" or "can you add extension
> 241 to this queue/ring group" can be made without having to involve
> technical staff.
>
> Would be very interested to hear what others in a similar position have
> done to overcome this growth problem. Did you write your own interface? Did
> you buy something off the shelf? Is there something in the FOSS marketplace
> that'll do the job?
>
> Regards,
>
> Chris
> --
> For full contact details visit http://www.minotaur.it
> This email is made from 100% recycled electrons
>
>
>
>
> ------------------------------
>
> Message: 5
> Date: Fri, 09 Apr 2010 14:46:48 +0300
> From: Stelios Koroneos <skoroneos at digital-opsis.com>
> Subject: Re: [asterisk-biz] The hunt for a workable Asterisk GUI
> To: Commercial and Business-Oriented Asterisk Discussion
>        <asterisk-biz at lists.digium.com>
> Message-ID: <1270813608.14364.49.camel at localhost.localdomain>
> Content-Type: text/plain
>
> On Fri, 2010-04-09 at 11:09 +0100, Chris Bagnall wrote:
> > Greetings list,
> >
> > I've traditionally been a proponent of the "manual configuration "
> approach, using .conf files and command lines to give us the greatest
> possible flexibility when writing call flows for customers. I've shied away
> from web interfaces as being overly restrictive and limiting what we can do
> for our customers.
> >
> > However, as we've grown as a company, taken on more customers (and hence
> more staff), there's become an ever-growing need for certain operations to
> be carried out by admin staff, rather than always having to be passed down
> to technical staff (who often have better things to do). I'm sure it's a
> problem faced by many companies on the list.
> >
> > So, what to do about it? Obviously there are "user-friendly" interfaces
> like FreePBX available, but they take over the *whole* asterisk config,
> shoehorning the user into their own fairly tight confines. Don't get me
> wrong, FreePBX is great as a company PBX installed on an on-site server, but
> it isn't much good as a VoIP hosting platform.
> >
> > What I think we're looking for is a fairly simple web interface to
> manipulate the tables used by Realtime. It doesn't have to be friendly. It
> doesn't have to be pretty. It just has to be easy enough for admin staff to
> use (with training, obviously) so that trivial call flow changes such as
> "please forward my calls to this mobile number" or "can you add extension
> 241 to this queue/ring group" can be made without having to involve
> technical staff.
> >
> > Would be very interested to hear what others in a similar position have
> done to overcome this growth problem. Did you write your own interface? Did
> you buy something off the shelf? Is there something in the FOSS marketplace
> that'll do the job?
> >
> > Regards,
> >
> > Chris
>
> We have been working on our gui for sometime now, targeting mostly
> embedded devices where mysql for settings its not just an overkill but
> most of the times simply can't run due to system architecture and
> resources.
> With that factor, we decided to "bite the bullet" and build our own gui
> from scratch.
>
> What i can say after this experience is 1st that there is no "silver
> bullet" regarding to gui's and 2nd that there are basically two types of
> gui's people want.
>
> One is what i call the "we do it all" gui like FreeePBX and others that
> try to cover the diverse requirement their extended user base has.
> The cost to this is an added "layer" of complex setup files and
> dialplans, which is pretty much tailored to what the gui designer thinks
> is the "correct way" of doing things.
> This kind of gui's are mostly for end-users or power-users that don't
> know or don't want to know the inner workings of a complex asterisk
> system. "They just want it to work" (tm)
> People more technically inclined with asterisk find this kind of gui
> rather restrictive and/or complex
>
> On the other hand there is the "spartan" gui, which does some pretty
> basic and usually time consuming tasks like the ones you mention or
> phone provisioning for example.
> These type of gui's also add a layer but usually its much smaller and
> are easier for professionals to customize.
>
> The main issue i have seen in both cases, over the years i have been
> working on this,is that the gui is pretty much "tied" with the
> underlaying system and dialplan philosophy.
> If you try to make a "generic", or "we do it all gui" you end up with
> something so big which is pretty much a system on its own, with its own
> quirks and gotsa's.
>
> On the other hand, people are "accustomed" to gui's coming with "off the
> self devices" (phones, routers you name it) and expect something similar
> to exist for asterisk.
>
> The problem here that few people realize, is that the "off the self
> appliances" have "standard" hardware and philosophy to start with, while
> with asterisk the sky is the limit.
>
> Just think how many different interconnections there are from 1 port
> analog, to isdn bri and multi port PRI with different channel drivers
> and settings, or how "pickup" works over different channels and you get
> the picture.
>
> So my advise is if you think that the available gui's are too big or
> complex find one of the "spartan" gui's and try to customize it or
> (shameless plug follows) contact me of list to arrange of demo of or
> gui.
>
>
> --
> Stelios S. Koroneos
>
> Digital OPSiS - Embedded Intelligence
>
> Tel +30 210 9858296 Ext 100
> Fax +30 210 9858298
> http://www.digital-opsis.com
>
>
>
>
> ------------------------------
>
> Message: 6
> Date: Fri, 9 Apr 2010 07:48:37 -0400
> From: Zeeshan Zakaria <zishanov at gmail.com>
> Subject: Re: [asterisk-biz] The hunt for a workable Asterisk GUI
> To: Commercial and Business-Oriented Asterisk Discussion
>        <asterisk-biz at lists.digium.com>
> Message-ID:
>        <s2u5ad99e891004090448x4509dbb4tb065ec80b1b4ed14 at mail.gmail.com>
> Content-Type: text/plain; charset="iso-8859-1"
>
> I faced the same situation, and ended up programming my own GUI for this
> purpose, using realtime. It doesn't have queue support yet. I have
> programmed another multi-tenant solution for a client who provides hosted
> PBX solution. I was thinking of fine tuning he code I have and make it
> public, but don't know when I'll get time for that.
>
> FreePBX is good but for single tenants, though with some modifications it
> can be used as multi-tenant solution too. But I don't like Reloading of it
> for after every single change, and prefer real-time approach.
>
> Zeeshan A Zakaria
>
> --
> Sent from my Android phone with K-9 Mail.
>
> On 2010-04-09 6:16 AM, "Chris Bagnall" <lists at minotaur.cc> wrote:
>
> Greetings list,
>
> I've traditionally been a proponent of the "manual configuration "
> approach,
> using .conf files and command lines to give us the greatest possible
> flexibility when writing call flows for customers. I've shied away from web
> interfaces as being overly restrictive and limiting what we can do for our
> customers.
>
> However, as we've grown as a company, taken on more customers (and hence
> more staff), there's become an ever-growing need for certain operations to
> be carried out by admin staff, rather than always having to be passed down
> to technical staff (who often have better things to do). I'm sure it's a
> problem faced by many companies on the list.
>
> So, what to do about it? Obviously there are "user-friendly" interfaces
> like
> FreePBX available, but they take over the *whole* asterisk config,
> shoehorning the user into their own fairly tight confines. Don't get me
> wrong, FreePBX is great as a company PBX installed on an on-site server,
> but
> it isn't much good as a VoIP hosting platform.
>
> What I think we're looking for is a fairly simple web interface to
> manipulate the tables used by Realtime. It doesn't have to be friendly. It
> doesn't have to be pretty. It just has to be easy enough for admin staff to
> use (with training, obviously) so that trivial call flow changes such as
> "please forward my calls to this mobile number" or "can you add extension
> 241 to this queue/ring group" can be made without having to involve
> technical staff.
>
> Would be very interested to hear what others in a similar position have
> done
> to overcome this growth problem. Did you write your own interface? Did you
> buy something off the shelf? Is there something in the FOSS marketplace
> that'll do the job?
>
> Regards,
>
> Chris
> --
> For full contact details visit http://www.minotaur.it
> This email is made from 100% recycled electrons
>
>
> --
> _____________________________________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
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> ------------------------------
>
> Message: 7
> Date: Fri, 9 Apr 2010 09:05:37 -0700
> From: Rick Orford <rick at comcanada.ca>
> Subject: Re: [asterisk-biz] Orlando DID's
> To: Commercial and Business-Oriented Asterisk Discussion
>        <asterisk-biz at lists.digium.com>
> Message-ID:
>        <i2xddd3f0de1004090905o8fdf54e1j48667d440141b89 at mail.gmail.com>
> Content-Type: text/plain; charset="iso-8859-1"
>
> Hi Luis,
>
> ComCanada can offer DIDs and port from Orlando FL.
>
> Contact me off list and I'll get you set up.
>
> --
> Regards,
>
> Rick Orford, Account Director
> ComCanada Communications Inc.
> www.comcanada.ca
> Tel/Fax: (604) 998-4500 x6008
>
> Customer Experience is very important to us.
> Please forward any feedback to my manager
> at andrea at comcanada.ca.
>
>
> On Thu, Apr 8, 2010 at 8:17 PM, Luis Mata <luism at ocntechnologies.com>
> wrote:
>
> > Does any one have Orlando FL DID's (area code 407) need info.
> >
> > When you send info , let me know if you also accept number portability.
> >
> > thank you
> >
> > Luis B. Mata
> > http://www.ocntechnologies.com
> > 954-889-8626
> >
> >
> > --
> > _____________________________________________________________________
> > -- Bandwidth and Colocation Provided by http://www.api-digital.com --
> >
> > asterisk-biz mailing list
> > To UNSUBSCRIBE or update options visit:
> >   http://lists.digium.com/mailman/listinfo/asterisk-biz
> >
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>
> _______________________________________________
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>
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> End of asterisk-biz Digest, Vol 69, Issue 8
> *******************************************
>



-- 
Kind regards,


Nauman Ibrahim Syed
Manager Carrier Relations and Fraud Prevention.
Great offer 1000 USA DIDs for 150$.
Didx special offer :http://www.youtube.com/watch?v=iew-LVDVbLs
Direct #:1-718-395-8986.
Fax: 1-206-339-4203.
Gtalk:ni at supertec.com <Gtalk%3Ani at supertec.com>
MSN:sales at supertec.com <MSN%3Asales at supertec.com>
Skype:salesdidx
Email:ni at supertec.com <Email%3Ani at supertec.com>
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