Dear Luis, We can provide you 407 area code. Could you email me at [email protected] , so that I could reply you back with the quote. Regards,
On Fri, Apr 9, 2010 at 9:06 PM, <[email protected]>wrote: > Send asterisk-biz mailing list submissions to > [email protected] > > 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 > [email protected] > > You can reach the person managing the list at > [email protected] > > When replying, please edit your Subject line so it is more specific > than "Re: Contents of asterisk-biz digest..." > > > 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 <[email protected]> > Subject: [asterisk-biz] Dutch language recording > To: [email protected] > Message-ID: <[email protected]> > 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 [email protected] > 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 <[email protected]> > Subject: [asterisk-biz] Orlando DID's > To: [email protected] > Message-ID: > < > 32522430.35031270783034481.javamail.r...@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 <[email protected]> > Subject: [asterisk-biz] Friday Apr 9th 2010 @ 12 Noon EDT: More Cloud > Telephony > To: VOIP Users Conference <[email protected]>, > Asterisk Users Mailing List - Non-Commercial Discussion > <[email protected]>, Commercial and > Business-Oriented > Asterisk Discussion <[email protected]> > Message-ID: > <[email protected]> > 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:200...@login./zipdx.com is best for g722 wideband-capable phones > and accepts g711 as well > > You can also call sip:7463#[email protected] to connect to the > Talkshoe bridge. > > /r > > > > ------------------------------ > > Message: 4 > Date: Fri, 9 Apr 2010 11:09:00 +0100 > From: "Chris Bagnall" <[email protected]> > Subject: [asterisk-biz] The hunt for a workable Asterisk GUI > To: "'Commercial and Business-Oriented Asterisk Discussion'" > <[email protected]> > Message-ID: <0a3001cad7cc$ae6252e0$0b26f8...@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 <[email protected]> > Subject: Re: [asterisk-biz] The hunt for a workable Asterisk GUI > To: Commercial and Business-Oriented Asterisk Discussion > <[email protected]> > Message-ID: <[email protected]> > 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 <[email protected]> > Subject: Re: [asterisk-biz] The hunt for a workable Asterisk GUI > To: Commercial and Business-Oriented Asterisk Discussion > <[email protected]> > Message-ID: > <[email protected]> > 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" <[email protected]> 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 -- > > asterisk-biz mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-biz > -------------- next part -------------- > An HTML attachment was scrubbed... > URL: > http://lists.digium.com/pipermail/asterisk-biz/attachments/20100409/565cfeac/attachment-0001.htm > > ------------------------------ > > Message: 7 > Date: Fri, 9 Apr 2010 09:05:37 -0700 > From: Rick Orford <[email protected]> > Subject: Re: [asterisk-biz] Orlando DID's > To: Commercial and Business-Oriented Asterisk Discussion > <[email protected]> > Message-ID: > <[email protected]> > 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 [email protected]. > > > On Thu, Apr 8, 2010 at 8:17 PM, Luis Mata <[email protected]> > 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 > > > -------------- next part -------------- > An HTML attachment was scrubbed... > URL: > http://lists.digium.com/pipermail/asterisk-biz/attachments/20100409/6457336b/attachment.htm > > ------------------------------ > > _______________________________________________ > --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 > > 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:[email protected] <gtalk%[email protected]> MSN:[email protected] <msn%[email protected]> Skype:salesdidx Email:[email protected] <email%[email protected]>
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