Not sure if anybody noticed, but it seems like Skype For Asterisk is out. $66 per channels, pretty pricey http://store.digium.com/productview.php?product_code=1SFA0001 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090817/cd8c6546/attachment.htm
I am using the beta and its pretty good for remote access for clients It would help if they had some discount structure for volume Cheers Duncan Pascal Bruno wrote:> Not sure if anybody noticed, but it seems like Skype For Asterisk is out. > > $66 per channels, pretty pricey > > http://store.digium.com/productview.php?product_code=1SFA0001 > ------------------------------------------------------------------------ > > _______________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > AstriCon 2009 - October 13 - 15 Phoenix, Arizona > Register Now: http://www.astricon.net > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users
On Mon, 17 Aug 2009, Pascal Bruno wrote:> Not sure if anybody noticed, but it seems like Skype For Asterisk is out. > > $66 per channels, pretty pricey > > http://store.digium.com/productview.php?product_code=1SFA0001Yes, pretty pricey indeed especially considering that you can buy Skype ATA adapters for the same amount (or less). But then again, who needs Skype for business purposes anyways, i don't think there is a huge market for it. I will add one channel to our PBX and will see if anybody will call us using skype.
I would have happily bought 20 channels at $10/channel, but at most will be buying only a single channel now :\ Pascal Bruno wrote:> Not sure if anybody noticed, but it seems like Skype For Asterisk is out. > > $66 per channels, pretty pricey > > http://store.digium.com/productview.php?product_code=1SFA0001 > > > ------------------------------------------------------------------------ > > _______________________________________________ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > AstriCon 2009 - October 13 - 15 Phoenix, Arizona > Register Now: http://www.astricon.net > > asterisk-users mailing list > To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users
Pricing is a very legitimate way to minimise support effort. It winnows down the market size to a point where the company offering the goods can sustain the projected per user support issues. You can always drop the price later on when you have a better handle on the per user support issue. Michael On Tue, 18 Aug 2009 10:06:25 -0500, Casey Boone wrote:>I would have happily bought 20 channels at $10/channel, but at most will >be buying only a single channel now :\ > > > >Pascal Bruno wrote: >> Not sure if anybody noticed, but it seems like Skype For Asterisk is out. >> >> $66 per channels, pretty pricey >> >> http://store.digium.com/productview.php?product_code=1SFA0001 >> >> >> ------------------------------------------------------------------------ >> >> _______________________________________________ >> -- Bandwidth and Colocation Provided by http://www.api-digital.com -- >> >> AstriCon 2009 - October 13 - 15 Phoenix, Arizona >> Register Now: http://www.astricon.net >> >> asterisk-users mailing list >> To UNSUBSCRIBE or update options visit: >> http://lists.digium.com/mailman/listinfo/asterisk-users > > >_______________________________________________ >-- Bandwidth and Colocation Provided by http://www.api-digital.com -- > >AstriCon 2009 - October 13 - 15 Phoenix, Arizona >Register Now: http://www.astricon.net > >asterisk-users mailing list >To UNSUBSCRIBE or update options visit: > http://lists.digium.com/mailman/listinfo/asterisk-users >-- Michael Graves mgraves<at>mstvp.com http://www.mgraves.org o713-861-4005 c713-201-1262 sip:mgraves at mstvp.onsip.com skype mjgraves Twitter mjgraves
On Tue, Aug 25, 2009 at 10:04 AM, Sanjoy Rath <sanjoy_rath at hotmail.com>wrote:> Hello, > > I am developing an asterisk autodialer. I am looking for the following > information: > > 1. Detailed Configuration Documentation for Asterisk Autodialer > 2. Volume Testing Strategy > 3. Lessons Learnt from past Asterisk Autodialer configuration > 4. What are the different asterisk autodialer functionality that have been > implemented > > Your response will be appreciated. > > Thanks, > Sanjoy. >How are you developing anything when you are asking for not just tips but EVERYTHING! Hire a consultant or go to that nerdvittles article. -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090825/0964525b/attachment.htm
Thanks Steve for helpful reply. Cheers, SR. Date: Tue, 25 Aug 2009 10:26:37 -0400 From: stotaro at first-notification.com To: asterisk-users at lists.digium.com Subject: Re: [asterisk-users] Asterisk Autodialer On Tue, Aug 25, 2009 at 10:04 AM, Sanjoy Rath <sanjoy_rath at hotmail.com> wrote: Hello, I am developing an asterisk autodialer. I am looking for the following information: 1. Detailed Configuration Documentation for Asterisk Autodialer 2. Volume Testing Strategy 3. Lessons Learnt from past Asterisk Autodialer configuration 4. What are the different asterisk autodialer functionality that have been implemented Your response will be appreciated. Thanks, Sanjoy. How are you developing anything when you are asking for not just tips but EVERYTHING! Hire a consultant or go to that nerdvittles article. -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) _________________________________________________________________ More storage. Better anti-spam and antivirus protection. Hotmail makes it simple. http://go.microsoft.com/?linkid=9671357 -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090825/6a8d87e2/attachment.htm
On Tue, Aug 25, 2009 at 1:29 PM, Alex Balashov <abalashov at evaristesys.com>wrote:> Sanjoy Rath wrote: > > > I would prefer to use AMI. Let me start looking into AMI. I would like > > to include functionalities like upload numbers to call from an > > interface, i want reports back numbers called, setup call time etc. Let > > me look up AMI. Thanks Alex for the info. > > Yep. If you need that level of detail, AMI is definitely the right > approach.I do not quite agree, I have developed a system exactly like that using call files, and I do have an interface to upload the numbers to call, you can setup call time, and the the delay to wait between each call. To me it was very straight forward and it works great, you can make a few thousands of calls a day depending on how many lines/channels you have. -------------- next part -------------- An HTML attachment was scrubbed... URL: http://lists.digium.com/pipermail/asterisk-users/attachments/20090825/f5ab9bc7/attachment.htm