Re: [asterisk-users] Skype For Asterisk (SFA): any replacement?
On Thursday 01 December 2011, Hans Witvliet wrote: On Thu, 2011-12-01 at 14:02 +, A J Stiles wrote: On Thursday 01 December 2011, gincantalupo wrote: Hi all, any idea about how to replace Skype For Asterisk? Thank You. Giorgio 1. Migrate your Skype users over to a better product which supports proper open standards. perhaps you missed it, but the installed base of skype is unfortunately slightly (,,,) larger than the amount of peope that are using a decent product. Alas Then it's simply a bigger job than the original suggestion made it seem. When -- not if -- Skype give up supporting their anti-telecommunications product altogether, every single one of those users is going to be left in the lurch. And that might be the critical mass that brings on the revolution. We can only hope :) 2. Write to your elected representatives asking that they order Skype to release documentation on their protocols to allow third party interoperability (as is already required under EU law). 3. make it a offence to use any closed source products like skype. ;-) Huge fines, jail centences or worse. [How about an appendice to the Thora, Quran or Bible, even better, forbid it by the sharia] You may jest, but now you are seeing *EXACTLY* why closed, proprietary standards are a bad idea -- something I have been saying almost ever since Skype was first launched. Note, not necessarily closed *source*, but closed *standards*. The two are easily confused, but not quite the same. An Open Source program can only ever implement open standards, since the Source Code implicitly documents the standards. But Closed Source programs can, and often do, implement open standards. And wherever they do, then there are usually alternative, Open Source programs that do the same job. Every aspect of a program's interaction with the outside world -- communications protocols, save file formats and similar -- must be documented to the point where any competent programmer could write a program which interacts seamlessly with the application that originally generated them. That documentation may well be the Source Code for the program itself, of course; or it could just be something like the RFCs -- in which case, the will is surely out there for someone within the Open Source community to do the rest. Anything less is just blatant anti-competitive behaviour. -- AJS Answers come *after* questions. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Skype For Asterisk (SFA): any replacement?
Hi all, any idea about how to replace Skype For Asterisk? Thank You. Giorgio -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Skype For Asterisk (SFA): any replacement?
On 12/01/2011 08:30 AM, gincantalupo wrote: any idea about how to replace Skype For Asterisk? Replace with what? -- Alex Balashov - Principal Evariste Systems LLC 260 Peachtree Street NW Suite 2200 Atlanta, GA 30303 Tel: +1-678-954-0670 Fax: +1-404-961-1892 Web: http://www.evaristesys.com/ -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Skype For Asterisk (SFA): any replacement?
On Thursday 01 December 2011, gincantalupo wrote: Hi all, any idea about how to replace Skype For Asterisk? Thank You. Giorgio 1. Migrate your Skype users over to a better product which supports proper open standards. 2. Write to your elected representatives asking that they order Skype to release documentation on their protocols to allow third party interoperability (as is already required under EU law). -- AJS Answers come *after* questions. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Skype For Asterisk (SFA): any replacement?
On Thu, Dec 1, 2011 at 8:30 AM, gincantalupo gincantal...@fgasoftware.com wrote: Hi all, any idea about how to replace Skype For Asterisk? Thank You. Giorgio We are going through this right now and have chosen to Pay The Man via per channel subscription to Skype Connect. Watch the fun video at: http://www.skype.com/intl/en/business/skype-connect/ :-) Skype-For-Asterisk is a vastly superior product/service but someone at Skype woke up one day and said, Hey we can't let that product succeed and lose control of some valuable fees. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Skype For Asterisk (SFA): any replacement?
On Thu, 2011-12-01 at 14:02 +, A J Stiles wrote: On Thursday 01 December 2011, gincantalupo wrote: Hi all, any idea about how to replace Skype For Asterisk? Thank You. Giorgio 1. Migrate your Skype users over to a better product which supports proper open standards. perhaps you missed it, but the installed base of skype is unfortunately slightly (,,,) larger than the amount of peope that are using a decent product. Alas 2. Write to your elected representatives asking that they order Skype to release documentation on their protocols to allow third party interoperability (as is already required under EU law). 3. make it a offence to use any closed source products like skype. ;-) Huge fines, jail centences or worse. [How about an appendice to the Thora, Quran or Bible, even better, forbid it by the sharia] -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Skype For Asterisk (SFA): any replacement?
Hi Alex, replace with anything which could make Asterisk connect to Skype network, make and receive calls, etc...the usual stuff. Giorgio On 12/01/2011 02:40 PM, Alex Balashov wrote: On 12/01/2011 08:30 AM, gincantalupo wrote: any idea about how to replace Skype For Asterisk? Replace with what? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Skype For Asterisk (SFA)
Dear Abdul Basit, http://nerdvittles.com/index.php?p=784 works, I tested it few months back and it works. Cant say if its still working or not. On Wed, Nov 16, 2011 at 5:01 PM, Abdul Basit basit.e...@gmail.com wrote: Any has Skype For Asterisk (SFA) license. http://www.digium.com/en/products/software/skypeforasterisk.php PLEASE NOTE: Skype for Asterisk is no longer available for sale. Skype for Asterisk will be supported for two more years, until July 26, 2013. I want to test this thing. Any Idea. any free solution. there is one http://nerdvittles.com/index.php?p=784 Tying to test but dont know if its workable or not. I will appreciate if any one can share his testing/implementation. -- Regards, Abdul Basit | +92 32 1416 4196 | +92 30 0841 1445 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Thanks Regards, Umair Bari -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Skype For Asterisk (SFA)
Any has Skype For Asterisk (SFA) license. http://www.digium.com/en/products/software/skypeforasterisk.php PLEASE NOTE: Skype for Asterisk is no longer available for sale. Skype for Asterisk will be supported for two more years, until July 26, 2013. I want to test this thing. Any Idea. any free solution. there is one http://nerdvittles.com/index.php?p=784 Tying to test but dont know if its workable or not. I will appreciate if any one can share his testing/implementation. -- Regards, Abdul Basit | +92 32 1416 4196 | +92 30 0841 1445 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Skype For Asterisk (SFA)
Yes, Skype was a good thing. R.I.P On Wed, Nov 16, 2011 at 5:01 PM, Abdul Basit basit.e...@gmail.com wrote: Any has Skype For Asterisk (SFA) license. http://www.digium.com/en/products/software/skypeforasterisk.php PLEASE NOTE: Skype for Asterisk is no longer available for sale. Skype for Asterisk will be supported for two more years, until July 26, 2013. I want to test this thing. Any Idea. any free solution. there is one http://nerdvittles.com/index.php?p=784 Tying to test but dont know if its workable or not. I will appreciate if any one can share his testing/implementation. -- Regards, Abdul Basit | +92 32 1416 4196 | +92 30 0841 1445 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Skype For Asterisk (SFA)
I can tell you that siptosis is workable but the support has been dropped recently as well. It is a great program and especially the paid version with trunk builder i.e. you can have multiple skype instances On Wed, Nov 16, 2011 at 8:01 PM, Abdul Basit basit.e...@gmail.com wrote: Any has Skype For Asterisk (SFA) license. http://www.digium.com/en/products/software/skypeforasterisk.php PLEASE NOTE: Skype for Asterisk is no longer available for sale. Skype for Asterisk will be supported for two more years, until July 26, 2013. I want to test this thing. Any Idea. any free solution. there is one http://nerdvittles.com/index.php?p=784 Tying to test but dont know if its workable or not. I will appreciate if any one can share his testing/implementation. -- Regards, Abdul Basit | +92 32 1416 4196 | +92 30 0841 1445 -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Skype For Asterisk (SFA)
On Wednesday 16 November 2011, Abdul Basit wrote: Any has Skype For Asterisk (SFA) license. http://www.digium.com/en/products/software/skypeforasterisk.php PLEASE NOTE: Skype for Asterisk is no longer available for sale. Skype for Asterisk will be supported for two more years, until July 26, 2013. I want to test this thing. Any Idea. any free solution. there is one http://nerdvittles.com/index.php?p=784 Tying to test but dont know if its workable or not. I will appreciate if any one can share his testing/implementation. You would be better off persuading Skype users to transition to something else. Skype is the absolute antithesis of the whole point of telephony, which is to connect people together. This includes, implicitly, the ability for subscribers on one telecommunications provider's network to call subscribers on another network. Imagine if, say, Vodafone subscribers were unable to call up BT subscribers? Well, this is *exactly* what Skype are trying to create by keeping their protocols proprietary. Of course there remains a small but finite probability that Skype will be successfully reverse-engineered, the Source Code leaked, or Skype's owners forced to publish its communications protocols before the 2013 deadline. But it would be extreme folly to bet the family farm on this happening. It's time to start seriously evaluating Asterisk-compatible alternatives to Skype. -- AJS Answers come *after* questions. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Skype For Asterisk (SFA)
On Wed, 16 Nov 2011, A J Stiles wrote: You would be better off persuading Skype users to transition to something else. Skype is the absolute antithesis of the whole point of telephony, which is to connect people together. This includes, implicitly, the ability for subscribers on one telecommunications provider's network to call subscribers on another network. Imagine if, say, Vodafone subscribers were unable to call up BT subscribers? Well, this is *exactly* what Skype are trying to create by keeping their protocols proprietary. Of course there remains a small but finite probability that Skype will be successfully reverse-engineered, the Source Code leaked, or Skype's owners forced to publish its communications protocols before the 2013 deadline. But it would be extreme folly to bet the family farm on this happening. It's time to start seriously evaluating Asterisk-compatible alternatives to Skype. Sadly, my experience in the SOHO environment is that Skype wins. I tried to get my family to all use SIP videophones - and it worked for a couple of years - mostly. The downside was that they're mostly using crap domestic quality broadband and trying to use a videophone, or even a soft-phone on a PC just seemed too hard for them to grasp. They *all* moved to Skype recently - and I have to say I've been totally blown away at the ease of use and the quality of the calls - both sound and video. (And I'm using Linux too) The other thing - LAN to LAN calls STAY ON THE LAN! So I can Skype my wife next door and it doesn't use up any of my own broadband bandwidth wheras if I use a hosted SIP service, calls go out come back in again. Skype also seems to be able to run the lines at max. rate too - some sort of adaptive bandwidth - we get large and high resolution video calls from one end of the country to the other with the output bandwidth running at near max (800Kb sec in our case) And now I'm seeing some of my smaller business customers using Skype. For serious business calls too. It's free. They get video. It just works. No fiddling with NAT, port forwarding, never any hint of one-way audio. I really was skeptical at first, but Skype is here to stay - mostly because it just works. Even a complete computer idiot can install it and make it work. Give them a SIP phone, or SIP softphone and tell them to set it up and they'll just leave it alone as too complicated. As for interoerability - well there's Skype-Out. It works, it's set at a reasonable price level, so what more do you need? Once upon a time I would block Skype from working inside a corporate LAN and would recomend against it's use - now I'm told to explicitly allow it. Times are changing and I'm finding it harder to persuade small businesses to use SIP phones - and why should they... Gordon -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Skype For Asterisk (SFA)
I would agree, unfortunately. However, I still see it as a glorified webcam chat and not a telecommunication device like a SIP/soft phone. -Original Message- From: asterisk-users-boun...@lists.digium.com [mailto:asterisk-users-boun...@lists.digium.com] On Behalf Of Gordon Henderson Sent: Wednesday, November 16, 2011 11:45 AM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Skype For Asterisk (SFA) On Wed, 16 Nov 2011, A J Stiles wrote: You would be better off persuading Skype users to transition to something else. Skype is the absolute antithesis of the whole point of telephony, which is to connect people together. This includes, implicitly, the ability for subscribers on one telecommunications provider's network to call subscribers on another network. Imagine if, say, Vodafone subscribers were unable to call up BT subscribers? Well, this is *exactly* what Skype are trying to create by keeping their protocols proprietary. Of course there remains a small but finite probability that Skype will be successfully reverse-engineered, the Source Code leaked, or Skype's owners forced to publish its communications protocols before the 2013 deadline. But it would be extreme folly to bet the family farm on this happening. It's time to start seriously evaluating Asterisk-compatible alternatives to Skype. Sadly, my experience in the SOHO environment is that Skype wins. I tried to get my family to all use SIP videophones - and it worked for a couple of years - mostly. The downside was that they're mostly using crap domestic quality broadband and trying to use a videophone, or even a soft-phone on a PC just seemed too hard for them to grasp. They *all* moved to Skype recently - and I have to say I've been totally blown away at the ease of use and the quality of the calls - both sound and video. (And I'm using Linux too) The other thing - LAN to LAN calls STAY ON THE LAN! So I can Skype my wife next door and it doesn't use up any of my own broadband bandwidth wheras if I use a hosted SIP service, calls go out come back in again. Skype also seems to be able to run the lines at max. rate too - some sort of adaptive bandwidth - we get large and high resolution video calls from one end of the country to the other with the output bandwidth running at near max (800Kb sec in our case) And now I'm seeing some of my smaller business customers using Skype. For serious business calls too. It's free. They get video. It just works. No fiddling with NAT, port forwarding, never any hint of one-way audio. I really was skeptical at first, but Skype is here to stay - mostly because it just works. Even a complete computer idiot can install it and make it work. Give them a SIP phone, or SIP softphone and tell them to set it up and they'll just leave it alone as too complicated. As for interoerability - well there's Skype-Out. It works, it's set at a reasonable price level, so what more do you need? Once upon a time I would block Skype from working inside a corporate LAN and would recomend against it's use - now I'm told to explicitly allow it. Times are changing and I'm finding it harder to persuade small businesses to use SIP phones - and why should they... Gordon -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello 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 -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Skype For Asterisk (SFA)
On 11/16/2011 10:44 AM, Gordon Henderson wrote: The other thing - LAN to LAN calls STAY ON THE LAN! So I can Skype my wife next door and it doesn't use up any of my own broadband bandwidth wheras if I use a hosted SIP service, calls go out come back in again. Skype also seems to be able to run the lines at max. rate too - some sort of adaptive bandwidth - we get large and high resolution video calls from one end of the country to the other with the output bandwidth running at near max (800Kb sec in our case) As SIP endpoints (servers, phones, etc.) get upgraded to support the ICE NAT traversal mechanism, this will start happening for regular SIP calls as well. This *should* already happen with the Blink softphone, for example, since it fully supports ICE. Also note that you are using the term 'calls' when you really mean 'media streams'; in all of the cases you outlined, the 'call' signaling still follows the same path it did originally, but the media stream path can be shortened if the two endpoints are able to exchange media directly. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies Jabber: kflem...@digium.com | SIP: kpflem...@digium.com | Skype: kpfleming 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA Check us out at www.digium.com www.asterisk.org -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Skype For Asterisk (SFA)
On Wed, Nov 16, 2011 at 8:56 AM, Kevin P. Fleming kpflem...@digium.comwrote: As SIP endpoints (servers, phones, etc.) get upgraded to support the ICE NAT traversal mechanism, this will start happening for regular SIP calls as well. This *should* already happen with the Blink softphone, for example, since it fully supports ICE. Hi Kevin, Just curious on when we should expect to see the manufactures get on board with the ICE NAT? Does any particular manufacture stand out in implementing ICE NAT in their endpoints currently? Also what is Digium doing to promote it? -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Skype For Asterisk (SFA)
On Wednesday 16 November 2011, Gordon Henderson wrote: On Wed, 16 Nov 2011, A J Stiles wrote: You would be better off persuading Skype users to transition to something else. Sadly, my experience in the SOHO environment is that Skype wins. [stuff deleted] And now I'm seeing some of my smaller business customers using Skype. For serious business calls too. It's free. They get video. It just works. No fiddling with NAT, port forwarding, never any hint of one-way audio. [stuff deleted] As for interoerability - well there's Skype-Out. It works, it's set at a reasonable price level, so what more do you need? I need a rock-solid guarantee that nobody can pick up the ball and go home, leaving all former users effectively stranded. A single-vendor proprietary solution is a *massive* single-point failure. Multiple, competing but mutually-compatible proprietary solutions slightly less so. If there is even just one Open Source implementation out there, then this sort of thing can never happen. Once upon a time I would block Skype from working inside a corporate LAN and would recomend against it's use - now I'm told to explicitly allow it. Nothing goes near my company's LAN without Source Code. I figure if they don't want you to see it, there must be something in it that they wouldn't expect you to like it if you saw it. The level of paranoia on Skype's part only reinforces that impression. -- AJS Answers come *after* questions. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
