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
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): 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?
On Thu, Dec 1, 2011 at 8:30 AM, gincantalupo 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 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 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)
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 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
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
Re: [asterisk-users] Skype For Asterisk (SFA)
On Wed, Nov 16, 2011 at 8:56 AM, Kevin P. Fleming wrote: 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 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)
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 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)
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)
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 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)
Yes, Skype was a good thing. R.I.P On Wed, Nov 16, 2011 at 5:01 PM, 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. > > -- > 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 usage in the future
hi: thanks for all the information. I don't use skype and I ban skype at our network. but there are some people who use skype and want us to use skype to contact them. SFA is my saver because our users can use their phone to talk with skype users and no need to install any skype software. I hope skype die asap. but if it is alive, I must find someway to satisfy these skype customers... 2011/7/12 A J Stiles : > On Tuesday 12 Jul 2011, d tbsky wrote: >> hi: >> I am a SFA (skype for asterisk) user. I had ask Digium questions >> about SFA usage in the future. but they seem too busy to reply. so I >> tried at this list. I hope there are SFA users or Digium people can >> solve my confusion. > > Poor you! > > To my mind, Skype with its opaque, proprietary protocols is the exact opposite > of what telecommunications is supposed to be about. > > I can make a call, or send an SMS, from my HTC on Vodafone to my friend's > Samsung on Tesco without thinking twice about it, and that's the way we all > expect it to be. But if it hadn't been for governments enforcing standards, > the mobile networks could well have ended up fragmentated; with different > handset manufacturers and different network operators all using competing, > proprietary standards to lock one another out and their customers in. > >> 2. I saw SFA will not be supported after two years. my question is: >> although it is not supported, can I still use it? I want to buy more >> licenses now if I can still use it after two years even without official >> support. > > That depends entirely on whether Skype update their protocols and block out > the old ones. Two years is easily long enough for them to do that; > especially given the way Skype works. It could even do stealth upgrades in > the background. > > Look at it this way: You've got two years to migrate away from Skype and > start using something else -- and this time around, for the love of all > that's sane and wholesome, be sure to choose something that supports open > standards, so you can never get shafted the same way again. > > If someone manages successfully to reverse-engineer Skype during that time, > you *might* have a little longer; but I wouldn't bet the family farm on that. > > > -- > 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 > -- _ -- 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 usage in the future
On Tuesday 12 Jul 2011, d tbsky wrote: > hi: >I am a SFA (skype for asterisk) user. I had ask Digium questions > about SFA usage in the future. but they seem too busy to reply. so I > tried at this list. I hope there are SFA users or Digium people can > solve my confusion. Poor you! To my mind, Skype with its opaque, proprietary protocols is the exact opposite of what telecommunications is supposed to be about. I can make a call, or send an SMS, from my HTC on Vodafone to my friend's Samsung on Tesco without thinking twice about it, and that's the way we all expect it to be. But if it hadn't been for governments enforcing standards, the mobile networks could well have ended up fragmentated; with different handset manufacturers and different network operators all using competing, proprietary standards to lock one another out and their customers in. > 2. I saw SFA will not be supported after two years. my question is: > although it is not supported, can I still use it? I want to buy more > licenses now if I can still use it after two years even without official > support. That depends entirely on whether Skype update their protocols and block out the old ones. Two years is easily long enough for them to do that; especially given the way Skype works. It could even do stealth upgrades in the background. Look at it this way: You've got two years to migrate away from Skype and start using something else -- and this time around, for the love of all that's sane and wholesome, be sure to choose something that supports open standards, so you can never get shafted the same way again. If someone manages successfully to reverse-engineer Skype during that time, you *might* have a little longer; but I wouldn't bet the family farm on that. -- 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 usage in the future
On 07/12/2011 08:26 AM, Kevin P. Fleming wrote: > > It is unknown whether it will continue to be usable after that period; > Skype has the ability to disable SFA from accessing the Skype network > if they feel that is what they want to do. Since it won't get any > updates between now and then, it is very likely to be obsolete (from a > 'Skype protocol' point of view) in two years and it seems quite likely > that they won't want it accessing the network any more. It would be > best to plan for it being non-functional after the two year support > period is over. > Is there a project to replace Skype with a free software? Bob Rawlinson -- _ -- 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 usage in the future
On 07/11/2011 09:48 PM, d tbsky wrote: 1. SFA can not be registered after 26 July. so I want to prepare a backup machine for our server. I read in the document that I can re-register my SFA once. so I want to make sure if I can re-register with my backup server now, and in the same time my production machine still function correctly. and if my production machine is broken one day, my backup machine can go on line. in short: can I keep two machines with one license now? I hope so because I can not re-register later after 26 July. Yes, this is acceptable. 2. I saw SFA will not be supported after two years. my question is: although it is not supported, can I still use it? I want to buy more licenses now if I can still use it after two years even without official support. It is unknown whether it will continue to be usable after that period; Skype has the ability to disable SFA from accessing the Skype network if they feel that is what they want to do. Since it won't get any updates between now and then, it is very likely to be obsolete (from a 'Skype protocol' point of view) in two years and it seems quite likely that they won't want it accessing the network any more. It would be best to plan for it being non-functional after the two year support period is over. -- 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 usage in the future
Will you consider alternatives such as siptosis? The uncertainties are really there for SFA CK Lee On 12 Jul, 2011, at 10:48 AM, d tbsky wrote: > hi: > I am a SFA (skype for asterisk) user. I had ask Digium questions > about SFA usage in the future. but they seem too busy to reply. so I > tried at this list. I hope there are SFA users or Digium people can > solve my confusion. > > 1. SFA can not be registered after 26 July. so I want to prepare a > backup machine for our server. I read in the document that I can > re-register my SFA once. so I want to make sure if I can re-register with > my backup server now, and in the same time my production machine still > function correctly. and if my production machine is broken one day, my > backup machine can go on line. in short: can I keep two machines with one > license now? I hope so because I can not re-register later after 26 July. > > 2. I saw SFA will not be supported after two years. my question is: > although it is not supported, can I still use it? I want to buy more > licenses now if I can still use it after two years even without official > support. > > thanks a lot for your kindly help!! > > Regards, > tbskyd > > -- > _ > -- 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 - RIP
On Wed, May 25, 2011 at 10:53 AM, A J Stiles wrote: > Bgeh. Serves 'em right for using that POC! Who honestly *hadn't* seen this > coming since the day Skype was first released? Tim Panton, who's beenworking with SfA since it came out, posted this article today: http://vuc.li/meBRJd /r -- _ -- 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 - RIP
On Wednesday 25 May 2011, randulo wrote: > On Tue, May 24, 2011 at 10:50 PM, Matt Darnell wrote: > > "We expect that users of Skype for Asterisk will be able to continue > > using their Asterisk systems on the Skype network until at least July > > 26, 2013. Skype may extend this at their discretion." > > It's widely believed. However, it's very possible that this was not a > Microsoft decision but planned by Skype before the acquisition. Bgeh. Serves 'em right for using that POC! Who honestly *hadn't* seen this coming since the day Skype was first released? A subscriber with one telephone company has to be able at least to contact subscribers with any other telephone company -- that much ought to be self-evident. It is also highly desirable, where multiple telephone companies are competing for business in the same physical space, to be able to use the same equipment with any of them. Such interoperability requires open standards that can be implemented by anybody, and the most preferable way to achieve this is through an Open Source reference implementation (not just Asterisk; think OpenBSD and the Secure Shell, or Apache and HTTP). Skype's secretive, proprietary nature -- surely the absolute antithesis of what telecommunications needs to be about -- means that only Skype "subscribers" can talk to other Skype "subscribers". (It also potentially runs afoul of some European countries' telecommunications deregulation and competition laws -- except, as we all know, normal laws don't apply anywhere there is a computer involved). We in the Asterisk user community should be persuading Skype users to move to "proper" VoIP solutions, sooner rather than later -- even if that means recommending another proprietary product as a pragmatic intermediate measure. At least Caged products which *correctly* implement Open standards are likely candidates for Free drop-in replacements later. (Paraphrased and expanded from an earlier post by me on another forum.) -- 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 - RIP
On Tue, May 24, 2011 at 10:50 PM, Matt Darnell wrote: > "We expect that users of Skype for Asterisk will be able to continue > using their Asterisk systems on the Skype network until at least July > 26, 2013. Skype may extend this at their discretion." It's widely believed. However, it's very possible that this was not a Microsoft decision but planned by Skype before the acquisition. /r -- _ -- 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, Skype For SIP
--- On Mon, 7/19/10, Kevin P. Fleming wrote: > Usage of the standard Skype client is not "free"; it > involves acting as > part of the peer-to-peer Skype network > The Skype > business solutions (including Skype For Asterisk) don't > participate in > the peer-to-peer network > Any solution that uses a regular Skype client will be > limited to one > call at a time; Thanks for the explanation! It's crystal-clear now. Vieri -- _ -- 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, Skype For SIP
I have been using SiSky Enterprise Edition to integrate Skype with asterisk. You can even call saved skype users from your asterisk system, by creating speed dials in SiSky. Unfortunately it is not a free product but it is very reasonable. Thank you, Brad Finberg - Original Message - From: Alejandro Imass To: Asterisk Users Mailing List - Non-Commercial Discussion Cc: Date: Sunday, July 18 2010 8:57 AM Subject: Re: [asterisk-users] Skype for Asterisk, Skype For SIP On Sun, Jul 18, 2010 at 7:48 AM, Vieri wrote: > Hi, > > I'm trying to integrate Skype and Asterisk but I'm only interested in these 2 > things: > > 1) allow any Asterisk SIP extension to call any Skype "user". I do not need > to call landlines via Skype. > I think this is _explicitly_ not supported in the Skype for SIP docs. > 2) allow Internet Skype "users" to call my Asterisk PBX Skype "user" and > route the call to a specific Asterisk SIP extension. > Here is how it goes from my experience with Skype: each SIP channel will cost you about $5 a month, regardless if you have a landline number with them or not. Your account will be assigned a special Skype number 99x . With that number a Skype user can call you and it will be free. You _cannot_ call Skype users from your PBX, as I stated above, this is an explicit no-no in the docs. If you want to make calls from your PBX to landlines you have to buy Skype credit just like you would with a regular skype client. If you want land-lines to call your PBX you need to purchase a skype number which about $60 a year. > At first, I thought it would be simple and free. However, correct me if I'm > wrong but the Skype "user" I can use within the Asterisk PBX cannot be the > "standard type" (used by eg. desktop Skype applications) but needs to be > created by the Skype User Manager for Business Solutions. I believe this has > a price although Skype For SIP Open Beta seems to be free until Q4 2010. I think you can associate existing skype users to your Business Solutions manager but I still don't understand exactly how or why this is useful, and I don't think it has to do with you being able to call any of them from your PBX. Then again I haven't paid much attention to that and perhaps you have more insight into this. > > Has anyone found a way to make "pure Internet user-to-user" Skype/SIP calls > via Asterisk (no PSTN involved) for free? As I said above, once you have purchased your SIP channel you can make free calls to your PBX using the special number but it's only INBOUND AFAIK. Best, Alejandro Imass > > Thanks, > > Vieri > > > > > > -- > _ > -- 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-- _ -- 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, Skype For SIP
On 07/18/2010 12:18 PM, Steve Kennedy wrote: > On Sun, Jul 18, 2010 at 09:56:30AM -0700, Vieri wrote: > >>> As I said above, once you have purchased your SIP channel >>> you can make >>> free calls to your PBX using the special number but it's >>> only INBOUND >>> AFAIK. > [lots snipped] > > With Skype's just released SkypeKit it should be possible to build > any application with Skype support (costs $20 to register as a dev), > they've now got libraries for Linux and now Windows and MacOS X. > > SkypeKit is basically a headless Skype client. SkypeKit is currently single-user and single-call, just like the regular Skype client. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kflem...@digium.com 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, Skype For SIP
On 07/18/2010 11:56 AM, Vieri wrote: > I still don't see why one should pay for a channel when using a PBX but not > when using a client such as Skype. OK, I know that the Skype network is > proprietary and I have to accept whatever they say. Usage of the standard Skype client is not "free"; it involves acting as part of the peer-to-peer Skype network and helping to route calls and in some cases even helping to route media streams for calls. The Skype business solutions (including Skype For Asterisk) don't participate in the peer-to-peer network in this fashion, so every single user of these products does in fact increase the burden on Skype's own network resources. Their solution to this issue is to charge a nominal fee for access to the network. For Skype For Asterisk, calls are still free, and there is no per-channel charge, only a per-user charge (when it begins). This means that for a one-user cost per month, you can receive dozens of simultaneous calls from the Skype network into your Asterisk system. > However, if a "standard" user can call and receive for free then there should > be a way to do it from a PBX such as Asterisk. > > In fact, I came across this project: > http://www.mhspot.com/sts/siptosis.html > > It seems to be a bit of a "hack" in that it integrates a SIP PBX with a > standard Skype client (which doesn't necessarily have to be on the same > machine or same OS...). In short, one can use a standard Skype account and > not pay a cent for user-to-user calls. Any solution that uses a regular Skype client will be limited to one call at a time; the regular Skype client is not multi-user, and does not support multiple calls (calls can be placed on hold, but there cannot be more than one active call). If this suits your needs, you can certainly try it. There are other Skype gateway solutions that use a similar method, but they are not free. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kflem...@digium.com 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, Skype For SIP
On Sun, Jul 18, 2010 at 09:56:30AM -0700, Vieri wrote: > > As I said above, once you have purchased your SIP channel > > you can make > > free calls to your PBX using the special number but it's > > only INBOUND > > AFAIK. [lots snipped] With Skype's just released SkypeKit it should be possible to build any application with Skype support (costs $20 to register as a dev), they've now got libraries for Linux and now Windows and MacOS X. SkypeKit is basically a headless Skype client. Steve -- NetTek Ltd UK mob +44 7775 755503 UK +44 20 7993 2612 / US +1 310 857 7715 / Fax +44 20 7483 2455 Skype/GoogleTalk/AIM/Gizmo/.Mac/Twitter/FriendFeed stevekennedyuk Euro Tech News Blog http://eurotechnews.blogspot.com MSN st...@gbnet.net -- _ -- 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, Skype For SIP
--- On Sun, 7/18/10, Alejandro Imass wrote: > > Hi, > > > > I'm trying to integrate Skype and Asterisk but I'm > only interested in these 2 things: > > > > 1) allow any Asterisk SIP extension to call any Skype > "user". I do not need to call landlines via Skype. > > > > I think this is _explicitly_ not supported in the Skype for > SIP docs. > > > 2) allow Internet Skype "users" to call my Asterisk > PBX Skype "user" and route the call to a specific Asterisk > SIP extension. > > > > Here is how it goes from my experience with Skype: each SIP > channel > will cost you about $5 a month, regardless if you have a > landline > number with them or not. Your account will be assigned a > special Skype > number 99x . With that number a Skype user can > call you > and it will be free. You _cannot_ call Skype users from > your PBX, as I > stated above, this is an explicit no-no in the docs. If you > want to > make calls from your PBX to landlines you have to buy Skype > credit > just like you would with a regular skype client. If you > want > land-lines to call your PBX you need to purchase a skype > number which > about $60 a year. > > > > At first, I thought it would be simple and free. > However, correct me if I'm wrong but the Skype "user" I can > use within the Asterisk PBX cannot be the "standard type" > (used by eg. desktop Skype applications) but needs to be > created by the Skype User Manager for Business Solutions. I > believe this has a price although Skype For SIP Open Beta > seems to be free until Q4 2010. > > I think you can associate existing skype users to your > Business > Solutions manager but I still don't understand exactly how > or why this > is useful, and I don't think it has to do with you being > able to call > any of them from your PBX. Then again I haven't paid much > attention to > that and perhaps you have more insight into this. > > > > Has anyone found a way to make "pure Internet > user-to-user" Skype/SIP calls via Asterisk (no PSTN > involved) for free? > > As I said above, once you have purchased your SIP channel > you can make > free calls to your PBX using the special number but it's > only INBOUND > AFAIK. Thanks Alejandro, I still don't see why one should pay for a channel when using a PBX but not when using a client such as Skype. OK, I know that the Skype network is proprietary and I have to accept whatever they say. However, if a "standard" user can call and receive for free then there should be a way to do it from a PBX such as Asterisk. In fact, I came across this project: http://www.mhspot.com/sts/siptosis.html It seems to be a bit of a "hack" in that it integrates a SIP PBX with a standard Skype client (which doesn't necessarily have to be on the same machine or same OS...). In short, one can use a standard Skype account and not pay a cent for user-to-user calls. Can chan_skype do that? (it doesn't seem to) Has anyone tried SipToSis? Thanks, Vieri -- _ -- 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, Skype For SIP
On Sun, Jul 18, 2010 at 7:48 AM, Vieri wrote: > Hi, > > I'm trying to integrate Skype and Asterisk but I'm only interested in these 2 > things: > > 1) allow any Asterisk SIP extension to call any Skype "user". I do not need > to call landlines via Skype. > I think this is _explicitly_ not supported in the Skype for SIP docs. > 2) allow Internet Skype "users" to call my Asterisk PBX Skype "user" and > route the call to a specific Asterisk SIP extension. > Here is how it goes from my experience with Skype: each SIP channel will cost you about $5 a month, regardless if you have a landline number with them or not. Your account will be assigned a special Skype number 99x . With that number a Skype user can call you and it will be free. You _cannot_ call Skype users from your PBX, as I stated above, this is an explicit no-no in the docs. If you want to make calls from your PBX to landlines you have to buy Skype credit just like you would with a regular skype client. If you want land-lines to call your PBX you need to purchase a skype number which about $60 a year. > At first, I thought it would be simple and free. However, correct me if I'm > wrong but the Skype "user" I can use within the Asterisk PBX cannot be the > "standard type" (used by eg. desktop Skype applications) but needs to be > created by the Skype User Manager for Business Solutions. I believe this has > a price although Skype For SIP Open Beta seems to be free until Q4 2010. I think you can associate existing skype users to your Business Solutions manager but I still don't understand exactly how or why this is useful, and I don't think it has to do with you being able to call any of them from your PBX. Then again I haven't paid much attention to that and perhaps you have more insight into this. > > Has anyone found a way to make "pure Internet user-to-user" Skype/SIP calls > via Asterisk (no PSTN involved) for free? As I said above, once you have purchased your SIP channel you can make free calls to your PBX using the special number but it's only INBOUND AFAIK. Best, Alejandro Imass > > Thanks, > > Vieri > > > > > > -- > _ > -- 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 - what processors/platforms does it run on?
Hi! > I understand that SfA is a binary module? There are processors it will not > work on, correct? Are there limits as to operating system or distros? Requires kernel 2.6, does not work on 2.4 - as I had to find out the hard way (this is not documented - so embedded people: be aware!). Philipp -- _ -- 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
On Wed, 30 Dec 2009 11:43:59 -0800, vijay.go...@alliance-infotech.com wrote: > >case 2: This skype account (rexesbposolutions) has been assigned with a >online virtual number (00 44 20 ). If somebody dial this number >from their landline/cellphone, call is transfered to Asterisk queue but >it shows some problem related to G729 codecs. following are Asterisk CLI >log: I had the same problem. I contacted Digium support and this was the answer: You can download the G.729 codec from the following link: http://www.digium.com/en/docs/G729/g729-download.php Install the codec_g729a.so binary in /usr/lib/asterisk/modules/ and restart the asterisk service. I followed the advice and the problem is resolved. You can try. Bye -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Skype for Asterisk
Dear,
there is a problem in codec translation..so change the ulaw codec to
g729. .if problem persist then u must have same codex on asterisk server and
clients (skype)...
On Mon, Jan 4, 2010 at 11:24 AM, Tim Panton wrote:
>
> On 30 Dec 2009, at 19:43, vijay.go...@alliance-infotech.com wrote:
>
>
> Hi Sir,
>
> We have integrated Skype with Asterisk (skype user id:- rexesbposolutions).
> Each call which is coming to skype account is getting transfered to Asterisk
> Queue. It has following two cases:
>
> case 1: When we call from normal skype account to skype account
> (rexesbposolutions), everything is working fine.
>
> case 2: This skype account (rexesbposolutions) has been assigned with a
> online virtual number (00 44 20 ). If somebody dial this number
> from their landline/cellphone, call is transfered to Asterisk queue but it
> shows some problem related to G729 codecs. following are Asterisk CLI log:
>
> Executing [...@skypeincoming:1]
> Answer("Skype/rexesbposolutions-084159e8", "") in new stack
> -- Executing [...@skypeincoming:2]
> Wait("Skype/rexesbposolutions-084159e8", "5") in new stack
> -- Executing [...@skypeincoming:3]
> GotoIfTime("Skype/rexesbposolutions-084159e8",
> "9:00-18:00|mon-fri|*|*?sky|s|1") in new stack
> -- Goto (sky,s,1)
> -- Executing [...@sky:1] Playback("Skype/rexesbposolutions-084159e8",
> "enter") in new stack
> -- Playing 'enter' (language 'en')
> [Dec 29 17:27:14] WARNING[4997]: channel.c:3106 set_format: Unable to find
> a codec translation path from 0x100 (g729) to 0x4 (ulaw)
> [Dec 29 17:27:14] WARNING[4997]: file.c:147 ast_stopstream: Unable to
> restore format back to 4
> -- Executing [...@sky:2] Queue("Skype/rexesbposolutions-084159e8",
> "markq|t|||900") in new stack
> -- Started music on hold, class 'default', on
> Skype/rexesbposolutions-084159e8
> [Dec 29 17:27:14] WARNING[4997]: channel.c:3106 set_format: Unable to find
> a codec translation path from 0x100 (g729) to 0x40 (slin)
> [Dec 29 17:27:14] WARNING[4997]: res_musiconhold.c:251 ast_moh_files_next:
> Unable to open file '/var/lib/asterisk/moh/manolo_camp-morning_coffee': No
> such file or directory
> -- Stopped music on hold on Skype/rexesbposolutions-084159e8
> [Dec 29 17:27:14] WARNING[4997]: channel.c:3106 set_format: Unable to find
> a codec translation path from 0x100 (g729) to 0x2 (gsm)
> [Dec 29 17:27:14] WARNING[4997]: res_musiconhold.c:204 moh_files_release:
> Unable to restore channel 'Skype/rexesbposolutions-084159e8' to format '2'
> -- Playing periodic announcement
> [Dec 29 17:27:34] WARNING[4997]: channel.c:3106 set_format: Unable to find
> a codec translation path from 0x100 (g729) to 0x2 (gsm)
> -- Playing 'queue' (language 'en')
> [Dec 29 17:27:43] WARNING[4997]: channel.c:3106 set_format: Unable to find
> a codec translation path from 0x100 (g729) to 0x2 (gsm)
> [Dec 29 17:27:43] WARNING[4997]: file.c:147 ast_stopstream: Unable to
> restore format back to 2
> == Spawn extension (sky, s, 2) exited non-zero on
> 'Skype/rexesbposolutions-084159e8'
> [Dec 29 17:27:43] NOTICE[4997]: core.cpp:2133 sfa_call_hangup: ending call
>
>
>
> following are output of some commands:-
>
> *CLI> core show translation
>
> Translation times between formats (in milliseconds) for one second of
> data
> Source Format (Rows) Destination Format (Columns)
>
> g723 gsm ulaw alaw g726aal2 adpcm slin lpc10 g729 speex ilbc g726
> g722
> g723- ---- -- -- --
> --
> gsm- -222 21 26 --
> 2-
> ulaw- 2-12 21 26 --
> 2-
> alaw- 21-2 21 26 --
> 2-
> g726aal2- 222- 21 26 --
> 2-
> adpcm- 2222 -1 26 --
> 2-
> slin- 1111 1- 15 --
> 1-
> lpc10- 2222 21 -6 --
> 2-
> g729- 6666 65 6- --
> 6-
> speex- ---- -- -- --
> --
> ilbc- ---- -- -- --
> --
> g726- 2222 21 26 --
> --
> g722- ---- -- -- --
> --
>
>
> *CLI> help g729
> g729 show hostid Show G.729 Host-ID
>g729 show licenses Show G.729 Licenses and Usage
> g729 show version Show G.729 Module Version
>
> *CLI> g729 show hostid
> Host-ID: 1a:cf:bb:80:da:e8:3b:dc:8c:e0:97:fe:a1:fb:65:c5:3c:e2:0b:be
>
> *CLI> g729 show licenses
> 0/0 encoders/decoders of 1 licensed channels are currently in use
>
> Licenses Found:
> File: ***-*.lic -- Key: ***-* --
Re: [asterisk-users] Skype for Asterisk
On 30 Dec 2009, at 19:43, vijay.go...@alliance-infotech.com wrote:
>
> Hi Sir,
>
> We have integrated Skype with Asterisk (skype user id:- rexesbposolutions).
> Each call which is coming to skype account is getting transfered to Asterisk
> Queue. It has following two cases:
>
> case 1: When we call from normal skype account to skype account
> (rexesbposolutions), everything is working fine.
>
> case 2: This skype account (rexesbposolutions) has been assigned with a
> online virtual number (00 44 20 ). If somebody dial this number from
> their landline/cellphone, call is transfered to Asterisk queue but it shows
> some problem related to G729 codecs. following are Asterisk CLI log:
>
> Executing [...@skypeincoming:1]
> Answer("Skype/rexesbposolutions-084159e8", "") in new stack
> -- Executing [...@skypeincoming:2]
> Wait("Skype/rexesbposolutions-084159e8", "5") in new stack
> -- Executing [...@skypeincoming:3]
> GotoIfTime("Skype/rexesbposolutions-084159e8",
> "9:00-18:00|mon-fri|*|*?sky|s|1") in new stack
> -- Goto (sky,s,1)
> -- Executing [...@sky:1] Playback("Skype/rexesbposolutions-084159e8",
> "enter") in new stack
> -- Playing 'enter' (language 'en')
> [Dec 29 17:27:14] WARNING[4997]: channel.c:3106 set_format: Unable to find a
> codec translation path from 0x100 (g729) to 0x4 (ulaw)
> [Dec 29 17:27:14] WARNING[4997]: file.c:147 ast_stopstream: Unable to restore
> format back to 4
> -- Executing [...@sky:2] Queue("Skype/rexesbposolutions-084159e8",
> "markq|t|||900") in new stack
> -- Started music on hold, class 'default', on
> Skype/rexesbposolutions-084159e8
> [Dec 29 17:27:14] WARNING[4997]: channel.c:3106 set_format: Unable to find a
> codec translation path from 0x100 (g729) to 0x40 (slin)
> [Dec 29 17:27:14] WARNING[4997]: res_musiconhold.c:251 ast_moh_files_next:
> Unable to open file '/var/lib/asterisk/moh/manolo_camp-morning_coffee': No
> such file or directory
> -- Stopped music on hold on Skype/rexesbposolutions-084159e8
> [Dec 29 17:27:14] WARNING[4997]: channel.c:3106 set_format: Unable to find a
> codec translation path from 0x100 (g729) to 0x2 (gsm)
> [Dec 29 17:27:14] WARNING[4997]: res_musiconhold.c:204 moh_files_release:
> Unable to restore channel 'Skype/rexesbposolutions-084159e8' to format '2'
> -- Playing periodic announcement
> [Dec 29 17:27:34] WARNING[4997]: channel.c:3106 set_format: Unable to find a
> codec translation path from 0x100 (g729) to 0x2 (gsm)
> -- Playing 'queue' (language 'en')
> [Dec 29 17:27:43] WARNING[4997]: channel.c:3106 set_format: Unable to find a
> codec translation path from 0x100 (g729) to 0x2 (gsm)
> [Dec 29 17:27:43] WARNING[4997]: file.c:147 ast_stopstream: Unable to restore
> format back to 2
> == Spawn extension (sky, s, 2) exited non-zero on
> 'Skype/rexesbposolutions-084159e8'
> [Dec 29 17:27:43] NOTICE[4997]: core.cpp:2133 sfa_call_hangup: ending call
>
>
>
> following are output of some commands:-
>
> *CLI> core show translation
>
> Translation times between formats (in milliseconds) for one second of data
> Source Format (Rows) Destination Format (Columns)
>
> g723 gsm ulaw alaw g726aal2 adpcm slin lpc10 g729 speex ilbc g726
> g722
> g723- ---- -- -- ---
> -
> gsm- -222 21 26 --2
> -
> ulaw- 2-12 21 26 --2
> -
> alaw- 21-2 21 26 --2
> -
> g726aal2- 222- 21 26 --2
> -
> adpcm- 2222 -1 26 --2
> -
> slin- 1111 1- 15 --1
> -
> lpc10- 2222 21 -6 --2
> -
> g729- 6666 65 6- --6
> -
> speex- ---- -- -- ---
> -
> ilbc- ---- -- -- ---
> -
> g726- 2222 21 26 ---
> -
> g722- ---- -- -- ---
> -
>
>
> *CLI> help g729
> g729 show hostid Show G.729 Host-ID
>g729 show licenses Show G.729 Licenses and Usage
> g729 show version Show G.729 Module Version
>
> *CLI> g729 show hostid
> Host-ID: 1a:cf:bb:80:da:e8:3b:dc:8c:e0:97:fe:a1:fb:65:c5:3c:e2:0b:be
>
> *CLI> g729 show licenses
> 0/0 encoders/decoders of 1 licensed channels are currently in use
>
> Licenses Found:
> File: ***-*.lic -- Key: ***-* -- Host-ID:
> 1a:cf:bb:80:da:e8:3b:dc:8c:e0:97:fe:a1:fb:65:c5:3c:e2:0b:be -- Channels: 1
> (Expires: 2029-11-30) (OK)
>
> *CLI> g729 show version
Re: [asterisk-users] Skype for Asterisk callfile question
On Wed, 2 Sep 2009, Matt Riddell wrote: > On 2/09/09 7:45 PM, Remco Barendse wrote: >> So i create a callfile that looks like this: >> --- >> Channel: SIP/228 >> MaxRetries: 0 >> Dial(Skype/asterisk...@somebodyonskype) >> Priority: 1 >> Callerid: Somebodyonskype > > You're combining technologies there :) Not hindered by any knowledge i was trying to get things working :) Thanks, it seems to make sense to Asterisk now :) The first time i configured SFA to allow incoming calls only, reloading the module does not allow outbound calls still (direction is not mentioed as a directive for which asterisk needs to be restarted but it doesn't really matter). Thanks again for your help. ___ -- 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
Re: [asterisk-users] Skype for Asterisk callfile question
On 2/09/09 7:45 PM, Remco Barendse wrote: > So i create a callfile that looks like this: > --- > Channel: SIP/228 > MaxRetries: 0 > Dial(Skype/asterisk...@somebodyonskype) > Priority: 1 > Callerid: Somebodyonskype You're combining technologies there :) You can do: Channel Context Extension Priority Or Channel Application Data Looks like you want Channel: SIP/228 Application: Dial Data: Skype/asterisk...@somebodyonskype -- Cheers, Matt Riddell Director ___ http://www.venturevoip.com/news.php (Daily Asterisk News) http://www.venturevoip.com/st.php (SmoothTorque Predictive Dialer) http://www.venturevoip.com/c3.php (ConduIT3 PABX Systems) ___ -- 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
Re: [asterisk-users] Skype for Asterisk???
On Wed, 19 Aug 2009, Terry Wilson wrote: > I haven't seen (or heard of) it happening. Please post a bug report > on http://betareports.digium.com/mantis/ with a backtrace from one of > the core dumps along with the relevant information about your setup, > dialplan, chan_skype.conf, etc. If there is a crash, I need to fix > it. :-) I tried to install /asterisk-1.4/x86-32/skypeforasterisk-1.4_1.0.1-x86_32.tar.gz instead of 1.4_1.0.0 but this version causes asterisk to segfault immediately after starting, with the previous version i had occasional segfaults, now asterisk never starts. I wanted to setup Skype for Asterisk on a production PBX where segfaults, downtime and frequent reboots are a nuisance ofcourse. I'll try to go down the debug path but the options to take down the production box for debugging are somewhat limited :) ___ -- 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
Re: [asterisk-users] Skype for Asterisk???
On Aug 20, 2009, at 3:41 AM, Remco Barendse wrote: > I never used Skype myself but i installed it to try and i noticed > that i > got added by lots of strange skype users (spam bots?), i guess some of > those were trying some funny stuff on my skype for asterisk account. I > want to use Skype for Asterisk only for incoming calls at this time. There is a direction=incoming option to limit Skype to inbound calls only (per-user). You can also set auth_policy to deny, block, or require a pssword so that your contact details are released to only those people that you want. > Is there a how to "filing bug reports for dummies" ? http://www.asterisk.org/developers/bug-guidelines > I am running Asterisk 1.4.26.1 with the latest FreePBX and Skype for > Asterisk is the only add-on. I did *not* enable the G.729 codec in the > config neither did i try to install the G.729 codec license that comes > with Skype for Asterisk (nowhere it said that the G.729 is required > for > correct operation of Skype and i don't care about the bandwidth at > this > time). The README file that ships with Skype for Asterisk states: Beta licenses will not be recognized by the G.729 module and therefore Beta users will need to obtain a separate license for the G.729 codec to guarantee that they can process the audio for SkypeIn and SkypeOut calls. > The core dumps i can just delete, they are useless? After filing a bug report with a backtrace from them posted. See doc/ backtrace.txt in an Asterisk distribution or http://svn.digium.com/svn/asterisk/branches/1.4/doc/backtrace.txt for information on how to create one. ___ -- 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
Re: [asterisk-users] Skype for Asterisk???
On Wed, 19 Aug 2009, Terry Wilson wrote: >>> Have you posted a bug describing the issues you are having at >>> http://betareports.digium.com/mantis/ >>> yet? I would love to have the opportunity to actually fix any bugs >>> that people find. :-) >> >> I installed the 1.0 release of Skype for Asterisk and last night on my >> production box running Asterisk 1.26.1 i got segfaults and 32 core >> dumps, >> all happened in a time frame between 01:04 - 01:08 at night (so 4 >> minutes). >> >> Anyone else seeing this? > > I haven't seen (or heard of) it happening. Please post a bug report > on http://betareports.digium.com/mantis/ with a backtrace from one of > the core dumps along with the relevant information about your setup, > dialplan, chan_skype.conf, etc. If there is a crash, I need to fix > it. :-) I never used Skype myself but i installed it to try and i noticed that i got added by lots of strange skype users (spam bots?), i guess some of those were trying some funny stuff on my skype for asterisk account. I want to use Skype for Asterisk only for incoming calls at this time. Is there a how to "filing bug reports for dummies" ? I am running Asterisk 1.4.26.1 with the latest FreePBX and Skype for Asterisk is the only add-on. I did *not* enable the G.729 codec in the config neither did i try to install the G.729 codec license that comes with Skype for Asterisk (nowhere it said that the G.729 is required for correct operation of Skype and i don't care about the bandwidth at this time). The core dumps i can just delete, they are useless? ___ -- 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
Re: [asterisk-users] Skype for Asterisk???
>> Have you posted a bug describing the issues you are having at >> http://betareports.digium.com/mantis/ >> yet? I would love to have the opportunity to actually fix any bugs >> that people find. :-) > > I installed the 1.0 release of Skype for Asterisk and last night on my > production box running Asterisk 1.26.1 i got segfaults and 32 core > dumps, > all happened in a time frame between 01:04 - 01:08 at night (so 4 > minutes). > > Anyone else seeing this? I haven't seen (or heard of) it happening. Please post a bug report on http://betareports.digium.com/mantis/ with a backtrace from one of the core dumps along with the relevant information about your setup, dialplan, chan_skype.conf, etc. If there is a crash, I need to fix it. :-) ___ -- 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
Re: [asterisk-users] Skype for Asterisk???
Michael Graves wrote: > I wonder if that was not a codec specific issue, but rather the matter > of their license to the p2p technology provided by JoltID? Since Skype > has recently dveloped their own codec (SILK) they could easily drop > support for any codec that they previously licensed from outside. I > think that the failure to ge a new license on a codec would not be a > major issue for them. > > Failure to renew the license on the p2p transport technology is a much > more significant problem. > > Michael > That's probably what it was, It does appear to be trying to remove Jolt support. http://www.theregister.co.uk/2009/07/31/skype_joltid/ ___ -- 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
Re: [asterisk-users] Skype for Asterisk???
I wonder if that was not a codec specific issue, but rather the matter
of their license to the p2p technology provided by JoltID? Since Skype
has recently dveloped their own codec (SILK) they could easily drop
support for any codec that they previously licensed from outside. I
think that the failure to ge a new license on a codec would not be a
major issue for them.
Failure to renew the license on the p2p transport technology is a much
more significant problem.
Michael
--Original Message Text---
From: David fire
Date: Wed, 19 Aug 2009 09:04:59 -0300
a few days ago slashdot (sorry i havent the link now) wrote about skype
has a very huge problem whit a licence in a core codec, and if they
dont get an aregment whit the codec owner they will close the doors...
David
2009/8/19 Thomas Kenyon
Julian Lyndon-Smith wrote:
> Nope - but you are also running on an unsupported version of asterisk,
> so I am not surprised. From the readme:
>
> ===[ Installation Overview
> ]===
>
> It is required that the proper version of Asterisk is installed prior to
> installing Skype For Asterisk. Skype For Asterisk is currently supported on:
>
>Asterisk 1.4 versions >= 1.4.25
>Asterisk 1.6.0 versions >= 1.6.0.6
>Asterisk 1.6.1 versions >= 1.6.1.5
>
Ah didn't spot that, if you are running 1.6.1, you need a version that
isn't available yet.
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Re: [asterisk-users] Skype for Asterisk???
a few days ago slashdot (sorry i havent the link now) wrote about skype has
a very huge problem whit a licence in a core codec, and if they dont get an
aregment whit the codec owner they will close the doors...
David
2009/8/19 Thomas Kenyon
> Julian Lyndon-Smith wrote:
> > Nope - but you are also running on an unsupported version of asterisk,
> > so I am not surprised. From the readme:
> >
> > ===[ Installation Overview
> ]===
> >
> > It is required that the proper version of Asterisk is installed prior to
> > installing Skype For Asterisk. Skype For Asterisk is currently supported
> on:
> >
> >Asterisk 1.4 versions >= 1.4.25
> >Asterisk 1.6.0 versions >= 1.6.0.6
> >Asterisk 1.6.1 versions >= 1.6.1.5
> >
> Ah didn't spot that, if you are running 1.6.1, you need a version that
> isn't available yet.
>
> ___
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Re: [asterisk-users] Skype for Asterisk???
Julian Lyndon-Smith wrote: > Nope - but you are also running on an unsupported version of asterisk, > so I am not surprised. From the readme: > > ===[ Installation Overview > ]=== > > It is required that the proper version of Asterisk is installed prior to > installing Skype For Asterisk. Skype For Asterisk is currently supported on: > >Asterisk 1.4 versions >= 1.4.25 >Asterisk 1.6.0 versions >= 1.6.0.6 >Asterisk 1.6.1 versions >= 1.6.1.5 > Ah didn't spot that, if you are running 1.6.1, you need a version that isn't available yet. ___ -- 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
Re: [asterisk-users] Skype for Asterisk???
Oops sorry, the Asterisk version should read 1.4.26.1 On Wed, 19 Aug 2009, Julian Lyndon-Smith wrote: Nope - but you are also running on an unsupported version of asterisk, so I am not surprised. From the readme: ===[ Installation Overview ]=== It is required that the proper version of Asterisk is installed prior to installing Skype For Asterisk. Skype For Asterisk is currently supported on: Asterisk 1.4 versions >= 1.4.25 Asterisk 1.6.0 versions >= 1.6.0.6 Asterisk 1.6.1 versions >= 1.6.1.5 Previous versions of Asterisk WILL NOT work properly with Skype For Asterisk. It is also important to make sure that the major version of Skype For Asterisk downloaded matches the version of Asterisk installed on the system. Trying to compile Skype For Asterisk 1.4 versions on Asterisk 1.6.0 while fail, etc. There is no version of Skype For Asterisk for Asterisk trunk. Julian 2009/8/19 Remco Barendse : On Tue, 18 Aug 2009, Terry Wilson wrote: That does sound a bit pricey, although it it's as stable as the latest beta, I wont be buying it at all. Have you posted a bug describing the issues you are having at http://betareports.digium.com/mantis/ yet? I would love to have the opportunity to actually fix any bugs that people find. :-) I installed the 1.0 release of Skype for Asterisk and last night on my production box running Asterisk 1.26.1 i got segfaults and 32 core dumps, all happened in a time frame between 01:04 - 01:08 at night (so 4 minutes). Anyone else seeing this? ___ -- 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 ___ -- 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
Re: [asterisk-users] Skype for Asterisk???
Nope - but you are also running on an unsupported version of asterisk, so I am not surprised. From the readme: ===[ Installation Overview ]=== It is required that the proper version of Asterisk is installed prior to installing Skype For Asterisk. Skype For Asterisk is currently supported on: Asterisk 1.4 versions >= 1.4.25 Asterisk 1.6.0 versions >= 1.6.0.6 Asterisk 1.6.1 versions >= 1.6.1.5 Previous versions of Asterisk WILL NOT work properly with Skype For Asterisk. It is also important to make sure that the major version of Skype For Asterisk downloaded matches the version of Asterisk installed on the system. Trying to compile Skype For Asterisk 1.4 versions on Asterisk 1.6.0 while fail, etc. There is no version of Skype For Asterisk for Asterisk trunk. Julian 2009/8/19 Remco Barendse : > On Tue, 18 Aug 2009, Terry Wilson wrote: > >>> That does sound a bit pricey, although it it's as stable as the latest >>> beta, I wont be buying it at all. >> >> Have you posted a bug describing the issues you are having at >> http://betareports.digium.com/mantis/ >> yet? I would love to have the opportunity to actually fix any bugs >> that people find. :-) > > I installed the 1.0 release of Skype for Asterisk and last night on my > production box running Asterisk 1.26.1 i got segfaults and 32 core dumps, > all happened in a time frame between 01:04 - 01:08 at night (so 4 > minutes). > > Anyone else seeing this? > > ___ > -- 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
Re: [asterisk-users] Skype for Asterisk???
On Tue, 18 Aug 2009, Terry Wilson wrote: >> That does sound a bit pricey, although it it's as stable as the latest >> beta, I wont be buying it at all. > > Have you posted a bug describing the issues you are having at > http://betareports.digium.com/mantis/ > yet? I would love to have the opportunity to actually fix any bugs > that people find. :-) I installed the 1.0 release of Skype for Asterisk and last night on my production box running Asterisk 1.26.1 i got segfaults and 32 core dumps, all happened in a time frame between 01:04 - 01:08 at night (so 4 minutes). Anyone else seeing this? ___ -- 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
Re: [asterisk-users] Skype for Asterisk -- Codec support
Karl Fife wrote: > Any idea what timeframe? Not that I can disclose, no. Sorry. > Can I assume that SFA licenses now will be valid for future releases? Yes. It is quite unlikely (although not impossible, of course, like any piece of software) that existing SFA licenses would not be valid for future releases. -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kpflem...@digium.com Check us out at www.digium.com & www.asterisk.org ___ -- 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
Re: [asterisk-users] Skype for Asterisk???
> I just want to also remind people that Skype for SIP is also to be > released shortly. When I last talked to Skype they said it would be > out in late July. So I assume if you wait another few more weeks > the entire issue will be moot. No $60/channel fee, just the free > SIP platform for people using the business version of Skype. No Skype to Skype calls with Skype for SIP, only trunking. I would also be very surprised if there wasn't a fee on top of the per-minute fees. But, we'll see. ___ -- 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
Re: [asterisk-users] Skype for Asterisk -- Codec support
- Original Message - From: "Kevin P. Fleming" To: "Asterisk Users Mailing List - Non-Commercial Discussion" Sent: Tuesday, August 18, 2009 5:03 PM Subject: Re: [asterisk-users] Skype for Asterisk -- Codec support > Karl Fife wrote: > >> To some degree, I (and I'm sure other wideband disciples), feel somewhat >> like the only guy on the block with a fax machine. Per Metcalfe's law it >> seemed that Skype (as a easily accessible namespace supporting wideband) >> could have been a shot in the arm for >8khz telephony. >> >> I know that wideband SFA was originally on the roadmap. Has it been >> silently deleted, or is it going to be released in some next-generation >> SFA? > > It will appear in a release in the future, but did not make the cut for > the first release. Part of the reason for that is that Skype themselves > have just begun switching to a new wideband codec, and it did not make > sense to incorporate support for the old one in a brand new product... > but the new one is not ready for use yet either. We'll get there :-) Any idea what timeframe? Can I assume that SFA licenses now will be valid for future releases? Thanks -Karl > -- > Kevin P. Fleming > Digium, Inc. | Director of Software Technologies > 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA > skype: kpfleming | jabber: kpflem...@digium.com > Check us out at www.digium.com & www.asterisk.org > > ___ > -- 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
Re: [asterisk-users] Skype for Asterisk???
> That does sound a bit pricey, although it it's as stable as the latest > beta, I wont be buying it at all. Have you posted a bug describing the issues you are having at http://betareports.digium.com/mantis/ yet? I would love to have the opportunity to actually fix any bugs that people find. :-) ___ -- 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
Re: [asterisk-users] Skype for Asterisk -- Codec support
Karl Fife wrote: > To some degree, I (and I'm sure other wideband disciples), feel somewhat > like the only guy on the block with a fax machine. Per Metcalfe's law it > seemed that Skype (as a easily accessible namespace supporting wideband) > could have been a shot in the arm for >8khz telephony. > > I know that wideband SFA was originally on the roadmap. Has it been > silently deleted, or is it going to be released in some next-generation SFA? It will appear in a release in the future, but did not make the cut for the first release. Part of the reason for that is that Skype themselves have just begun switching to a new wideband codec, and it did not make sense to incorporate support for the old one in a brand new product... but the new one is not ready for use yet either. We'll get there :-) -- Kevin P. Fleming Digium, Inc. | Director of Software Technologies 445 Jan Davis Drive NW - Huntsville, AL 35806 - USA skype: kpfleming | jabber: kpflem...@digium.com Check us out at www.digium.com & www.asterisk.org ___ -- 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
Re: [asterisk-users] Skype for Asterisk???
I just want to also remind people that Skype for SIP is also to be released shortly. When I last talked to Skype they said it would be out in late July. So I assume if you wait another few more weeks the entire issue will be moot. No $60/channel fee, just the free SIP platform for people using the business version of Skype. Nicholas Blasgen Partner / Network Operations Refractive Dialer LLC (724) 252-7436 On Tue, Aug 18, 2009 at 10:35 AM, Pascal Bruno wrote: > Lol but he has a good point and makes a lot of sense. Never thought about > that strategy... > > > > On Tue, Aug 18, 2009 at 12:16 PM, Thomas Kenyon > wrote: > >> Michael Graves wrote: >> > 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 >> > >> You make it sound like you're saying it's expensive because it doesn't >> work :-) >> >> ___ >> -- 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 > ___ -- 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
Re: [asterisk-users] Skype for Asterisk???
Lol but he has a good point and makes a lot of sense. Never thought about that strategy... On Tue, Aug 18, 2009 at 12:16 PM, Thomas Kenyon wrote: > Michael Graves wrote: > > 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 > > > You make it sound like you're saying it's expensive because it doesn't > work :-) > > ___ > -- 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
Re: [asterisk-users] Skype for Asterisk???
Michael Graves wrote: > 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 > You make it sound like you're saying it's expensive because it doesn't work :-) ___ -- 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
Re: [asterisk-users] Skype for Asterisk???
Casey Boone wrote: > I would have happily bought 20 channels at $10/channel, but at most will > be buying only a single channel now :\ > That does sound a bit pricey, although it it's as stable as the latest beta, I wont be buying it at all. ___ -- 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
Re: [asterisk-users] Skype for Asterisk???
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 mgravesmstvp.com http://www.mgraves.org o713-861-4005 c713-201-1262 sip:mgra...@mstvp.onsip.com skype mjgraves Twitter mjgraves ___ -- 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
Re: [asterisk-users] Skype for Asterisk???
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
Re: [asterisk-users] Skype for Asterisk???
Geoff Lane wrote: > On Tuesday, August 18, 2009, Gordon Henderson wrote: > >> I was under the impression that Three (who I guess you're using) >> placed a regular call over their network then "Skyped" it at their >> HQ - rather than have the Skype client actually reside in the >> handset.. (And I'm suspecting their 3G limitation is that they want >> to use their own 3G network rather than pay Orange for the call >> over their 2G network) > > I am using Three. At first I got two Three S1 Skypephones and was > allegedly one of the first business customers to take up those phones. > One handset was replaced under warranty for a basic Nokia (can't > remember the model number) that offered Bluetooth and could run the > Skype application. > > In both cases, AIUI you run the Skype client on your handset, which > uses 3G/HSDPA data bandwidth to connect to Skype via a NAT router in > Three's network. > > FWIW and IMO the S1 is rubbish. However some tell me that the S1 is > part of the problem I have using Skype and that the S2 is considerably > better. > It's kinda a mixture of both, the client on the handset sets up the call that is a regular voice call through their gateway. A bit (although not much) like the way international roaming SIMS make calls. ___ -- 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
Re: [asterisk-users] Skype for Asterisk???
On Tue, 18 Aug 2009, Geraint Lee wrote: > Good luck with the N95... my experiences of the N95 and SIP haven't been > great... the phone likes to restart... regularly. Nokia may well have fixed > these glitches by now though. Getting it configured was a bit of a mission > too... and as expected the battery life shoots down when it's enabled... > so... again... good luck, and hope that nokia have fixed all those issues :p > saying that though, to be fair, it did work most of the time, just had to > put up with annoying restarts, sometimes at very inconvenient times... like > when the phone rang. I've used the SIP + Wi-Fi client on my Nokia E90 with a good degree of success - it would only be for fun use I guess with my wifes phone (We have DECT all over the house, so no real need). > As for skype, I can see how it could be useful, I've worked with a few > developers who's choice of communication method has been skype (I hate it > myself though!), so being able to let them call a skype number and have it > direct to a real phone would be quite useful. I've had a few customer ask for it - wanting to use their desk phone to call Skype numbers... "it's all voip, isn't it?" ... So who knows. Bet they don't want to pay for it!!! Gordon ___ -- 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
Re: [asterisk-users] Skype for Asterisk???
Good luck with the N95... my experiences of the N95 and SIP haven't been great... the phone likes to restart... regularly. Nokia may well have fixed these glitches by now though. Getting it configured was a bit of a mission too... and as expected the battery life shoots down when it's enabled... so... again... good luck, and hope that nokia have fixed all those issues :p saying that though, to be fair, it did work most of the time, just had to put up with annoying restarts, sometimes at very inconvenient times... like when the phone rang. As for skype, I can see how it could be useful, I've worked with a few developers who's choice of communication method has been skype (I hate it myself though!), so being able to let them call a skype number and have it direct to a real phone would be quite useful. 2009/8/18 Gordon Henderson > > On Tue, 18 Aug 2009, Geoff Lane wrote: > > > On Tuesday, August 18, 2009, Remco Barendse wrote: > > > >> But then again, who needs Skype for business purposes anyways, i > >> don't think there is a huge market for it. > > > > Me ... at least in theory! Our cellphones have built-in Skype, so a > > Skype gateway should give me call forwarding and diversion to our > > cellphones free of charge. > > > > So far Skype as implemented on our mobiles has proved too unreliable > > period for business use. It seems only available when we can get a > > 3G/HSDPA signal and even then the system regularly logs us out of > > Skype and sometimes doesn't log us back in. However, if and when my > > cellular provider get Skype sorted out on their system ... > > I was under the impression that Three (who I guess you're using) placed a > regular call over their network then "Skyped" it at their HQ - rather than > have the Skype client actually reside in the handset.. (And I'm suspecting > their 3G limitation is that they want to use their own 3G network rather > than pay Orange for the call over their 2G network) > > But then again, wifey's just gotten a new Three mobile (N95 - end of line, > but a cheap deal) It has built in SIP via Wi-Fi and skype, so I might have > a play with it when I can prise it out of her hands... > > It does worry me that I see many so-called business people advertising > Skype numbers on their business cards, etc. To me it rings of cheapness. I > can almost always tell when someone calls me using their Skype out service > - the quality is dreadful, and I end up calling them on their > regular landline. Cheapskates who can't/won't pay for a decent Internet > service. > > I'm going to ask my customers if they want to be able to call Skype > numbers, but I'd probably have to charge for it to justify the cost of the > license(s) required. > > Gordon > > ___ > -- 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
Re: [asterisk-users] Skype for Asterisk???
On Tuesday, August 18, 2009, Gordon Henderson wrote: > I was under the impression that Three (who I guess you're using) > placed a regular call over their network then "Skyped" it at their > HQ - rather than have the Skype client actually reside in the > handset.. (And I'm suspecting their 3G limitation is that they want > to use their own 3G network rather than pay Orange for the call > over their 2G network) I am using Three. At first I got two Three S1 Skypephones and was allegedly one of the first business customers to take up those phones. One handset was replaced under warranty for a basic Nokia (can't remember the model number) that offered Bluetooth and could run the Skype application. In both cases, AIUI you run the Skype client on your handset, which uses 3G/HSDPA data bandwidth to connect to Skype via a NAT router in Three's network. FWIW and IMO the S1 is rubbish. However some tell me that the S1 is part of the problem I have using Skype and that the S2 is considerably better. -- Geoff ___ -- 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
Re: [asterisk-users] Skype for Asterisk???
On Tue, 18 Aug 2009, Geoff Lane wrote: > On Tuesday, August 18, 2009, Remco Barendse wrote: > >> But then again, who needs Skype for business purposes anyways, i >> don't think there is a huge market for it. > > Me ... at least in theory! Our cellphones have built-in Skype, so a > Skype gateway should give me call forwarding and diversion to our > cellphones free of charge. > > So far Skype as implemented on our mobiles has proved too unreliable > period for business use. It seems only available when we can get a > 3G/HSDPA signal and even then the system regularly logs us out of > Skype and sometimes doesn't log us back in. However, if and when my > cellular provider get Skype sorted out on their system ... I was under the impression that Three (who I guess you're using) placed a regular call over their network then "Skyped" it at their HQ - rather than have the Skype client actually reside in the handset.. (And I'm suspecting their 3G limitation is that they want to use their own 3G network rather than pay Orange for the call over their 2G network) But then again, wifey's just gotten a new Three mobile (N95 - end of line, but a cheap deal) It has built in SIP via Wi-Fi and skype, so I might have a play with it when I can prise it out of her hands... It does worry me that I see many so-called business people advertising Skype numbers on their business cards, etc. To me it rings of cheapness. I can almost always tell when someone calls me using their Skype out service - the quality is dreadful, and I end up calling them on their regular landline. Cheapskates who can't/won't pay for a decent Internet service. I'm going to ask my customers if they want to be able to call Skype numbers, but I'd probably have to charge for it to justify the cost of the license(s) required. Gordon ___ -- 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
Re: [asterisk-users] Skype for Asterisk???
On Tuesday, August 18, 2009, Remco Barendse wrote: > But then again, who needs Skype for business purposes anyways, i > don't think there is a huge market for it. Me ... at least in theory! Our cellphones have built-in Skype, so a Skype gateway should give me call forwarding and diversion to our cellphones free of charge. So far Skype as implemented on our mobiles has proved too unreliable period for business use. It seems only available when we can get a 3G/HSDPA signal and even then the system regularly logs us out of Skype and sometimes doesn't log us back in. However, if and when my cellular provider get Skype sorted out on their system ... -- Geoff ___ -- 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
Re: [asterisk-users] Skype for Asterisk???
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=1SFA0001 Yes, 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. ___ -- 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
Re: [asterisk-users] Skype for Asterisk???
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 ___ -- 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
Re: [asterisk-users] Skype for Asterisk: Public Beta available
On Thu, Jul 30, 2009 at 8:50 PM, John Todd wrote: > > I know many of you have been waiting for this for a while, so I'll > keep this short: The Skype for Asterisk Public Beta is now available > on the Digium store. > > We are pleased to announce the open beta of Skype For Asterisk is > ready to begin and we look forward to you participation. To obtain > your copy of the software, please visit Digium’s web store and > purchase (for zero dollars) the Skype For Asterisk product. The web > store does require a Digium.com account, which can be set up during > the purchase process if you don’t already have one. > Once the web store process is complete, you will be e-mailed your > license key and directions on where to download Skype For Asterisk > beta software. > > It crashes my box after the incoming call is answered :( http://betareports.digium.com/mantis/view.php?id=21 -- Shimi ___ -- 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
Re: [asterisk-users] Skype for Asterisk: Public Beta available
Pascal Bruno wrote: > Well I think thats what the problem was, I dont have it named as eth0. > So if your NIC is not labeled eth0 you cannot use skypeforasterisk??? > Why cant it just scan you nic handles? Can someone point me to where I > can change the NIC name in the source file or something??? > I don't know about centos, but in debian the file /etc/udev/rules.d/70-persistent-net.rules decides which interfaces are named what. ___ -- 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
Re: [asterisk-users] Skype for Asterisk: Public Beta available
Hi, This is the error message I get. Any idea where I may find some further debug logs? > [Aug 3 08:01:23] ERROR[23831] chan_skype.c: Unable to start Skype For > Asterisk library. Thanks, Emrah Tim Panton wrote: > I don't know then. My understanding is that the message is caused by > the wrong skypeforasterisk process running. > > - did you (ever) run it as a different user ? > > If it is a test box, you could try a full reboot. > > Tim. > > On 2 Aug 2009, at 19:35, Emrah wrote: > >> Hi Tim, >> >> I don't have any skypeforasterisk process running. I tried to killall >> -9 asterisk but it did not solve my issue. >> Any other suggestions? >> Thanks for your help, >> Emrah >> Tim Panton wrote: >>> I had that too, I cured it by kill -9 'ing the skypeforasterisk >>> process that was left over from >>> the previous version of the beta. >>> >>> Hope that helps. >>> >>> Tim. >>> >>> On 2 Aug 2009, at 11:20, Emrah wrote: >>> I reported an issue on Mantis (#14). Waiting for an update. http://betareports.digium.com/mantis/view.php?id=14 Emrah Emrah wrote: > Hi Thomas, > > I am experiencing the same problem, with the same error messages. > Running Asterisk 1.6.1.0 on a Debian 2.6.24-etchnhalf.1-686 > > Regards, > Emrah > Thomas Kenyon wrote: > >> Thomas Kenyon wrote: >> >> >>> [2009-08-02 08:46:05] ERROR[23719]: astobj2.c:116 INTERNAL_OBJ: bad >>> magic number 0x25765ca0 for 0x1390e20 >>> [2009-08-02 08:46:05] ERROR[23719]: astobj2.c:116 INTERNAL_OBJ: bad >>> magic number 0x25765ca0 for 0x1390e20 >>> >>> >>> >> chef*CLI> skype show users >> Skype Users >> [2009-08-02 09:39:35] ERROR[25506]: astobj2.c:116 INTERNAL_OBJ: bad >> magic number 0x70796b73 for 0x7f4fe0044340 >> [2009-08-02 09:39:35] ERROR[25506]: astobj2.c:116 INTERNAL_OBJ: bad >> magic number 0x70796b73 for 0x7f4fe0044340 >> >> Sorry, these are the error messages. >> >> ___ >> -- 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 >>> >>> Tim Panton - Web/VoIP consultant and implementor >>> www.westhawk.co.uk >>> >>> >>> >>> >>> >>> >>> ___ >>> -- 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 > > Tim Panton - Web/VoIP consultant and implementor > www.westhawk.co.uk > > > > > > ___ > -- 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
Re: [asterisk-users] Skype for Asterisk: Public Beta available
I don't know then. My understanding is that the message is caused by the wrong skypeforasterisk process running. - did you (ever) run it as a different user ? If it is a test box, you could try a full reboot. Tim. On 2 Aug 2009, at 19:35, Emrah wrote: Hi Tim, I don't have any skypeforasterisk process running. I tried to killall -9 asterisk but it did not solve my issue. Any other suggestions? Thanks for your help, Emrah Tim Panton wrote: I had that too, I cured it by kill -9 'ing the skypeforasterisk process that was left over from the previous version of the beta. Hope that helps. Tim. On 2 Aug 2009, at 11:20, Emrah wrote: I reported an issue on Mantis (#14). Waiting for an update. http://betareports.digium.com/mantis/view.php?id=14 Emrah Emrah wrote: Hi Thomas, I am experiencing the same problem, with the same error messages. Running Asterisk 1.6.1.0 on a Debian 2.6.24-etchnhalf.1-686 Regards, Emrah Thomas Kenyon wrote: Thomas Kenyon wrote: [2009-08-02 08:46:05] ERROR[23719]: astobj2.c:116 INTERNAL_OBJ: bad magic number 0x25765ca0 for 0x1390e20 [2009-08-02 08:46:05] ERROR[23719]: astobj2.c:116 INTERNAL_OBJ: bad magic number 0x25765ca0 for 0x1390e20 chef*CLI> skype show users Skype Users [2009-08-02 09:39:35] ERROR[25506]: astobj2.c:116 INTERNAL_OBJ: bad magic number 0x70796b73 for 0x7f4fe0044340 [2009-08-02 09:39:35] ERROR[25506]: astobj2.c:116 INTERNAL_OBJ: bad magic number 0x70796b73 for 0x7f4fe0044340 Sorry, these are the error messages. ___ -- 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 Tim Panton - Web/VoIP consultant and implementor www.westhawk.co.uk ___ -- 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 Tim Panton - Web/VoIP consultant and implementor www.westhawk.co.uk smime.p7s Description: S/MIME cryptographic signature ___ -- 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
Re: [asterisk-users] Skype for Asterisk: Public Beta available
Hi Tim, I don't have any skypeforasterisk process running. I tried to killall -9 asterisk but it did not solve my issue. Any other suggestions? Thanks for your help, Emrah Tim Panton wrote: > I had that too, I cured it by kill -9 'ing the skypeforasterisk > process that was left over from > the previous version of the beta. > > Hope that helps. > > Tim. > > On 2 Aug 2009, at 11:20, Emrah wrote: > >> I reported an issue on Mantis (#14). >> Waiting for an update. >> http://betareports.digium.com/mantis/view.php?id=14 >> >> Emrah >> Emrah wrote: >>> Hi Thomas, >>> >>> I am experiencing the same problem, with the same error messages. >>> Running Asterisk 1.6.1.0 on a Debian 2.6.24-etchnhalf.1-686 >>> >>> Regards, >>> Emrah >>> Thomas Kenyon wrote: >>> Thomas Kenyon wrote: > [2009-08-02 08:46:05] ERROR[23719]: astobj2.c:116 INTERNAL_OBJ: bad > magic number 0x25765ca0 for 0x1390e20 > [2009-08-02 08:46:05] ERROR[23719]: astobj2.c:116 INTERNAL_OBJ: bad > magic number 0x25765ca0 for 0x1390e20 > > > chef*CLI> skype show users Skype Users [2009-08-02 09:39:35] ERROR[25506]: astobj2.c:116 INTERNAL_OBJ: bad magic number 0x70796b73 for 0x7f4fe0044340 [2009-08-02 09:39:35] ERROR[25506]: astobj2.c:116 INTERNAL_OBJ: bad magic number 0x70796b73 for 0x7f4fe0044340 Sorry, these are the error messages. ___ -- 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 > > Tim Panton - Web/VoIP consultant and implementor > www.westhawk.co.uk > > > > > > ___ > -- 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
Re: [asterisk-users] Skype for Asterisk: Public Beta available
I had that too, I cured it by kill -9 'ing the skypeforasterisk process that was left over from the previous version of the beta. Hope that helps. Tim. On 2 Aug 2009, at 11:20, Emrah wrote: I reported an issue on Mantis (#14). Waiting for an update. http://betareports.digium.com/mantis/view.php?id=14 Emrah Emrah wrote: Hi Thomas, I am experiencing the same problem, with the same error messages. Running Asterisk 1.6.1.0 on a Debian 2.6.24-etchnhalf.1-686 Regards, Emrah Thomas Kenyon wrote: Thomas Kenyon wrote: [2009-08-02 08:46:05] ERROR[23719]: astobj2.c:116 INTERNAL_OBJ: bad magic number 0x25765ca0 for 0x1390e20 [2009-08-02 08:46:05] ERROR[23719]: astobj2.c:116 INTERNAL_OBJ: bad magic number 0x25765ca0 for 0x1390e20 chef*CLI> skype show users Skype Users [2009-08-02 09:39:35] ERROR[25506]: astobj2.c:116 INTERNAL_OBJ: bad magic number 0x70796b73 for 0x7f4fe0044340 [2009-08-02 09:39:35] ERROR[25506]: astobj2.c:116 INTERNAL_OBJ: bad magic number 0x70796b73 for 0x7f4fe0044340 Sorry, these are the error messages. ___ -- 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 Tim Panton - Web/VoIP consultant and implementor www.westhawk.co.uk smime.p7s Description: S/MIME cryptographic signature ___ -- 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
Re: [asterisk-users] Skype for Asterisk: Public Beta available
Well I think thats what the problem was, I dont have it named as eth0. So if your NIC is not labeled eth0 you cannot use skypeforasterisk??? Why cant it just scan you nic handles? Can someone point me to where I can change the NIC name in the source file or something??? On Sun, Aug 2, 2009 at 1:05 PM, Steve Totaro wrote: > On Sun, Aug 2, 2009 at 12:13 PM, randulo wrote: > >> On Sun, Aug 2, 2009 at 8:24 AM, Pascal Bruno wrote: >> > So what do you think I can do to register my license? I am running >> > Asterisk 1.6.10 on CentOS 5. >> >> >>> Could not generate Host-ID. >> >>> Make sure that you have eth0 enabled. >> >> The MAC is used in the scheme to register and it looks like it can't >> be read for some reason. There must be a direct channel to Digium for >> the support of this kind, though. Have you tried contacting them? >> >> [waits for John Todd to chime in here...] >> > > > Is eth0 enabled? Is it named eth0? > > What does ifconfig eth0 tell you? > > I have seen many Dell servers where the two NICs are labeled eth1 and eth2 > or whatever, but in Linux, they are backwards. Eth2 show up as eth0 and > eth1 shows up as eth1 in Linux. > > Wasted a good half hour to forty five minutes trying to figure out why I > couldn't get the network up. > > -- > Thanks, > Steve Totaro > +18887771888 (Toll Free) > +12409381212 (Cell) > +12024369784 (Skype) > > ___ > -- 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
Re: [asterisk-users] Skype for Asterisk: Public Beta available
On Sun, Aug 2, 2009 at 12:13 PM, randulo wrote: > On Sun, Aug 2, 2009 at 8:24 AM, Pascal Bruno wrote: > > So what do you think I can do to register my license? I am running > > Asterisk 1.6.10 on CentOS 5. > > >>> Could not generate Host-ID. > >>> Make sure that you have eth0 enabled. > > The MAC is used in the scheme to register and it looks like it can't > be read for some reason. There must be a direct channel to Digium for > the support of this kind, though. Have you tried contacting them? > > [waits for John Todd to chime in here...] > Is eth0 enabled? Is it named eth0? What does ifconfig eth0 tell you? I have seen many Dell servers where the two NICs are labeled eth1 and eth2 or whatever, but in Linux, they are backwards. Eth2 show up as eth0 and eth1 shows up as eth1 in Linux. Wasted a good half hour to forty five minutes trying to figure out why I couldn't get the network up. -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) ___ -- 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
Re: [asterisk-users] Skype for Asterisk: Public Beta available
On Sun, Aug 2, 2009 at 8:24 AM, Pascal Bruno wrote: > So what do you think I can do to register my license? I am running > Asterisk 1.6.10 on CentOS 5. >>> Could not generate Host-ID. >>> Make sure that you have eth0 enabled. The MAC is used in the scheme to register and it looks like it can't be read for some reason. There must be a direct channel to Digium for the support of this kind, though. Have you tried contacting them? [waits for John Todd to chime in here...] ___ -- 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
Re: [asterisk-users] Skype for Asterisk: Public Beta available
So what do you think I can do to register my license? I am running
Asterisk 1.6.10 on CentOS 5.
Sent from my iPod
On Aug 2, 2009, at 3:49 AM, Thomas Kenyon
wrote:
> Pascal Bruno wrote:
>> Unfortunately for me, I cannot register my license. Kept saying:
>>
>> Could not generate Host-ID.
>> Make sure that you have eth0 enabled.
>>
>> Any help would be appreciated
>>
> It uses the same licensing scheme as the G.729 licenses (so as soon as
> you need to upgrade the machine, or set up LACP or VPN or any other
> type
> of virtual interface or in the case of G.729 you change the codec to a
> newer version {since you've upgraded to a new version of asterisk that
> doesn't support older ones} that doesn't support the old name for the
> codec, you need to re-register).
>
> Or as in your case, it doesn't like the names of the network
> interfaces.
>
> It's all a total PITA.
>
> Fwiw, the Skype channel driver stopped working on my machine a while
> ago. I never did track down the cause.
>
> When res_skypeforasterisk starts, 39 res_skypeforasterisk processes
> start and 1 skypewatcher service starts.
>
> If I start it manually after asterisk has started, usually asterisk
> segfaults, (not always).
>
> Although Sometimes it starts up properly but can't log anyone in,
> Either
> the user is stated as Logged Out or Connection Error, usually if I
> type
> skype show users I get the following error message:
>
> [2009-08-02 08:46:05] ERROR[23719]: astobj2.c:116 INTERNAL_OBJ: bad
> magic number 0x25765ca0 for 0x1390e20
> [2009-08-02 08:46:05] ERROR[23719]: astobj2.c:116 INTERNAL_OBJ: bad
> magic number 0x25765ca0 for 0x1390e20
>
> (Debian 5.0.2 x64 running kernel 2.6.30.2, asterisk 1.6.1.1 and
> skypeforasterisk-1.6.1_0.9.10-x86_64)
>
> ___
> -- 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
___
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Register Now: http://www.astricon.net
asterisk-users mailing list
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Re: [asterisk-users] Skype for Asterisk: Public Beta available
I reported an issue on Mantis (#14). Waiting for an update. http://betareports.digium.com/mantis/view.php?id=14 Emrah Emrah wrote: > Hi Thomas, > > I am experiencing the same problem, with the same error messages. > Running Asterisk 1.6.1.0 on a Debian 2.6.24-etchnhalf.1-686 > > Regards, > Emrah > Thomas Kenyon wrote: > >> Thomas Kenyon wrote: >> >> >>> [2009-08-02 08:46:05] ERROR[23719]: astobj2.c:116 INTERNAL_OBJ: bad >>> magic number 0x25765ca0 for 0x1390e20 >>> [2009-08-02 08:46:05] ERROR[23719]: astobj2.c:116 INTERNAL_OBJ: bad >>> magic number 0x25765ca0 for 0x1390e20 >>> >>> >>> >> chef*CLI> skype show users >> Skype Users >> [2009-08-02 09:39:35] ERROR[25506]: astobj2.c:116 INTERNAL_OBJ: bad >> magic number 0x70796b73 for 0x7f4fe0044340 >> [2009-08-02 09:39:35] ERROR[25506]: astobj2.c:116 INTERNAL_OBJ: bad >> magic number 0x70796b73 for 0x7f4fe0044340 >> >> Sorry, these are the error messages. >> >> ___ >> -- 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
Re: [asterisk-users] Skype for Asterisk: Public Beta available
Hi Thomas, I am experiencing the same problem, with the same error messages. Running Asterisk 1.6.1.0 on a Debian 2.6.24-etchnhalf.1-686 Regards, Emrah Thomas Kenyon wrote: > Thomas Kenyon wrote: > >> [2009-08-02 08:46:05] ERROR[23719]: astobj2.c:116 INTERNAL_OBJ: bad >> magic number 0x25765ca0 for 0x1390e20 >> [2009-08-02 08:46:05] ERROR[23719]: astobj2.c:116 INTERNAL_OBJ: bad >> magic number 0x25765ca0 for 0x1390e20 >> >> > chef*CLI> skype show users > Skype Users > [2009-08-02 09:39:35] ERROR[25506]: astobj2.c:116 INTERNAL_OBJ: bad > magic number 0x70796b73 for 0x7f4fe0044340 > [2009-08-02 09:39:35] ERROR[25506]: astobj2.c:116 INTERNAL_OBJ: bad > magic number 0x70796b73 for 0x7f4fe0044340 > > Sorry, these are the error messages. > > ___ > -- 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
Re: [asterisk-users] Skype for Asterisk: Public Beta available
Thomas Kenyon wrote: > > [2009-08-02 08:46:05] ERROR[23719]: astobj2.c:116 INTERNAL_OBJ: bad > magic number 0x25765ca0 for 0x1390e20 > [2009-08-02 08:46:05] ERROR[23719]: astobj2.c:116 INTERNAL_OBJ: bad > magic number 0x25765ca0 for 0x1390e20 > chef*CLI> skype show users Skype Users [2009-08-02 09:39:35] ERROR[25506]: astobj2.c:116 INTERNAL_OBJ: bad magic number 0x70796b73 for 0x7f4fe0044340 [2009-08-02 09:39:35] ERROR[25506]: astobj2.c:116 INTERNAL_OBJ: bad magic number 0x70796b73 for 0x7f4fe0044340 Sorry, these are the error messages. ___ -- 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
Re: [asterisk-users] Skype for Asterisk: Public Beta available
Pascal Bruno wrote:
> Unfortunately for me, I cannot register my license. Kept saying:
>
> Could not generate Host-ID.
> Make sure that you have eth0 enabled.
>
> Any help would be appreciated
>
It uses the same licensing scheme as the G.729 licenses (so as soon as
you need to upgrade the machine, or set up LACP or VPN or any other type
of virtual interface or in the case of G.729 you change the codec to a
newer version {since you've upgraded to a new version of asterisk that
doesn't support older ones} that doesn't support the old name for the
codec, you need to re-register).
Or as in your case, it doesn't like the names of the network interfaces.
It's all a total PITA.
Fwiw, the Skype channel driver stopped working on my machine a while
ago. I never did track down the cause.
When res_skypeforasterisk starts, 39 res_skypeforasterisk processes
start and 1 skypewatcher service starts.
If I start it manually after asterisk has started, usually asterisk
segfaults, (not always).
Although Sometimes it starts up properly but can't log anyone in, Either
the user is stated as Logged Out or Connection Error, usually if I type
skype show users I get the following error message:
[2009-08-02 08:46:05] ERROR[23719]: astobj2.c:116 INTERNAL_OBJ: bad
magic number 0x25765ca0 for 0x1390e20
[2009-08-02 08:46:05] ERROR[23719]: astobj2.c:116 INTERNAL_OBJ: bad
magic number 0x25765ca0 for 0x1390e20
(Debian 5.0.2 x64 running kernel 2.6.30.2, asterisk 1.6.1.1 and
skypeforasterisk-1.6.1_0.9.10-x86_64)
___
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Re: [asterisk-users] Skype for Asterisk: Public Beta available
Unfortunately for me, I cannot register my license. Kept saying: Could not generate Host-ID. Make sure that you have eth0 enabled. On Sat, Aug 1, 2009 at 9:01 PM, Tom Browning wrote: > > Nice job. It worked right away for me with my 10 channel trial license. > Asterisk 1.4.26 > > I'm already building a dtmf access menu to bridge to my SIP world :-) > > As much I hate Skype for being a closed system, it would make the ultimate > "remote" Asterisk extension as Skype drills through so many firewalls that > block SIP and IAX and just about everything else. > > > > > > On Thu, Jul 30, 2009 at 1:50 PM, John Todd wrote: > >> >> I know many of you have been waiting for this for a while, so I'll >> keep this short: The Skype for Asterisk Public Beta is now available >> on the Digium store. >> >> We are pleased to announce the open beta of Skype For Asterisk is >> ready to begin and we look forward to you participation. To obtain >> your copy of the software, please visit Digium’s web store and >> purchase (for zero dollars) the Skype For Asterisk product. The web >> store does require a Digium.com account, which can be set up during >> the purchase process if you don’t already have one. >> Once the web store process is complete, you will be e-mailed your >> license key and directions on where to download Skype For Asterisk >> beta software. >> >> This is a "time-expiring" beta - the software will stop working on >> August 31. The download is also currently time-limited - it will be >> available until August 7 on our website. After the 31st, you would >> need to have purchased a license for the SfA software (sorry, no >> pricing that I can give you right now - that will be a separate >> announcement. I'm just the community guy - I have no idea about >> pricing or commercial contracts or the like, so please wait until >> that's been announced as I will find out about the same time as you >> do. :-) >> >> Trial "purchase" page: >> http://store.digium.com/productview.php?product_code=804-00019 >> >> JT >> >> --- >> John Todd >> email:jt...@digium.com >> Digium, Inc. | Asterisk Open Source Community Director >> 445 Jan Davis Drive NW - Huntsville AL 35806 - USA >> direct: +1-256-428-6083 http://www.digium.com/ >> >> >> >> >> ___ >> -- 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 > ___ -- 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
Re: [asterisk-users] Skype for Asterisk: Public Beta available
Unfortunately for me, I cannot register my license. Kept saying: Could not generate Host-ID. Make sure that you have eth0 enabled. Any help would be appreciated On Sat, Aug 1, 2009 at 9:01 PM, Tom Browning wrote: > > Nice job. It worked right away for me with my 10 channel trial license. > Asterisk 1.4.26 > > I'm already building a dtmf access menu to bridge to my SIP world :-) > > As much I hate Skype for being a closed system, it would make the ultimate > "remote" Asterisk extension as Skype drills through so many firewalls that > block SIP and IAX and just about everything else. > > > > > > On Thu, Jul 30, 2009 at 1:50 PM, John Todd wrote: > >> >> I know many of you have been waiting for this for a while, so I'll >> keep this short: The Skype for Asterisk Public Beta is now available >> on the Digium store. >> >> We are pleased to announce the open beta of Skype For Asterisk is >> ready to begin and we look forward to you participation. To obtain >> your copy of the software, please visit Digium’s web store and >> purchase (for zero dollars) the Skype For Asterisk product. The web >> store does require a Digium.com account, which can be set up during >> the purchase process if you don’t already have one. >> Once the web store process is complete, you will be e-mailed your >> license key and directions on where to download Skype For Asterisk >> beta software. >> >> This is a "time-expiring" beta - the software will stop working on >> August 31. The download is also currently time-limited - it will be >> available until August 7 on our website. After the 31st, you would >> need to have purchased a license for the SfA software (sorry, no >> pricing that I can give you right now - that will be a separate >> announcement. I'm just the community guy - I have no idea about >> pricing or commercial contracts or the like, so please wait until >> that's been announced as I will find out about the same time as you >> do. :-) >> >> Trial "purchase" page: >> http://store.digium.com/productview.php?product_code=804-00019 >> >> JT >> >> --- >> John Todd >> email:jt...@digium.com >> Digium, Inc. | Asterisk Open Source Community Director >> 445 Jan Davis Drive NW - Huntsville AL 35806 - USA >> direct: +1-256-428-6083 http://www.digium.com/ >> >> >> >> >> ___ >> -- 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 > ___ -- 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
Re: [asterisk-users] Skype for Asterisk: Public Beta available
Nice job. It worked right away for me with my 10 channel trial license. Asterisk 1.4.26 I'm already building a dtmf access menu to bridge to my SIP world :-) As much I hate Skype for being a closed system, it would make the ultimate "remote" Asterisk extension as Skype drills through so many firewalls that block SIP and IAX and just about everything else. On Thu, Jul 30, 2009 at 1:50 PM, John Todd wrote: > > I know many of you have been waiting for this for a while, so I'll > keep this short: The Skype for Asterisk Public Beta is now available > on the Digium store. > > We are pleased to announce the open beta of Skype For Asterisk is > ready to begin and we look forward to you participation. To obtain > your copy of the software, please visit Digium’s web store and > purchase (for zero dollars) the Skype For Asterisk product. The web > store does require a Digium.com account, which can be set up during > the purchase process if you don’t already have one. > Once the web store process is complete, you will be e-mailed your > license key and directions on where to download Skype For Asterisk > beta software. > > This is a "time-expiring" beta - the software will stop working on > August 31. The download is also currently time-limited - it will be > available until August 7 on our website. After the 31st, you would > need to have purchased a license for the SfA software (sorry, no > pricing that I can give you right now - that will be a separate > announcement. I'm just the community guy - I have no idea about > pricing or commercial contracts or the like, so please wait until > that's been announced as I will find out about the same time as you > do. :-) > > Trial "purchase" page: > http://store.digium.com/productview.php?product_code=804-00019 > > JT > > --- > John Todd > email:jt...@digium.com > Digium, Inc. | Asterisk Open Source Community Director > 445 Jan Davis Drive NW - Huntsville AL 35806 - USA > direct: +1-256-428-6083 http://www.digium.com/ > > > > > ___ > -- 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
Re: [asterisk-users] Skype for Asterisk: Public Beta available
> I have problems with it... > > [Jul 30 14:34:17] NOTICE[30529]: core.cpp:1153 skype_cp_handler: > Found license 'XX' providing 1 concurrent calls > [Jul 30 14:34:17] NOTICE[30529]: core.cpp:1000 display_host: Skype > For Asterisk Host-ID: X > [Jul 30 14:34:17] NOTICE[30529]: core.cpp:1320 sfa_startup: Found a > total of 1 Skype For Asterisk licenses > [Jul 30 14:34:21] WARNING[30613]: core.cpp:286 kill_skypewatcher: > sending SIGTERM to 30614 failed with No such process > *CLI> [Jul 30 14:34:27] ERROR[30529]: core.cpp:1551 sfa_startup: > Skype engine failed to start. > [Jul 30 14:34:27] ERROR[30529]: chan_skype.c:3032 load_module: > Unable to start Skype For Asterisk library. It is possible that you have a hung skypeforasterisk process. After exiting Asterisk, try doing: ps aux|grep skypeforasterisk If you see any processes kill (or kill -9) them and start asterisk and see if you still have the issue. Also, run "skype show licenses" and make sure that everything with the license looks ok. ___ -- 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
Re: [asterisk-users] Skype for Asterisk: Public Beta available
On 7/30/09, Steve Totaro wrote: > The first time is always free :) > > On Thu, Jul 30, 2009 at 1:50 PM, John Todd wrote: > >> >> I know many of you have been waiting for this for a while, so I'll >> keep this short: The Skype for Asterisk Public Beta is now available >> on the Digium store. >> >> We are pleased to announce the open beta of Skype For Asterisk is >> ready to begin and we look forward to you participation. To obtain >> your copy of the software, please visit Digium’s web store and >> purchase (for zero dollars) the Skype For Asterisk product. The web >> store does require a Digium.com account, which can be set up during >> the purchase process if you don’t already have one. >> Once the web store process is complete, you will be e-mailed your >> license key and directions on where to download Skype For Asterisk >> beta software. >> >> This is a "time-expiring" beta - the software will stop working on >> August 31. The download is also currently time-limited - it will be >> available until August 7 on our website. After the 31st, you would >> need to have purchased a license for the SfA software (sorry, no >> pricing that I can give you right now - that will be a separate >> announcement. I'm just the community guy - I have no idea about >> pricing or commercial contracts or the like, so please wait until >> that's been announced as I will find out about the same time as you >> do. :-) >> >> Trial "purchase" page: >> http://store.digium.com/productview.php?product_code=804-00019 >> >> JT >> >> --- >> John Todd >> email:jt...@digium.com >> Digium, Inc. | Asterisk Open Source Community Director >> 445 Jan Davis Drive NW - Huntsville AL 35806 - USA >> direct: +1-256-428-6083 http://www.digium.com/ >> >> >> >> >> ___ >> -- 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 >> > > > > -- > Thanks, > Steve Totaro > +18887771888 (Toll Free) > +12409381212 (Cell) > +12024369784 (Skype) > ___ -- 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
Re: [asterisk-users] Skype for Asterisk: Public Beta available
The first time is always free :) On Thu, Jul 30, 2009 at 1:50 PM, John Todd wrote: > > I know many of you have been waiting for this for a while, so I'll > keep this short: The Skype for Asterisk Public Beta is now available > on the Digium store. > > We are pleased to announce the open beta of Skype For Asterisk is > ready to begin and we look forward to you participation. To obtain > your copy of the software, please visit Digium’s web store and > purchase (for zero dollars) the Skype For Asterisk product. The web > store does require a Digium.com account, which can be set up during > the purchase process if you don’t already have one. > Once the web store process is complete, you will be e-mailed your > license key and directions on where to download Skype For Asterisk > beta software. > > This is a "time-expiring" beta - the software will stop working on > August 31. The download is also currently time-limited - it will be > available until August 7 on our website. After the 31st, you would > need to have purchased a license for the SfA software (sorry, no > pricing that I can give you right now - that will be a separate > announcement. I'm just the community guy - I have no idea about > pricing or commercial contracts or the like, so please wait until > that's been announced as I will find out about the same time as you > do. :-) > > Trial "purchase" page: > http://store.digium.com/productview.php?product_code=804-00019 > > JT > > --- > John Todd > email:jt...@digium.com > Digium, Inc. | Asterisk Open Source Community Director > 445 Jan Davis Drive NW - Huntsville AL 35806 - USA > direct: +1-256-428-6083 http://www.digium.com/ > > > > > ___ > -- 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 > -- Thanks, Steve Totaro +18887771888 (Toll Free) +12409381212 (Cell) +12024369784 (Skype) ___ -- 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
Re: [asterisk-users] Skype for Asterisk: Public Beta available
I have problems with it... [Jul 30 14:34:17] NOTICE[30529]: core.cpp:1153 skype_cp_handler: Found license 'XX' providing 1 concurrent calls [Jul 30 14:34:17] NOTICE[30529]: core.cpp:1000 display_host: Skype For Asterisk Host-ID: X [Jul 30 14:34:17] NOTICE[30529]: core.cpp:1320 sfa_startup: Found a total of 1 Skype For Asterisk licenses [Jul 30 14:34:21] WARNING[30613]: core.cpp:286 kill_skypewatcher: sending SIGTERM to 30614 failed with No such process *CLI> [Jul 30 14:34:27] ERROR[30529]: core.cpp:1551 sfa_startup: Skype engine failed to start. [Jul 30 14:34:27] ERROR[30529]: chan_skype.c:3032 load_module: Unable to start Skype For Asterisk library. John Todd escreveu: > I know many of you have been waiting for this for a while, so I'll > keep this short: The Skype for Asterisk Public Beta is now available > on the Digium store. > > We are pleased to announce the open beta of Skype For Asterisk is > ready to begin and we look forward to you participation. To obtain > your copy of the software, please visit Digium’s web store and > purchase (for zero dollars) the Skype For Asterisk product. The web > store does require a Digium.com account, which can be set up during > the purchase process if you don’t already have one. > Once the web store process is complete, you will be e-mailed your > license key and directions on where to download Skype For Asterisk > beta software. > > This is a "time-expiring" beta - the software will stop working on > August 31. The download is also currently time-limited - it will be > available until August 7 on our website. After the 31st, you would > need to have purchased a license for the SfA software (sorry, no > pricing that I can give you right now - that will be a separate > announcement. I'm just the community guy - I have no idea about > pricing or commercial contracts or the like, so please wait until > that's been announced as I will find out about the same time as you > do. :-) > > Trial "purchase" page: >http://store.digium.com/productview.php?product_code=804-00019 > > JT > > --- > John Todd email:jt...@digium.com > Digium, Inc. | Asterisk Open Source Community Director > 445 Jan Davis Drive NW - Huntsville AL 35806 - USA > direct: +1-256-428-6083 http://www.digium.com/ > > > > > ___ > -- 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 -- Diego Aguirre (DagMoller) Infodag Consultoria FWD#: 459696 Enum#: +55 21 8871-4916 (e164.org) DUNDi-br#: 21 8871-4916 ___ -- 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
Re: [asterisk-users] Skype for Asterisk. Any return of experience ?
On 27 Jun 2009, at 10:06, Olivier wrote: Hi, As many remember, almost one year this Skype for Asterisk extension program was announced. Has anyone tried it ? Is there any available pricelist ? I've just had a talk on Skype for Asterisk accepted at Astricon (www.astricon.net ), so if you can wait that long, you come along and I'll try and tell you what SFA can do. In the meanwhile - it often crops up on the voipusers conference (www.vuc.me ) on a Friday. In fact I've been running an experiment allowing people to call the conference from Skype (using SFA of course). Feel free to call in and try it this Friday. Tim. Tim Panton - Web/VoIP consultant and implementor www.westhawk.co.uk smime.p7s Description: S/MIME cryptographic signature ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Skype for Asterisk. Any return of experience ?
I received a phone call asking for specs, how I'd use, etc, etc and they said they'd be turning my beta account up in another 6 weeks. That was 3 weeks ago. PB On Mon, Jun 29, 2009 at 1:31 AM, randulo wrote: > >>> Though they have written me back twice to say "coming soon" I am still > >>> waiting for the software... > >>> > >> So you'd rather have it even when it hasn't been finished? > > > > Umm, no, but then when a company says "looking for beta testers - please > > sign up now!" and then four months later has nothing to let me beta test, > > I am a bit put off. > > The beta was limited. Digium wants to open it but says Skype > themselves are delaying the operation. I have compelling reasons to > believe this, even though I can't put them out in public. > > I was surprised too at the apparent slowness, but I think it will > happen in good time. > > ___ > -- Bandwidth and Colocation Provided by http://www.api-digital.com -- > > 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 -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Skype for Asterisk. Any return of experience ?
>>> Though they have written me back twice to say "coming soon" I am still >>> waiting for the software... >>> >> So you'd rather have it even when it hasn't been finished? > > Umm, no, but then when a company says "looking for beta testers - please > sign up now!" and then four months later has nothing to let me beta test, > I am a bit put off. The beta was limited. Digium wants to open it but says Skype themselves are delaying the operation. I have compelling reasons to believe this, even though I can't put them out in public. I was surprised too at the apparent slowness, but I think it will happen in good time. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Skype for Asterisk. Any return of experience ?
On Sun, 28 Jun 2009, Thomas Kenyon wrote: > Jeff LaCoursiere wrote: >> On Sun, 28 Jun 2009, randulo wrote: >> >>> On Sat, Jun 27, 2009 at 11:06 AM, Olivier wrote: Hi, Has anyone tried it ? Is there any available pricelist ? >>> It is possible no one wants to answer this due to the NDA they had to sign? >>> >> >> Though they have written me back twice to say "coming soon" I am still >> waiting for the software... >> > So you'd rather have it even when it hasn't been finished? Umm, no, but then when a company says "looking for beta testers - please sign up now!" and then four months later has nothing to let me beta test, I am a bit put off. > > I'm sure that as soon as it is complete and stable there will be pricing > and availability announced. > Indeed. j ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Skype for Asterisk. Any return of experience ?
Jeff LaCoursiere wrote: > On Sun, 28 Jun 2009, randulo wrote: > >> On Sat, Jun 27, 2009 at 11:06 AM, Olivier wrote: >>> Hi, >>> >>> Has anyone tried it ? >>> Is there any available pricelist ? >> It is possible no one wants to answer this due to the NDA they had to sign? >> > > Though they have written me back twice to say "coming soon" I am still > waiting for the software... > So you'd rather have it even when it hasn't been finished? I'm sure that as soon as it is complete and stable there will be pricing and availability announced. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Skype for Asterisk. Any return of experience ?
On Sun, 28 Jun 2009, randulo wrote: > On Sat, Jun 27, 2009 at 11:06 AM, Olivier wrote: >> Hi, >> >> Has anyone tried it ? >> Is there any available pricelist ? > > It is possible no one wants to answer this due to the NDA they had to sign? > Though they have written me back twice to say "coming soon" I am still waiting for the software... j ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] Skype for Asterisk. Any return of experience ?
On Sat, Jun 27, 2009 at 11:06 AM, Olivier wrote: > Hi, > > Has anyone tried it ? > Is there any available pricelist ? It is possible no one wants to answer this due to the NDA they had to sign? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
