Re: [asterisk-users] Skype for SIP
On 15/6/10 06:22, Randy R wrote: By the way, I am currently testing this product from Skype. I would like to be able to receive calls ona Skype name on our pbx. 1) It works beautifully and you don't have to do anything in particular. 2) It's disproportionally expensive which is why I want Skype for Asterisk to work. SfS costs $5 per month per channel just to test the beta! I find that insane, but I wanted to test it. In October, they will begin charging for Skype Manager (required for SfS) and a per seat charge for that. SfA also requires Skype Manager, and only works with users that were created with it. (At Skypes insistance afaict). The only architectures supported by SfA at the moment are x86 and x86-64. Also afaik, video still doesn't work with 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 SIP
On Tue, Jun 15, 2010 at 9:46 AM, Thomas Kenyon dig...@sanguinarius.co.uk wrote: On 15/6/10 06:22, Randy R wrote: In October, they will begin charging for Skype Manager (required for SfS) and a per seat charge for that. SfA also requires Skype Manager, and only works with users that were created with it. (At Skypes insistance afaict). Yes, correct. And they are charging per name starting this fall. SO if the names die when you don't pay for Skype Manager, screw it and Skype which I personally never use, that will become expensive for little benefit. We'll teach people to use SIP clients which are free or fixed cost. The only architectures supported by SfA at the moment are x86 and x86-64. ok. Also afaik, video still doesn't work with it. Don't care about that. -- _ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- New to Asterisk? Join us for a live introductory webinar every Thurs: http://www.asterisk.org/hello asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
[asterisk-users] Skype for SIP
By the way, I am currently testing this product from Skype. I would like to be able to receive calls ona Skype name on our pbx. 1) It works beautifully and you don't have to do anything in particular. 2) It's disproportionally expensive which is why I want Skype for Asterisk to work. SfS costs $5 per month per channel just to test the beta! I find that insane, but I wanted to test it. In October, they will begin charging for Skype Manager (required for SfS) and a per seat charge for that. My guess is that the one channel would cost me about $15/month. For our use, I'd much rather use SfA. You can test SfS is you like by just registering a SIP phone to their servers. That's what I am doing now. I haven't tested outgoing calls to SKype names, but they are possible. Probably, like OpenSkY they are made using numbers in a table that equate to Skype names. /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 TO SIP (Was SIP to Skype)
I noticed a quote from you at GigOM: Even if this is complete vaporware at least their heart is in the right direction. Last year they announced Skype for Asterisk which is still not yet released and it’s unclear what the pricing will be. Skype For SIP is similar in that it is not yet available and pricing details are murky but both are steps in the right direction. Om uses a title to his post that is something like Michael Robertson says SIP for 'Skype is vaporware' Is that what you were saying? Or did you mean even if it were vaporware? I've emailed you separately about something else. /r ___ -- 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
[asterisk-users] Skype TO SIP (Was SIP to Skype)
From: Guillermo Salas M. gsa...@manta.telconet.net http://www.gizmo5.com/opensky Free calls are available up to 5 minutes. If you need longer calls there's a commercial service you can purchase. Can be used to receive calls from skype? Yes it can. For example anyone who calls me now on Skype at michaelGizmo5 it will ring the IP phone connected to our Asterisk system, my SIP software and my Nokia e71i wifi phone. It is really liberating to have my Skype identity ring anywhere *I* like! This is configured at my.gizmo5.com where the user enters in their Skype credentials. Once they do that they will have the ability to forward their calls to any SIP address. This may be impractical for larger network use. For that scenario Gizmo5 is building an API so that remote networks can easily activate the service without requiring users to visit the Gizmo5 website. This will be part of the commercial service for OpenSky. If you would like to test this, please send me email. -- MR ___ -- 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 SIP
On 23 Mar 2009, at 19:42, Gordon Henderson wrote: Anyone connected up to it yet? http://www.skypeforsip.com/ It would seem to make Digiums chan_skype rather pointness, or am I missing something? Or is this Digiums chan_skype in a hosted box somewhere? Gordon There are fewer limitations to SFA than SFS. SFA gets presence and full user info, plus it can make calls to Skype users, which SFS cant. I'm hoping that Digium will extend this difference by adding support for text and perhaps video... Here's an example of something SFS can't do: sfa.westhawk.co.uk/skype/call.xsql?key=echo123 (a quick demo I knocked up with the SFA beta) 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 SIP
On Tue, Mar 24, 2009 at 9:37 AM, Tim Panton t...@westhawk.co.uk wrote: There are fewer limitations to SFA than SFS. SFA gets presence and full user info, plus it can make calls to Skype users, which SFS cant. I'm hoping that Digium will extend this difference by adding support for text and perhaps video... Of course if we're not currently running asterisk, SFS looks attractive. I'm interested in trying it and hope to report on the performance. /r ___ -- 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
[asterisk-users] Skype for SIP
Anyone connected up to it yet? http://www.skypeforsip.com/ It would seem to make Digiums chan_skype rather pointness, or am I missing something? Or is this Digiums chan_skype in a hosted box somewhere? Gordon ___ -- 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 SIP
I wonder why they only allow G.729 with this ... where's the great sound of the skype call now ? Martin On Mon, Mar 23, 2009 at 2:42 PM, Gordon Henderson gordon+aster...@drogon.net gordon%2baster...@drogon.net wrote: Anyone connected up to it yet? http://www.skypeforsip.com/ It would seem to make Digiums chan_skype rather pointness, or am I missing something? Or is this Digiums chan_skype in a hosted box somewhere? Gordon ___ -- 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
[asterisk-users] skype and SIP hardware for linux
I'm looking at the http://support.a-link.com/phonemate/IPU1.htm phone because it works with Skype (from Linux), but can do SIP, too. Not necessarily asterisk related, but possibly. My networking situation might require IAX if I'm running Linux and want to use SIP, I'm not certain (Skype works fine). Putting that unknown aside for the moment, how does this phone work under either Skype or as a SIP phone? The information I have on the driver, skypemate, is a bit sketchy. According to A-Link, the phone complies with SIP, http://www.a-link.com/us_us/IPU1.html, but the details are sketchy. No information is provided as to the interface for configuring SIP. The user manual, http://support.a-link.com/phonemate/Manual/IPU1manual_for_Linux.pdf, details using Skype but not SIP. Any user experience with this phone? For instance, has anyone used it with gizmo project or free world dialup, or even Skype? thanks, Thufir ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
Re: [asterisk-users] skype and SIP hardware for linux
It''s a USB Sound card / keypad / display, not a phone. It contols a softphone on the PC it's plugged into - they say it works with XLite - the SIP setup will be done in Xlite, not the 'phone'. Peter On 05/11/06, Thufir [EMAIL PROTECTED] wrote: I'm looking at the http://support.a-link.com/phonemate/IPU1.htm phone because it works with Skype (from Linux), but can do SIP, too. Not necessarily asterisk related, but possibly. My networking situation might require IAX if I'm running Linux and want to use SIP, I'm not certain (Skype works fine). Putting that unknown aside for the moment, how does this phone work under either Skype or as a SIP phone? The information I have on the driver, skypemate, is a bit sketchy. According to A-Link, the phone complies with SIP, http://www.a-link.com/us_us/IPU1.html, but the details are sketchy. No information is provided as to the interface for configuring SIP. The user manual, http://support.a-link.com/phonemate/Manual/IPU1manual_for_Linux.pdf, details using Skype but not SIP. Any user experience with this phone? For instance, has anyone used it with gizmo project or free world dialup, or even Skype? thanks, Thufir ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users -- Peter Bowyer Email: [EMAIL PROTECTED] ___ --Bandwidth and Colocation provided by Easynews.com -- asterisk-users mailing list To UNSUBSCRIBE or update options visit: http://lists.digium.com/mailman/listinfo/asterisk-users
