Re: [asterisk-users] Astricon people please post the announcement
On Sep 25, 2008, at 9:00 PM, Darrick Hartman wrote: Dean Collins wrote: I'd also like to know what happens when someone 'chats' to the account connected to the Asterisk server. Lots of questions about this one. There's definitely a demand for it so I can see why Digium would be interested in exploring this option. Time will tell how well it will work. I'm personally not too excited about bolt-on binaries which are probably not compatible with uClibc (and therefore Astlinux). That leaves us in the same place as we are with codec_g729. We're at the mercy of whoever creates these binaries to produce one that will work for us. Darrick According to them today, if the user initiates calls with someone outside the pbx, it will not go through the pbx. The user can register both to skyp and the asterisk also register the user. So, if the user initiates contact to another it is peer to peer outside the pbx. smime.p7s Description: S/MIME cryptographic signature ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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] Astricon people please post the announcement
On Thu, Sep 25, 2008 at 05:25:52PM -0700, Fred Posner wrote: On Sep 25, 2008, at 11:06 AM, Steve Anness wrote: So what a minute. They will charge us to use Skype with our Asterisk servers? Yes, I think I shall move along. Steve I talked with both Skype and Digium today at Astricon for a while on this... it's actually going to be amazing. The license for Skype will be the same way you license g.729. So yes, it's not free... but you're only paying for in use channel capabilities... There are already a number of such Skype connectors. Some of them claim to be free (that is: no charge). Some of them take money. I'm not sure if Skype/eBay sees any of that. They tend to at least bend Skype's license if not break it completely (e.g: run a client in a XNest server is a common trick to work around the requirement in the license of the Skype API for an interactive client). So now that we have a blessed client for which Skype/eBay gets payed, what happens to those others? Will they still be legal? -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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] Astricon people please post the announcement
On Thu, 25 Sep 2008 18:00:00 +0100, Tim Panton wrote: It's essentially a channel driver. Licensed per channel in the same way that the g729 codec is. which would mean that us freebsd folks are going to be left out. oh well. -- Regards, /\_/\ All dogs go to heaven. [EMAIL PROTECTED](0 0) http://www.openmalaysiablog.com/ +==oOO--(_)--OOo==+ | for a in past present future; do| | for b in clients employers associates relatives neighbours pets; do | | echo The opinions here in no way reflect the opinions of my $a $b. | | done; done | +=+ ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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] Astricon people please post the announcement
I have tryed skip2pbx 580€ yeastar 60 €, the quality is the way behind of a good sip provider, thay are simply not suitable for business, i hope it would not be the case of asterisk addon. Also i wonder if skype auto relay will be disabled (bandwith), wait and see... ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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] Astricon people please post the announcement
On 26 Sep 2008, at 11:17, Grygoriy Dobrovolskyy wrote: I have tryed skip2pbx 580€ yeastar 60 €, the quality is the way behind of a good sip provider, thay are simply not suitable for business, i hope it would not be the case of asterisk addon. Also i wonder if skype auto relay will be disabled (bandwith), wait and see... The Asterisk team said that a) the skype for asterisk code does not act as a supernode - i.e. it only routes traffic for local users, this was one of their requirements. b) they _think_ that in the case where both ends of a skype to skype call are 'local' the huge majority of the bandwidth remains local. c) there will be configuration options controlling which of the transport methods skype for asterisk will use. So you can disable skype over port 443 if you want to ensure that port is available for your ssl webserver (for example) Tim. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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] Astricon people please post the announcement
On 26 Sep 2008, at 04:36, Dean Collins wrote: I'd also like to know what happens when someone 'chats' to the account connected to the Asterisk server. I asked Mark about that. They expect to have text to work right, when associated with a voice call. It is less clear what happens it it is _just_ a text session. Olle tells me that 1.6 can do text only calls (he's been working on an asterisk for the deaf project) so there is a decent chance they will get it to work. Tim. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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] Astricon people please post the announcement
Get Olle to call in for once in his life! Mark did say IM and video, IM first. It's all gonna happen. (just not right away) On Fri, Sep 26, 2008 at 3:34 PM, Tim Panton [EMAIL PROTECTED] wrote: On 26 Sep 2008, at 04:36, Dean Collins wrote: I'd also like to know what happens when someone 'chats' to the account connected to the Asterisk server. I asked Mark about that. They expect to have text to work right, when associated with a voice call. It is less clear what happens it it is _just_ a text session. Olle tells me that 1.6 can do text only calls (he's been working on an asterisk for the deaf project) so there is a decent chance they will get it to work. Tim. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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 2008 - September 22 - 25 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] Astricon people please post the announcement
On Fri, Sep 26, 2008 at 03:44:01PM +0200, randulo wrote: Get Olle to call in for once in his life! Mark did say IM and video, IM first. It's all gonna happen. (just not right away) On the topic of #pidgin they say, amomng others, Pidgin does NOT support voice or video. Likewise we should state on #asterisk Asterisk does NOT support text chats. There are some awkward methods for sending some text messages over some channels (SMS in european POTS, SIMPLE and simpler texxt messages in SIP, XMMP for Jingle, and well, probably nothing in IAX. Bristuff ads even a few more bits there). But do we actually care routing those messages from one place to another? This is a major limitation of Asterisk for me. Text messages require much lower a bandwith and a text connection is much easier to setup. Hence it can work even when a voip connection is lousy. -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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] Astricon people please post the announcement
2008/9/26 randulo [EMAIL PROTECTED] Get Olle to call in for once in his life! Mark did say IM and video, IM first. It's all gonna happen. (just not right away) http://lists.digium.com/mailman/listinfo/asterisk-users Video ? that could be really nice but limited to pc/macasteriskwhatever. There are tonns of 3G phones on the market, so why not to adapt software fot the videocalls over wifi ? such a client is my dream for about a year, and i dont care it it would be a skype or else. A new product for that purpose is not a solution, but adapting software to existing 3G phones will open a HUGE market recently created and closed for 3G operators w/licence. Any suggestions ? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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] Astricon people please post the announcement
On Fri, Sep 26, 2008 at 11:59:35AM +0300, Tzafrir Cohen wrote: On Thu, Sep 25, 2008 at 05:25:52PM -0700, Fred Posner wrote: On Sep 25, 2008, at 11:06 AM, Steve Anness wrote: So what a minute. They will charge us to use Skype with our Asterisk servers? Yes, I think I shall move along. Steve I talked with both Skype and Digium today at Astricon for a while on this... it's actually going to be amazing. The license for Skype will be the same way you license g.729. So yes, it's not free... but you're only paying for in use channel capabilities... There are already a number of such Skype connectors. Some of them claim to be free (that is: no charge). Some of them take money. I'm not sure if Skype/eBay sees any of that. They tend to at least bend Skype's license if not break it completely (e.g: run a client in a XNest server is a common trick to work around the requirement in the license of the Skype API for an interactive client). So now that we have a blessed client for which Skype/eBay gets payed, what happens to those others? Will they still be legal? I wonder if http://narod.ru/disk/2812178000/asterisk-skype.gz.html is legal. -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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] Astricon people please post the announcement
Brian J. Murrell wrote: And so will this channel driver also allow Skype to use my resources (CPU, bandwidth -- i.e. Internet for which many have usage caps, etc.) the way the Skype client does? The Skype engine in Skype For Asterisk does not currently have 'relay' support, so it does not route calls or media any calls that it is not involved in. However, this will be present in the production release of the product, but when it appears we will also document its behavior and the configuration options that can be used to control it. -- Kevin P. Fleming Director of Software Technologies Digium, Inc. - The Genuine Asterisk Experience (TM) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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] Astricon people please post the announcement
- Tzafrir Cohen [EMAIL PROTECTED] wrote: There are some awkward methods for sending some text messages oversome channels (SMS in european POTS, SIMPLE and simpler texxt messages in SIP, XMMP for Jingle, and well, probably nothing in IAX. Bristuff ads even a few more bits there). But do we actually care routing those messages from one place to another? This is a major limitation of Asterisk for me. Text messages require much lower a bandwith and a text connection is much easier to setup. Hence it can work even when a voip connection is lousy. The specific thing here (that makes handling text messages within the framework of the more complicated protocol attractive) is *addressibility*. If you already have a path to someone, why should you be forced to *discover* another path to them for some other, simpler protocol? Cheers, -- jra -- Jay R. Ashworth Baylink [EMAIL PROTECTED] Designer The Things I Think RFC 2100 Ashworth Associates http://baylink.pitas.com '87 e24 St Petersburg FL USA http://photo.imageinc.us +1 727 647 1274 Those who cast the vote decide nothing. Those who count the vote decide everything. -- (Josef Stalin) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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] Astricon people please post the announcement
Dean Collins wrote: I'd also like to know what happens when someone 'chats' to the account connected to the Asterisk server. Keeping in mind that the product has not yet entered beta testing... at this time, all chat messages are ignored by the Skype For Asterisk product. We have discussed today the possibility of being able to send chats to Skype users (sort of a way to do 'screen pop' information for a call you are about to send them), but haven't got any plans at the moment for incoming chat messages. -- Kevin P. Fleming Director of Software Technologies Digium, Inc. - The Genuine Asterisk Experience (TM) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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] Astricon people please post the announcement
2008/9/26 Kevin P. Fleming [EMAIL PROTECTED] Brian J. Murrell wrote: And so will this channel driver also allow Skype to use my resources (CPU, bandwidth -- i.e. Internet for which many have usage caps, etc.) the way the Skype client does? The Skype engine in Skype For Asterisk does not currently have 'relay' support, so it does not route calls or media any calls that it is not involved in. However, this will be present in the production release of the product, but when it appears we will also document its behavior and the configuration options that can be used to control it. -- Kevin P. Fleming Director of Software Technologies Digium, Inc. - The Genuine Asterisk Experience (TM) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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 Will it be packed into the base asterisk package, or to asterisk-addons? or into some third party ? Would it be possible to buy some comminication licences use them while disabling the 'relay' function ? ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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] Astricon people please post the announcement
Grygoriy Dobrovolskyy wrote: Will it be packed into the base asterisk package, or to asterisk-addons? or into some third party ? Would it be possible to buy some comminication licences use them while disabling the 'relay' function ? Skype For Asterisk will be distributed as a separate package. We do not know yet what (if any) requirements will have to be handled for disabling the relay functionality. -- Kevin P. Fleming Director of Software Technologies Digium, Inc. - The Genuine Asterisk Experience (TM) ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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
[asterisk-users] Astricon people please post the announcement
Digium is making a big announcement today at Astricon. So who's gonna post this and where? I must know before I go to sleep. It may change my life! r ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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] Astricon people please post the announcement
So Skype finally will talk to Asterisk Excellent news! On Thu, Sep 25, 2008 at 6:17 PM, randulo [EMAIL PROTECTED] wrote: Digium is making a big announcement today at Astricon. So who's gonna post this and where? I must know before I go to sleep. It may change my life! r ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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] Astricon people please post the announcement
http://bit.ly/asterskype ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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] Astricon people please post the announcement
So does this mean that my users who currently have skype running on their systems won't have to install anything new once I get things rolling on the Asterisk server? Steve On 9/25/08 11:38 AM, randulo [EMAIL PROTECTED] wrote: So Skype finally will talk to Asterisk Excellent news! On Thu, Sep 25, 2008 at 6:17 PM, randulo [EMAIL PROTECTED] wrote: Digium is making a big announcement today at Astricon. So who's gonna post this and where? I must know before I go to sleep. It may change my life! r ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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 2008 - September 22 - 25 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] Astricon people please post the announcement
I know a lot of linux and open source people think it's superfluous, but a pseudo chan_skype is huge (assuming it works as advertised). It means anyone with Skype can connect to your server presence. And presumably you can call people via Skype. And use Skype out, etc. On Thu, Sep 25, 2008 at 6:47 PM, Steve Anness [EMAIL PROTECTED] wrote: So does this mean that my users who currently have skype running on their systems won't have to install anything new once I get things rolling on the Asterisk server? Steve On 9/25/08 11:38 AM, randulo [EMAIL PROTECTED] wrote: So Skype finally will talk to Asterisk Excellent news! On Thu, Sep 25, 2008 at 6:17 PM, randulo [EMAIL PROTECTED] wrote: Digium is making a big announcement today at Astricon. So who's gonna post this and where? I must know before I go to sleep. It may change my life! r ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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 2008 - September 22 - 25 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 2008 - September 22 - 25 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] Astricon people please post the announcement
It's essentially a channel driver. Licensed per channel in the same way that the g729 codec is. Limited private beta opening soon. Tim. On 25 Sep 2008, at 17:47, Steve Anness wrote: So does this mean that my users who currently have skype running on their systems won't have to install anything new once I get things rolling on the Asterisk server? Steve On 9/25/08 11:38 AM, randulo [EMAIL PROTECTED] wrote: So Skype finally will talk to Asterisk Excellent news! On Thu, Sep 25, 2008 at 6:17 PM, randulo [EMAIL PROTECTED] wrote: Digium is making a big announcement today at Astricon. So who's gonna post this and where? I must know before I go to sleep. It may change my life! r ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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 2008 - September 22 - 25 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 2008 - September 22 - 25 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] Astricon people please post the announcement
They demoed it - everyone seems pretty confident it works as advertized. No wide-band codec (yet) Tim. On 25 Sep 2008, at 17:55, randulo wrote: I know a lot of linux and open source people think it's superfluous, but a pseudo chan_skype is huge (assuming it works as advertised). It means anyone with Skype can connect to your server presence. And presumably you can call people via Skype. And use Skype out, etc. On Thu, Sep 25, 2008 at 6:47 PM, Steve Anness [EMAIL PROTECTED] wrote: So does this mean that my users who currently have skype running on their systems won't have to install anything new once I get things rolling on the Asterisk server? Steve On 9/25/08 11:38 AM, randulo [EMAIL PROTECTED] wrote: So Skype finally will talk to Asterisk Excellent news! On Thu, Sep 25, 2008 at 6:17 PM, randulo [EMAIL PROTECTED] wrote: Digium is making a big announcement today at Astricon. So who's gonna post this and where? I must know before I go to sleep. It may change my life! r ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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 2008 - September 22 - 25 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 2008 - September 22 - 25 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 2008 - September 22 - 25 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] Astricon people please post the announcement
Hi all, Voiceroute is twittering abt it http://twitter.com/voiceroute Video with mark on announcement will be uploaded in 1 hour. http://youtube.com/voiceroute Ming On 9/25/08, randulo [EMAIL PROTECTED] wrote: Digium is making a big announcement today at Astricon. So who's gonna post this and where? I must know before I go to sleep. It may change my life! r ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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 -- Sent from Gmail for mobile | mobile.google.com Ming Yong CEO, www.voiceroute.org Druid - Open Source Unified Communications DID: +1-877-242-3704 Office: +1-866-915-2407 ext 301 SIP/email: [EMAIL PROTECTED] -- Attend Vikram's talk at ASTRICON 2008, 23-25 Sept 08, Glendale Arizona http://www.astricon.net/2008/glendale/web/confTracks.php#t193 Meet us at WEB 2.0 EXPO, 17-18 Sept 08, Booth #17 in Long Tail Pavilion, Javits Center, NYC http://druidweb20.eventbrite.com DruidCON 1-2 Oct Atlanta GA (The leading UC 2.0 Application platform) http://www.voiceroute.org/druidcon VoiceCON 08 San Francisco 10-13 Nov 08, Booth #738, Moscone North Convention Center, San Francisco, CA http://druidvoicecon.eventbrite.com UC 2.0 Video - Mozilla Ubiquity + Druid http://www.youtube.com/watch?v=f-5rDBPuGRc ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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] Astricon people please post the announcement
On Thu, Sep 25, 2008 at 06:38:24PM +0200, randulo wrote: So Skype finally will talk to Asterisk Excellent news! Great news! You mean that there is finally a free implementation of the skype protocol so I can start using it? -- Tzafrir Cohen icq#16849755 jabber:[EMAIL PROTECTED] +972-50-7952406 mailto:[EMAIL PROTECTED] http://www.xorcom.com iax:[EMAIL PROTECTED]/tzafrir ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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] Astricon people please post the announcement
On Thu, 2008-09-25 at 20:49 +0300, Tzafrir Cohen wrote: Great news! You mean that there is finally a free implementation of the skype protocol so I can start using it? Free? AFAICT, not. Neither free as in beer nor speech. Move along, nothing to see here. b. signature.asc Description: This is a digitally signed message part ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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] Astricon people please post the announcement
So what a minute. They will charge us to use Skype with our Asterisk servers? Yes, I think I shall move along. Steve On 9/25/08 12:53 PM, Brian J. Murrell [EMAIL PROTECTED] wrote: On Thu, 2008-09-25 at 20:49 +0300, Tzafrir Cohen wrote: Great news! You mean that there is finally a free implementation of the skype protocol so I can start using it? Free? AFAICT, not. Neither free as in beer nor speech. Move along, nothing to see here. b. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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 2008 - September 22 - 25 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] Astricon people please post the announcement
How is this exactly news? Hasn't chan_skype been around and available for a while now? How is this different? Eric On Thu, Sep 25, 2008 at 9:38 AM, randulo [EMAIL PROTECTED] wrote: So Skype finally will talk to Asterisk Excellent news! On Thu, Sep 25, 2008 at 6:17 PM, randulo [EMAIL PROTECTED] wrote: Digium is making a big announcement today at Astricon. So who's gonna post this and where? I must know before I go to sleep. It may change my life! r ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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 2008 - September 22 - 25 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] Astricon people please post the announcement
On Sep 25, 2008, at 11:06 AM, Steve Anness wrote: So what a minute. They will charge us to use Skype with our Asterisk servers? Yes, I think I shall move along. Steve I talked with both Skype and Digium today at Astricon for a while on this... it's actually going to be amazing. The license for Skype will be the same way you license g.729. So yes, it's not free... but you're only paying for in use channel capabilities... but think of the benefits... Skype will work just like let's say SIP or ZAP or IAX2 would in the dial plan... so you could have SKYPE\username registered as an extension... or ... even in a queue which I am really excited about. You could registered as many usernames as you want... and then have as many simultaneous calls as licenses... great for calling out and even really awesome for calling in. Plus, unlike regular skype, on the callin, you can have multiple channels. It's really very exciting. Fred Posner [EMAIL PROTECTED] Tel: +1 (212) 937-7844 x501 Fax: +1 (954) 252-4187 www.teamforrest.com Using VoIP? SIP:[EMAIL PROTECTED] smime.p7s Description: S/MIME cryptographic signature ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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] Astricon people please post the announcement
On Thu, 2008-09-25 at 17:25 -0700, Fred Posner wrote: I talked with both Skype and Digium today at Astricon for a while on this... it's actually going to be amazing. It's still early, but still, nobody has answered my question as to whether Skype will be using my Asterisk server's CPU and bandwidth to bridge calls between anonymous third parties (i.e. two people not involved in my call plan in any way, just using my Asterisk server as a bridge for their lame, NATted connectivity) they way they do with their client. Y'all do realize with Skype that they bridge calls between two parties using a third, anonymous, (donor) party, when the two parties cannot connect to each other because their NAT and/or firewalls are too restrictive to allow them to connect directly, right? b. signature.asc Description: This is a digitally signed message part ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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] Astricon people please post the announcement
I'd also like to know what happens when someone 'chats' to the account connected to the Asterisk server. Cheers, Dean -Original Message- From: [EMAIL PROTECTED] [mailto:[EMAIL PROTECTED] On Behalf Of Brian J. Murrell Sent: Thursday, 25 September 2008 11:25 PM To: Asterisk Users Mailing List - Non-Commercial Discussion Subject: Re: [asterisk-users] Astricon people please post the announcement On Thu, 2008-09-25 at 17:25 -0700, Fred Posner wrote: I talked with both Skype and Digium today at Astricon for a while on this... it's actually going to be amazing. It's still early, but still, nobody has answered my question as to whether Skype will be using my Asterisk server's CPU and bandwidth to bridge calls between anonymous third parties (i.e. two people not involved in my call plan in any way, just using my Asterisk server as a bridge for their lame, NATted connectivity) they way they do with their client. Y'all do realize with Skype that they bridge calls between two parties using a third, anonymous, (donor) party, when the two parties cannot connect to each other because their NAT and/or firewalls are too restrictive to allow them to connect directly, right? b. ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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] Astricon people please post the announcement
Dean Collins wrote: I'd also like to know what happens when someone 'chats' to the account connected to the Asterisk server. Lots of questions about this one. There's definitely a demand for it so I can see why Digium would be interested in exploring this option. Time will tell how well it will work. I'm personally not too excited about bolt-on binaries which are probably not compatible with uClibc (and therefore Astlinux). That leaves us in the same place as we are with codec_g729. We're at the mercy of whoever creates these binaries to produce one that will work for us. Darrick ___ -- Bandwidth and Colocation Provided by http://www.api-digital.com -- AstriCon 2008 - September 22 - 25 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
