Re: [asterisk-users] Asterisk in VMWare, how does it perform and what is the limit?
On Fri, Aug 7, 2009 at 7:25 PM, Pascal Bruno tipas...@gmail.com wrote: Where you able to compile DAHDI in a virtual environment? How about skype for asterisk? Has anyone tried that in a virtual environment? Seems like to register the license, digium tool is looking for a connection on eth0, and in a virtual environment I see the name as vnet0 or vnet1. At least that what I see on godaddy's virtual servers. I did that under VMWare (Server / formerly GSX), including the Skype for Asterisk, and it works (only after upgrading to 1.6.1.3-rc1, earlier version crashed after Skype call setup, but that's not related to the VM, but an asterisk bug...). Though it is merely a test environment, I haven't even tried more than one simultaneous call. HTH, -- 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
On Thu, Jul 30, 2009 at 8:50 PM, John Todd jt...@digium.com 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] single port voip gateways
On Mon, Aug 3, 2009 at 5:08 PM, Jerry Geis ge...@pagestation.com wrote: I have used the handytone 488 from grandstream in the past However I need to be able to send a number to a unit like the 488 and have it dial out. Is there a unit like this available? Basically a 488 unit that can place a call out. You are looking for a Media Gateway. Audiocodes is a well known manufacturer of these, but this is not a recommendation of any kind. There are also Channel Banks that support this (ones with FXO ports instead of FXS) HTH, -- 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
[asterisk-users] Requested transfer capability: 0x00 - SPEECH - How to change to 31KAUDIO?
Hi everyone! I'm having an issue calling the Numbers Information Service (similiar to 411 in the US) in my country. I use: TE110P, connected to a PRI line running on an E1. Besides that specific number, all calls pass through fine, in and out, no problems whatsoever. I called my Telco, and the guy did a comparison of my PRI call setup, and other calls that pass through and get fine to the Numbers Information Service. The only difference he could find, is that every other PBX in my country (mostly proprietary ones, I would assume...) - Request a transfer capability of 31KAUDIO, which I assume means 3.1KHz audio. I also assume SPEECH is actually 8KHz., which is more, and probably with higher quality. The guy at the telco said it may be the problem, maybe because the end system cannot reach the desired voice quality, or whatever (he never encountered that problem before...) I called Digium's hardware installation service, and they told me to change prilocaldialplan to unknown (I had it set to local before); I am not sure how this is related, because it seems to me not related to dialing at all - but Digium made the software and the hardware, so they must know :-) Anyways, that advice didn't really much help - my calls are still going out requesting SPEECH: -- Requested transfer capability: 0x00 - SPEECH Anybody has any advice on how to change this? Thanks! -- Shimi ___ --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
