Re: [libreoffice-website] Re: FYI Skype Ends Support For Open-Source Digium Asterisk VOIP PBX
Hi Friedrich, Am 31.05.2011 19:37, schrieb Friedrich Strohmaier: Hi Alex, *, Alexander Werner schrieb: I have setup a default mumble-server installation on vm3.documentfoundation.org, port 64738. Anyone who likes to test mumble, please log in with your desired username, then right klick on your user in the channel list and select register. Then drop me a short note and I will give you admin rights. I tried, but http://vm3.documentfoundation.org:64738/ only gives me an empty page.. You need to install the Mumble client from http://mumble.sourceforge.net/ to access the mumble-server. Cu, Alex -- Unsubscribe instructions: E-mail to website+h...@libreoffice.org Posting guidelines + more: http://wiki.documentfoundation.org/Netiquette List archive: http://listarchives.libreoffice.org/www/website/ All messages sent to this list will be publicly archived and cannot be deleted
Re: [libreoffice-website] Re: FYI Skype Ends Support For Open-Source Digium Asterisk VOIP PBX
Hi, Am 29.05.2011 13:28, schrieb Christoph Noack: Another option which - as far as I understand - doesn't need an own client is OpenMeetings, please see an earlier mail: http://www.mail-archive.com/marketing@libreoffice.org/msg01135.html Unforunately, the test login doesn't seem to work anymore ... so please have a look at the videos. From my point-of-view, this system would solve several issues at one (technical inexpert view). I have now also set up an OpenMeetings server for evaluation at http://vm3.documentfoundation.org:5080/openmeetings/ Feel free to register and drop me a note for rights. Cu, Alex -- Unsubscribe instructions: E-mail to website+h...@libreoffice.org Posting guidelines + more: http://wiki.documentfoundation.org/Netiquette List archive: http://listarchives.libreoffice.org/www/website/ All messages sent to this list will be publicly archived and cannot be deleted
Re: [libreoffice-website] Re: FYI Skype Ends Support For Open-Source Digium Asterisk VOIP PBX
Hi, Am 30.05.2011 13:11, schrieb Florian Effenberger: Cc'ing Alex, who IMHO has some insight on running a Mumble server. Alex, would it be suitable for our confcalls? How many resources does it need, how does it work with client firewalls? Clear advantages of mumble are low latency and high audio quality even through slow connections usig CELT and Speex codecs (with overhead, the needed bandwidth varies between 17.4kbit and 133.6kbit). (Internal) Userauthentication is based on a PKI, interconnection with other authentication sources is possible. Every connection is always AES-encrypted. Mumble is opensource and platform-independent. A disatvantage is the need to use the Qt-based client and the exclusive use of a custom protocol. There is no possibility to connect to the server via SIP. Setup of the server can be done quickly as packages exist, so evaluation is possible. Cu, Alex -- Unsubscribe instructions: E-mail to website+h...@libreoffice.org Posting guidelines + more: http://wiki.documentfoundation.org/Netiquette List archive: http://listarchives.libreoffice.org/www/website/ All messages sent to this list will be publicly archived and cannot be deleted
Re: [libreoffice-website] Re: FYI Skype Ends Support For Open-Source Digium Asterisk VOIP PBX
Hi Alex, thanks for the fast and good feedback! Alexander Werner wrote on 2011-05-30 13.45: Setup of the server can be done quickly as packages exist, so evaluation is possible. That would be indeed interesting! My take is that anything that is not accessible via phone is *not* good for us. Some people have issues with firewalls or a low-bandwith connection, and when people are on the road, they often only have a mobile with them, so a software-only-based solution doesn't sound to appealing to me. However, we can try to combine it with phones (Asterisk :-) and we can also use these tools when for a dedicated call, e.g. for certain teams, this solution without phones is enough. Florian -- Florian Effenberger flo...@documentfoundation.org Steering Committee and Founding Member of The Document Foundation Tel: +49 8341 99660880 | Mobile: +49 151 14424108 Skype: floeff | Twitter/Identi.ca: @floeff -- Unsubscribe instructions: E-mail to website+h...@libreoffice.org Posting guidelines + more: http://wiki.documentfoundation.org/Netiquette List archive: http://listarchives.libreoffice.org/www/website/ All messages sent to this list will be publicly archived and cannot be deleted
Re: [libreoffice-website] Re: FYI Skype Ends Support For Open-Source Digium Asterisk VOIP PBX
Hi all! Am Sonntag, den 29.05.2011, 01:10 +0200 schrieb Friedrich Strohmaier: Hi Florian, *, Florian Effenberger schrieb: indeed, we had the idea of setting up our own Asterisk, but at the moment, this is not feasible. [.. skype starts to exploit it's monopoly ..] [...] Probably a mumble server for audio conferencing can help. Not shure, whether this is suitable. http://mumble.sourceforge.net/ Another option which - as far as I understand - doesn't need an own client is OpenMeetings, please see an earlier mail: http://www.mail-archive.com/marketing@libreoffice.org/msg01135.html Unforunately, the test login doesn't seem to work anymore ... so please have a look at the videos. From my point-of-view, this system would solve several issues at one (technical inexpert view). Cheers, Christoph -- Unsubscribe instructions: E-mail to website+h...@libreoffice.org Posting guidelines + more: http://wiki.documentfoundation.org/Netiquette List archive: http://listarchives.libreoffice.org/www/website/ All messages sent to this list will be publicly archived and cannot be deleted
[libreoffice-website] Re: FYI Skype Ends Support For Open-Source Digium Asterisk VOIP PBX
Le 2011-05-28 03:31, Stanislas Garret a écrit : -BEGIN PGP SIGNED MESSAGE- Hash: SHA1 Here's the link : http://www.eweek.com/c/a/VOIP-and-Telephony/Skype-Ends-Support-For-Open-Source-Digium-Asterisk-VOIP-PBX-254184/ Stan Le 28/05/2011 09:07, Marc Paré a écrit : I think I read somewhere that Sophie mentioned we were setting up an Asterisk server. Looks like there will less Skype support for it. Cheers Marc Oooops! Thanks. Marc -- Unsubscribe instructions: E-mail to website+h...@libreoffice.org Posting guidelines + more: http://wiki.documentfoundation.org/Netiquette List archive: http://listarchives.libreoffice.org/www/website/ All messages sent to this list will be publicly archived and cannot be deleted
Re: [libreoffice-website] Re: FYI Skype Ends Support For Open-Source Digium Asterisk VOIP PBX
Hi, indeed, we had the idea of setting up our own Asterisk, but at the moment, this is not feasible. The Skype module support has been stopped, as Marc posted, and the SIP trunking Skype offers is insanely expensive - 6,95 USD per month per channel. So, for ten people to dial-in via Skype, it would cost us 69,50 USD per month. Getting dial-in numbers in foreign countries is also rather impossible. Per two incoming calls we would have to pay 12,90 € per month, and the next two callers would need to use another number. So, in a nutshell, an own Asterisk server makes only sense if either someone in need of a dial-in can provide us with a local VoIP number like sipgate offers or we directly use SIP Both options don't look senseful :( Florian -- Florian Effenberger flo...@documentfoundation.org Steering Committee and Founding Member of The Document Foundation Tel: +49 8341 99660880 | Mobile: +49 151 14424108 Skype: floeff | Twitter/Identi.ca: @floeff -- Unsubscribe instructions: E-mail to website+h...@libreoffice.org Posting guidelines + more: http://wiki.documentfoundation.org/Netiquette List archive: http://listarchives.libreoffice.org/www/website/ All messages sent to this list will be publicly archived and cannot be deleted
Re: [libreoffice-website] Re: FYI Skype Ends Support For Open-Source Digium Asterisk VOIP PBX
Hi Florian, *, Florian Effenberger schrieb: indeed, we had the idea of setting up our own Asterisk, but at the moment, this is not feasible. [.. skype starts to exploit it's monopoly ..] either someone in need of a dial-in can provide us with a local VoIP number like sipgate offers or we directly use SIP Both options don't look senseful :( Probably a mumble server for audio conferencing can help. Not shure, whether this is suitable. http://mumble.sourceforge.net/ Gruß/regards -- Friedrich Libreoffice-Box http://libreofficebox.org/ LibreOffice and more on CD/DVD images -- Unsubscribe instructions: E-mail to website+h...@libreoffice.org Posting guidelines + more: http://wiki.documentfoundation.org/Netiquette List archive: http://listarchives.libreoffice.org/www/website/ All messages sent to this list will be publicly archived and cannot be deleted
