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
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
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
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
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
Le 2011-05-28 03:31, Stanislas Garret a écrit :
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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
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,
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
