Hi All, As in previous years, Peter Gradwell of gradwell.com continues to sponsor Debian with free VoIP calls -- the only problem being that this year we don't have anything set up to let you use it.
As you may have noticed, I'm not at DebConf (sorry to be missing the fun, but it gives me a chance to watch the streams -- so many thanks to the videoteam for ensuring that I'm not getting enough sleep :-) ) That of course also means that I've not brought my desk phone to DC14. Also the VoIP server we used in previous years is no longer with us, so we had hoped to use the new rtc.debian.org setup, and route calls via Daniel Pocock's Asterisk server to get them to Gradwell, but it turns out that recent updates in WebRTC make that difficult, and Daniel may not have time to work around things before the end of Debconf. That being the case, if someone can provide a SIP phone, or an ATA + a phone, or failing that a laptop (with Jitsi and a headset) or a smartphone running Lumicall, that can be left somewhere that people can use it (Frontdesk? Hacklab?), then I should be able to set that up as an extension on my Freeswitch server, and thus route the calls to Gradwell. Of course, I could set up more than one extension, so if people are interested I could set them up with individual accounts, but that's going to be considerably more effort, because all the nice integration with LDAP went when the old server was deleted, so I'll be doing each one by hand -- feel free to persuade me that it's worth it -- I'm liable to be easy to persuade if you volunteer to let people use your phone (or whatever) once it's set up to make calls. Note that I'm 8 hours ahead of Portland (in Germany). Cheers, Phil. -- |)| Philip Hands [+44 (0)20 8530 9560] HANDS.COM Ltd. |-| http://www.hands.com/ http://ftp.uk.debian.org/ |(| Hugo-Klemm-Strasse 34, 21075 Hamburg, GERMANY
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