Fwd: FOSDEM 2013 Telephony/Communications Devroom Call for Presenters

Daniel Pocock daniel at pocock.com.au
Sat Dec 1 23:51:41 UTC 2012

The panel discussion and integration test effort below will hopefully be
a useful follow up to the FSF Europe "Hunt for a Skype alternative"

I hope to meet some of you at FOSDEM in February

-------- Original Message --------
Subject: 	[asterisk-dev] FOSDEM 2013 Telephony/Communications Devroom
Call for Presenters
Date: 	Sat, 24 Nov 2012 20:37:28 -0600
From: 	Matthew Jordan <mjordan at digium.com>
Reply-To: 	Asterisk Developers Mailing List <asterisk-dev at lists.digium.com>
To: 	Asterisk Developers Mailing List <asterisk-dev at lists.digium.com>


This is a call for presenters for the telephony and communications
devroom at FOSDEM 2013 - http://www.fosdem.org/.

We will be holding a day full of presentations on development topics in
the area of open source telephony and communications on Sunday, February
3rd. The schedule allows for presentations from 9:00 to 15:00. Following
the development topic presentations, a panel discussing the challenges
of federated and distributed free Real-time Communications (RTC) and its
significance to the Free and Open Source Software community will be
held, hosted by Daniel Pocock and Peter Saint-Andre.

Please submit all proposals no later than 2013-01-04. Notification of
accepted speakers will be provided by 2013-01-07. We will then work to
have a schedule finalized by 2013-01-11. Please include the following
with your submission:

1. Title of your presentation.
2. The duration of your talk (between 20 and 60 minutes). Please
indicate the minimum and maximum size slot that you are comfortable with.
3. The speaker(s) presenting.
4. A brief description of the content of the presentation.

This year, proposals will be reviewed and approved by a panel. That
panel consists of:
 * Daniel Pocock <daniel at pocock.com.au>
 * Peter Saint-Andre <stpeter at stpeter.im>
 * Matt Jordan <mjordan at digium.com>

If you would like to contact the devroom organizer directly, please
contact Matt Jordan <mjordan at digium.com>.

The room we will be using will have a projector, wifi, and 150 seats.

In addition to the Telephony DevRoom, this year we will be hosting a
Real Time Communication Integration Test effort. Projects that are
incorporating any RTC or VoIP capabilities are invited to participate in
the integration test efforts. This will be similar to SIPit, but
smaller, for open source projects, and independent of protocol. More
information regarding the RTC integration testing will be made available
on the telephony-devroom mailing list.

Feel free to forward this along to any people or mailing lists that you
think would be interested in this event.

Thank you!

Matthew Jordan
Digium, Inc. | Engineering Manager
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: http://digium.com & http://asterisk.org

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