FS weekly 2012 10 17

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Contents

Scheduled time

Wednesday October 17th, 2012 at 1700 UTC (1200 CDT / 1300 EDT)

See in your time zone.

Calling Instructions

  • SIP
    • sip:888@conference.freeswitch.org
    • Codecs supported: PCMU/PCMA, G.722, CELT, Speex, among many others
  • PSTN
Country Number Contributor
USA +1-919-386-9900
Spain +34-91-290-12-71 Thanks to SIPtize
UK +44-330-320-0105 Thanks to ziron.net
UK +44-1904-201-313 Thanks to ukddi.com (Routed by Steven Ayre)
UK +44-203-298-5931 Thanks to ukddi.com (Routed by Avi Marcus)
Germany +49-228-9293-9009 Thanks to Yiftach at ChooChee
Germany +49-2373-913-4009 Thanks to einfachVoIP.de
Australia +61-7-3188-7519 Thanks to Jay Binks - NetSIP.com.au
Israel +972-2-372-0394 Thanks to Avi Marcus - BestFone.com
Canada +1-438-800-0531 Thanks to NG Communications
France +33-975-181-606 Thanks to NG Communications
Netherlands +31-858-880-387 Thanks to NG Communications
South Africa +27-87-8204656 Thanks to Othos Telecom
Portugal +351-300505224 Thanks to Finesource

Note: When you join you will initially be muted and need to press 0 if you wish to speak (this reduces background noise on the conference).

At some times muting may be moderated (eg presentations). Pressing 0 will put you in a queue and you will be unmuted when it's your turn to speak.

Be sure to be logged into #freeswitch on irc.freenode.net! There are frequently related comments during the conference.

What are the Beeps in the conference? One Beep - someone has joined, two beeps - someone has left.

Agenda

Please add agenda items as needed. If you have a question you'd like to ask then edit the Items Needing Discussion section below. Be sure to put your name/nick on your question.

News, Notes, & Miscellaneous Fun Stuff

  • FS Book - Packt Pub looking for technical reviewers!
    • Requires real effort, so if you're interested please contact me via email (msc@freeswitch.org)

Items Needing Discussion

  • Chad Philips (hunmonk) is going to give us a brief overview of the process of getting his Jester project up and running over at github.

Featured Presentation

Questions For Developers

  • Add your questions here

Janitorial Items

  • Add your questions here

Items Needing Documentation

  • Revised Install guide, started here: V1.2_Download_&_Installation_Guide
  • CDR Handling main page:
    • pulling legs of a single calls together
      • from the A leg
    • determining PDD
    • ring time
    • missed call notification (as opposed to actual VM)
    • Noting phones that didn't respond to incoming call, despite registration
  • Variables:
    • bypass_media_after_bridge

Stuff started but needs some community input

  • Add your items here

Upcoming Presentations

  • Sept 2013
    • Sept 4 - Ben Langfeld, Ben Klang, Chris Rienzo (Rayo)
    • Sept 11 -
    • Sept 18 -
    • Sept 25 -
  • "Coming Soon"
    • Plivo update
    • Other surprises

Suggestions For Future Meetings & Future To Dos

  • Math: Sofia internals
  • Eliot Gable: mod_ha_cluster
    • What is mod_ha_cluster?
      • In short: N+X (N Masters + X slaves) HA replacement for "Pacemaker + Corosync managed FS"
    • For additional details, see mod_ha_cluster
  • Jeff Lenk: developing for FS in Windows environment
  • Echo
    • What/Why/Where
    • Troubleshooting
  • Contributing
    • How non-C programmers can help out - it's hard to write docs when you don't really know what's going on and referred to the code.

Presenters Needed For These Topics

  • Internals of modules like mod_callcenter and mod_conference
  • mod_shout/shoutcast/mod_vlc, esp with one-way conferences for scalability
  • Codec negotiation - early vs. late, why you need it, how to do it
  • T.38 - what it is, how to use it, etc.
  • Multi-tenancy (bounties welcome)
  • Steve Underwood - SpanDSP, T.38, etc. with FreeSWITCH
  • mod_fifo vs. mod_callcenter - why use one or the other? Strengths and weaknesses of each
  • embedding FS in other applications (libfreeswitch)
  • IPv6 - what it is, how to use, differences with IPv4, how to configure FS
  • WebRTC using Freeswicth

Need SIP Trunking or DIDs?

If you are looking for SIP Trunking, Origination or Termination please visit www.VoiceNetwork.ca

  • Unlimited Incoming DIDs from $3.95
  • Per Minute DIDs from $0.99 @ 0.01 per minute
  • Outbound Canadian Termination from $0.005 and USA from $ 0.0125
  • Free iNum's for each account

See Also

Return to main meeting page