FS weekly 2011 04 13

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Calling Instructions

Wednesday April 13th, 2011 at 1700 UTC (1200 CDT / 1pm EDT)
sip:888@conference.freeswitch.org or via the good old PSTN at +1-919-386-9900 or +44-3300-100-295
gtalk:conf+888@conference.freeswitch.org Or click on this link
Or call Skype the skype user "skypiax5", you'll be automatically connected (max 20 concurrent users).
Codecs: PCMU/PCMA, G.722, CELT, Speex, Skype, among others
Press 0 to mute/unmute your self. Press * to deaf/undeaf.

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


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

  • ClueCon - get registered NOW! http://www.cluecon.com
  • FSCB - started chapter 5; Ray and Darren working on ch4 still
  • ChangeLog updated through April 12!
  • Recent additions to FS
    • New sounds available (1.0.15) fixed dir-for_previous.wav recording
    • New chan var: rtp_disable_byteswap - see commit note for details
    • Add mod_timerfd for kernels >= 2.6.26 and libc >= 2.8 (commit note)
    • new sofia api: sofia_username_of (commit note)
    • New module: mod_event_zmq for you Zero MQ fans
    • New API: uuid_limit - API oriented limit (needs docs) (commit note)
    • New chan var transfer_on_fail - specify an auto-xfer destination for specified failure causes when doing a bridge (thanks jaybinks)
    • New API say_string (needs docs & examples) (commit note)
    • Merged file_string into dp_tools (needs docs & examples)
      • e.g. <action application="playback" data="file_string://<file1>!<file2>!<file3>"
      • if you make current then remember to remove formats/mod_file_string from your modules.conf!
    • Added ability to use native files with file_string (commit note)
    • Ability to specify language when using mod_say_en (commit note)
      • note the :lang you can do now: say <module_name>[:<lang>] <say_type> <say_method> [gender] <text>
      • Useful for those who have multiple sound sets for a single language
    • BEHAVIOR CHANGE - NATPMP/UPnP no longer "open by default" - you must specify nat mappings (See Mod_commands#nat_map) (commit note)
    • Allow from-domain to be set to "auto-aleg" for gateway config (commit note)
      • also have "auto-aleg-full" and auto-aleg-domain (commit note)
    • New "execute_on" function in core, so you can now do a stack of execute_on_answers (commit msg)

Featured Presentation

  • Sandro Gauci, author of SIPVicious

Upcoming Presentations

  • April 20th - J. Oquendo: PBX Honeypot and the VoIP abuse project
  • April 27th - Zac Wolfe of Safi Systems to talk about IVR builder stuff
  • Eliot Gable's HA stuff (no date yet)
  • Polycom (still working on a date)
  • All Upcoming Dates Open - Please let me (MSC) or NormT know if you have a topic or idea.

Questions For Developers

  • Add your questions here

Janitorial Items

  • Add your items here

Items Needing Documentation

  • Add your items here

Stuff started but needs some community input

  • Add your items here

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"
    • What is planned for mod_ha_cluster?
      • In short: manage interfaces, IPs, firewall rules, master/slave status; eventually, share registrations and call states without shared DB and do 'sofia recover' on failover
    • When will it be available?
      • In short: Master/slave, interface, IPs, and firewall management within the next couple of months (if I have the amount of time available that I expect to have)
    • For additional details, or if you would like to beta test, be on the call

Presenters Needed For These Topics

  • SIP 101 (beginning SIP, how does it work, how to look at different SIP packets)
  • mod_shout/shoutcast, 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)

FreeSWITCH High Availability Testing

Weekly FreeSWITCH High Availability testing during the conference call. At the end of the weekly conference call the lead developer of FreeSWITCH (anthm) will intentionally crash and restart the conference server. This tests the high Availability that is built into FreeSWITCH. Your call should continue once the server is restarted.

Depending on what type of NAT device will determine if your call is "Re-Invited" back to the conference.

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

See Also

Return to main meeting page