FS weekly 2010 07 14

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Calling Instructions

Wednesday July 14th at 1700 UTC (1200 CST / 1pm EST)
sip:888@conference.freeswitch.org or via the good old PSTN at +1-919-386-9900
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

  • FS book update - Release for July
    • Latest news from MSC, if any

Featured Presentation [ community call ]

  • Community talking

Upcoming Presentations

The dates for the following presentations are subject to change, so you need to check back.

  • July 21, 2010 - OPEN
  • July 28, 2010 - [intra]lanman - mod_xml_curl
  • August 4, 2010 - CLUECON BABY!
  • August 11, 2010 - DRK - mod_managed for billing and CDRs
  • August 18, 2010 - bkw - NAT and VoIP: let's figure it out

Questions For Developers

If you have questions for Tony, Mike, or Brian please add them here...

  • From mjgraves: what kinds of limitations have you found for developing for other platforms, like ARM-based systems? E.g. - limits with libraries being x86 only?

Items Needing Discussion

  • ClueCon Update
    • No more hotel, no more $599 discount, no more speaking slots. :)
  • Is anyone using T.38?
    • Please let us know how it's working for you
    • Which carrier(s)

Janitorial Items

  • Chan vars - let's crowdsource the descriptions of these vars
  • Need wiki volunteer: create_wiki_page.pl script
    • auto create new pages for chan vars, APIs, modules, etc.
  • Need volunteers to assist with the mail list moderation queue
    • CMF volunteered to assist during weekdays (GMT -7 aka Los Angeles)

Items Needing Documentation

We definitely need assistance with documenting stuff, especially new channels variables. A sample template for doing a channel variable can be found here: Variable_skeleton

Stuff started but needs some community input

  • Most important documentation task: T.38. Please see the email thread, especially bkw's comments.
    • Please add content to this page: T.38
  • Call Recording Channel variables section: Channel_Variables#Call_Recording_Related
    • New chan var: RECORD_HANGUP_ON_ERROR - when set to true the channel will hangup when record_session fails for whatever reason (default = false)
    • Please add pages for these chan vars...

Stuff yet to be documented

  • After the email thread about eavesdropping it looks like we need to document some mod_shout stuff:
    • how to uuid_record to an icecast server (including basic configs needed in icecast.xml)
    • how to use the telecast API (bonus: add syntax info to mod_shout.c)
  • New conference param: terminate-on-silence (commit msg)
  • Verbose events:
    • global param verbose-channel-events, fsctl verbose_events [on|off]
    • commit msg
  • New "disable hold" feature:
  • Channels to check if you have been auth'd:
  • New mod_conference chan var: conference_member_id (commit msg)
  • New mod_sofia chan var: sip_force_audio_fmtp (commit msg)
  • New mod_sofia chan var: sip_copy_multipart (commit msg)

User Tips & Tricks

If you have something you'd like to share with the community then by all means add it here and we will give you a few minutes on the conference call to discuss it!

  • Don't forget about the contrib folders! They are now in their own repository. See Git_Tips#Initial_Checkout for information on how to check it out.

Suggestions For Future Meetings & Future To Dos

Presenters Needed For These Topics

  • SIP 101 (beginning SIP, how does it work, how to look at different SIP packets)
  • NAT traversal debugging in FS environments (when to use autonat, how to diagnose your NAT routers, etc.)
  • mod_shout/shoutcast, esp with one-way conferences for scalability
  • Codec negotiation - early vs. late, why you need it, how to do it

Need SIP Trunking or DIDs?

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

  • Unlimited Incoming DIDs from $3.95
  • Per Minute DIDs from $0.99 @ 0.011 per minute

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