FS weekly 2011 02 16

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Contents

Calling Instructions

Wednesday February 16th at 1800 UTC (1200 CST / 1pm EST)
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.

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 dev dinner was great, thanks to all who donated!!
  • FS Cookbook: chapter 2 due two days ago, we're hacking away at it
  • Francois Delawarde has recorded Spanish (Castillian) and Portugese (Brazilian) sound prompts!
    • I still need to update the phrase files and do some mod_say patching but hopefully we'll have these new langs available soon
  • Yehavi B. has recorded Hebrew prompts and is getting the licensing squared away - hopefully in the next week or two we will have this language added as well.

Featured Presentation

  • No presentation, just getting us all caught up on the new stuff:
    • New SIP profile param: auto-jitter-buffer-msec (commit msg)
      • Documentation added here under RTP options
    • bind_meta_app now works on DTMFs A through D (commit msg)
    • New channel variable: temp_hold_music
    • Fixed routing behavior of inbound calls from gateways that only match gateway based on the gw request uri param we now honor the extension gateway param only if it is explicitly set, but will not route to the username param if extension is not set also, new special value for extension "auto" that should use the request uri unless it has gw+ and then it will use the to uri (commit msg])
    • More Sonus infection stuff (DTMF lengths) (commit msg)
    • Added mod_fsk (send/recv stuff inband prior to answer like analog caller ID)
    • Added origination_channel_name variable
    • Added/updated chan var last_transferred_conference
    • Added add send-presence-on-register (true|false|first-only) param to sofia and api command sofia global debug [presence|sla|none]
      • sofia param documented here
      • sofia global debug API documented here
    • add execute_on_originate var '<app> <arg>' to run in origination thread or '<app>::<arg>' to run async. also originating_leg_uuid variable to show the uuid of the originating leg on an outbound channel
      • execute_on_originate var documented here
      • originating_leg_uuid var stub here
    • Initial mod_snmp support!
    • Added record_restart_limit_on_dtmf channel var

Upcoming Presentations

  • J. Oquendo: PBX Honeypot and the VoIP abuse project (Has been delayed because of crazy weather in New England)
    • No updated date yet... stay tuned
  • All Upcoming Dates Open - Please let me (MSC) or NormT know if you have a topic or idea.

Questions For Developers

  • Ask your questions here...

Janitorial Items

  • Add your items here...

Items Needing Documentation

  • conference_enter_sound : committed 651acc62 (2010-12-21) -> Add a chan var conference_enter_sound to override conference enter-sound param on the profile

Stuff started but needs some community input

  • bind_digit_action has cool featureness, per anthm
anthm: is it doc'd how you can consume the events fired by bind_digit_action in either the inputcallback in embedded 
scripts or over ESL with divert_events enabled and if the name has exec: in it its a shortcut to make it exec apps ?

Stuff yet to be documented

  • New FIFO member_add param: taking_calls (commit msg)
  • New FIFO flag: no_media (commit msg)
  • New FIFO chan vars: fifo_bridged, fifo_manual_bridged (commit msg)
  • New FIFO param: outbound_ring_timeout (commit msg)
  • New FIFO param: default_lag (commit msg)
  • New sip profile param and chan var: manual-rtp-bugs and manual_rtp_bugs for our friends at Sonus (commit msg)
  • New sip profile param: presence-probe-on-register (commit msg)
  • New trick: use * in sofia_contact

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!

Suggestions For Future Meetings & Future To Dos

  • Math: Sofia internals

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

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