FS weekly 2010 11 17
From FreeSWITCH Wiki
Wednesday November 17th at 1700 UTC (1200 CST / 1pm EST)
sip:email@example.com or via the good old PSTN at +1-919-386-9900
gtalk:firstname.lastname@example.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
- Cookbook update
- FYI: mod_conference now uses the new switch_ivr_dmachine stuff (the stuff that powers bind_digit_action)
- New module: mod_mp4 (commit msg)
- If you know anything about this please add to wiki
- Vaccine for Sonus infection (commit msg)
- Use at your own risk, although if you're using Sonus then you're a risk taker anyway...
- New sound files have been recorded, will be rolled shortly
- New fun prompts
- Wake up call prompts
- DRK with more fun stuff
- All Open - Please let me (MSC) or NormT know if you have a topic or idea.
Questions For Developers
- Add your questions here
Items Needing Discussion
- Followup from last week: DRK has new info/files to share
- New FS codec behavior: commit note
- Add items here
Items Needing Documentation
- Add items here
Stuff started but needs some community input
- Add items here
Stuff yet to be documented
- New FIFO 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)
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!
LRN - Are you ready?
Are you getting the correct rate on your calls?
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)
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