FS weekly 2010 10 13
Wednesday October 13th 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
- TLS for Windows - ready for testing per Jeff Link
- Math added new feature: fs_encode, encode by ptime (Math can explain more)
- Moc added new API for mod_callcenter: callcenter_config queue list
- Moises Silva, Sangoma Corp. - FreeTDM updates
- Presentation: File:Freetdm-update.pdf
- All Open - Please let me (MSC) or NormT know if you have a topic or idea.
Questions For Developers
If you have questions for Tony, Mike, or Brian please add them here...
Items Needing Discussion
- ClueCon videos update - Mishehu still out of commish :'(
- Add items here
Items Needing Documentation
Stuff started but needs some community input
- Add items here
Stuff yet to be documented
- New mod_sofia chan var: sip_force_audio_fmtp (commit msg)
- New mod_sofia chan var: sip_copy_multipart (commit msg)
- 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 chan var: last_bridge_to (commit msg)
- New stats (in CDRs) rtcp_packet_count and rtcp_octet_count (commit msg)
- Moc's new tod thing (commit msg)
- Moc's new date/time thing (commit msg)
- Moc's new 3-letter day thing (commit msg)
- New dp app: bridge_export (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!
Tricks to receive CDR's http://wiki.freeswitch.org/wiki/Xmlcdrd
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 crash the conference server. This tests the high Availability that is built into FreeSWITCH.
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