FS weekly 2012 06 06
From FreeSWITCH Wiki
Contents |
Scheduled time
Wednesday June 6th, 2012 at 1700 UTC (1200 CDT / 1300 EDT)
Calling Instructions
- SIP
- sip:888@conference.freeswitch.org
- Codecs supported: PCMU/PCMA, G.722, CELT, Speex, among many others
- PSTN
| Country | Number | Contributor |
|---|---|---|
| USA | +1-919-386-9900 | |
| Spain | +34-91-290-12-71 | Thanks to SIPtize |
| UK | +44-330-320-0105 | Thanks to ziron.net |
| UK | +44-1904-201-313 | Thanks to ukddi.com (Routed by Steven Ayre) |
| UK | +44-203-298-5931 | Thanks to ukddi.com (Routed by Avi Marcus) |
| Germany | +49-228-9293-9009 | Thanks to Yiftach at ChooChee |
| Australia | +61-7-3188-7519 | Thanks to Jay Binks - NetSIP.com.au |
| Israel | +972-2-372-0394 | Thanks to Avi Marcus - BestFone.com |
| Canada | +1-438-800-0531 | Thanks to NG Communications |
| France | +33-975-181-606 | Thanks to NG Communications |
| Netherlands | +31-858-880-387 | Thanks to NG Communications |
| South Africa | +27-87-8204656 | Thanks to Othos Telecom |
- Google Talk
- conf+888@conference.freeswitch.org
- Skype
- skypiax5 (max 20 concurrent users)
- Use Call->Show Dialpad to unmute
- Flash (in-browser VoIP RTMP client)
Note: When you join you will initially be muted and need to press 0 if you wish to speak (this reduces background noise on the conference).
At some times muting may be moderated (eg presentations). Pressing 0 will put you in a queue and you will be unmuted when it's your turn to speak.
Be sure to be logged into #freeswitch on irc.freenode.net! There are frequently related comments during the conference.
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
- Darren's SIP 101 re-scheduled for Juen 20th
- Don't forget to sign up for ClueCon
- Has anyone tried FSClient on Windows yet?
- Update on "Bridge Book" question from last week
Featured Presentation
- First Look: mod_httapi
- MSC will be giving a beginner's look at HT TAPI. If you have a FreeSWITCH server then be sure to compile and load mod_httapi so that you tinker.
- Screen access:
- ssh guest@north01.racserv.com -p222
- putty -P 222 guest@north01.racserv.com
- Password: "cluecon"
Questions For Developers
- Add your questions here
Janitorial Items
- Add your questions here
Items Needing Documentation
- Add your items here
Items Needing Discussion
- Add your items here
Stuff started but needs some community input
- Add your items here
Upcoming Presentations
- May 2013
- May 1 - Omar from OrecX (call recording)
- May 8 - Dan Bogos from CGRateS project
- May 15 - Community scrum
- May 22 - Open
- May 29 - VoIPMonitor.org
- June 2013
- June 5 - Alexandr Dubovikov (HOMER)
- June 12 -
- June 19 -
- June 26 -
- "Coming Soon"
- Plivo update
- Other surprises
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"
- For additional details, see mod_ha_cluster
- What is mod_ha_cluster?
- Jeff Lenk: developing for FS in Windows environment
- Echo
- What/Why/Where
- Troubleshooting
- Contributing
- How non-C programmers can help out - it's hard to write docs when you don't really know what's going on and referred to the code.
Presenters Needed For These Topics
- Internals of modules like mod_callcenter and mod_conference
- mod_shout/shoutcast/mod_vlc, 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)
- IPv6 - what it is, how to use, differences with IPv4, how to configure FS
- WebRTC using Freeswicth
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
- Free iNum's for each account
