FS weekly 2010 01 08
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Contents |
Calling Instructions
Friday January 8 at 1700 UTC (1100 CST)
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", then press "1" on the Skype dialpad (max 20 concurrent users).
Codecs: PCMU/PCMA, G.722, CELT, Speex, Skype, among others
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.
Recap of last week
- API documentation - jlenk doing conf and show and jaugenstine checking conf & ivr-related stuff
- Screen casting needed, a few options have been discussed
News, Notes, & Miscellaneous Fun Stuff
- Update on David Fansler's attempts to re-order the wiki pages.
Items Needing Documentation
- New FreeSWITCH Communicator soft phone! Need volunteers to test, report bugs, etc.
- Once jmesquita has the wiki page up please contribute to it. FSComm
- If anyone has Sangoma T1 cards and can set up boost & document that would be awesome
Janitorial Items
- FS API options/syntax check
- Need volunteers to pick a few API cmds and scour the source and make sure that the syntax/help/wiki are all up-to-date. This is critical for 1.0.5
- Update Mod_dptools page - numerous undocumented apps are listed
- FreeSWITCH on-line infrastructure:
- Please ask around for volunteers who can help with:
- Drupal - themes, social networking integration, maintenance
- MediaWiki - maintenance
- Need people to stay on IRC channel all day and have their clients ding them when people use ~take-a-number. We don't expect people to ignore their day jobs for hours at a time, however we could definitely use some help with answering questions.
Items Needing Discussion
- New FS stuff
- Aforementioned FSComm softphone
- Request for debs package maintenance update: issue is just getting package to install with modules in a consistent way with deb - longer term this will impact positively on takeup of the software. I can't offer much time nor have specific deb packaging skills - I will try and contribute to wiki however :-) Sorry can't join conf call. Martin Bartos
- I would like to have some folks who know SIP/RTP/NAT share with the group some trouble-shooting tips. Javar submitted this debug trace: [1] for a setup where an ATA is at a remote location behind NAT. Calls from the ATA to other extensions works fine; calls to the ATA are failing. Would like to have someone in the know review the trace and then share with the group some debugging tips, what to look for, etc. Eventually I would like to have a NAT busting wiki page that has a list of symptoms and examples.
