FS weekly 2011 03 16
Wednesday March 2nd, 2011 at 1800 UTC (1200 CST / 1pm EST)
sip:firstname.lastname@example.org or via the good old PSTN at +1-919-386-9900 or +44-3300-100-295
gtalk:email@example.com 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
- FSCB Update - chapters 2 and 3 are done (I think - Darren needs to check in)
- I've just started chapter 4 which is due next Monday (ugh)
- March 30 - Sandro Gauci, author of SIPVicious
- 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
- Add your questions here
- Add your items here
Items Needing Documentation
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 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 sip varibles: sip_jitter_buffer_during_bridge
- Chan var I didn't know about: verbose_sdp (set in vars.xml) (See anthm's post on mailing list)
- New API and app: recovery_refresh (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!
I've got FS [sort of] working with a Cisco CUBE (Cisco Unified Border Element) which id's as: Agent: Cisco-SIPGateway/IOS-12.x
With info on the fs wiki: http://wiki.freeswitch.org/wiki/Cisco_Call_Manager it can now register to fs but only with the use of NDLB-connectile-dysfunction
If anyone has any additional info on interop with a CUBE please add it to the wiki and I'll do the same.
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