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Comment: Migration of unmigrated content due to installation of a new plugin


  • T.38 Gateway (Digium)
  • Performance of State Change Processing (Stefan Schmidt)
    • (work is already being done on this front)
  • SIP path support (Olle)
  • Group variables (Kobaz)
  • Pre-Dial (Kobaz) Finished, Committed
  • Hangup Handlers (Kobaz) Finished, Committed
  • Distributed extension state using SIP (Olle)
    • (resources in place, doing it, 1.4 done before Christmas, project pinana)
    • Review 2011: No changes known. Idle.
  • Manager event docs (Paul Belanger)
    • Review 2011: Code was created and working, but did not pass code review. Talked about it again a couple of months ago, and some work done.
      • About a day or two to get the framework completed
      • Then just need to insert the documentation into the code and then it could be completed.
  • Cross-platform documentation (Ben Klang)
    • (caveats for using Asterisk on operating system xyz; pull a PDF of the Wiki documentation into the source, don't forget to include basic installation information, and do it all in .txt - Ben)
    • Review 2011: Documentation updated for Solaris. Is on the wiki, and just needs to be put into a better location. Leif will help restructure part of the wiki to make the Linux and Solaris documentation (and other operating systems) a better format.
  • Fix libs to optionally init OpenSSL (Digium)
    • (or use existing tools; sort of a bug)
    • Review 2011: Code on reviewboard, need to confirm that the code solves the problem, confirmed it doesn't cause harm
      • Testing required on multiple platforms and libraries
  • Make ast_channel an opaque type (Digium)
    • Review 2011: Large project and has not been started. Should not be on P0.


  • Async DNS (TCP DNS and use a good resolver)
    • Review 2011: No change known.
  • Named ACLs (deluxepine)
    • Review 2011: Idle
  • SIP Security Events
    • Review 2011: Additional work was updated and put into Asterisk 10. Only reported manager authentication events prior to Asterisk 10.
  • Light weight means of holding NAT open in SIP (less complex than current qualify, Consider it done)
    • Review 2011: No change.
  • IPv6 for the restivus (IAX, Jabber/XMPP/Gtalk, Manager, etc.)
    • Review 2011: No change.
  • ConfBridge feature complete with MeetMe
    • Review 2011: Not entirely true, but very close.
  • Support sound file containers (matroska)
    • Review 2011: Suggestion to have (media) files used by Asterisk not just headerless files, so you could actually do things properly, like storing G729 that contains silent suppression information.
      • No change in Asterisk, but has been getting worked on for Asterisk SCF. Very complicated. Matroska is just a framework. Once stable for Asterisk SCF, we can consider building it for Asterisk as well.
  • RTMP client channel driver
    • Review 2011: No change.


  • Unique identifier for filtering log data to a call
    • (finishing what was already begun w/ Clod's project, CLI filtering; should take a look at what Stephan from's created)
    • Review 2011: Claude's patch was only for CLI filtering.
      • Discussion about in the logger.conf to change the configuration so that the 'core set verbose 5' (or debug, etc) that it does not affect all the configuration files when you just want to change the verbosity on the console. (
      • Configuration could be under a header, and then create your own filters for channels, and what verbosity,debug,etc. is output to a log file and console per file
      • Take Away: Need to have a discussion of what people would want and need (requirements gathering), and then we can investigate how difficult it would be to implement, and what the order of implementation.


Photos of attendees

We rock.




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