---
- branch: MAIN
  date: Sat Jan 28 19:11:36 UTC 2012
  files:
  - new: '1.21'
    old: '1.20'
    path: pkgsrc/comms/asterisk18/Makefile
    pathrev: pkgsrc/comms/asterisk18/Makefile@1.21
    type: modified
  - new: '1.12'
    old: '1.11'
    path: pkgsrc/comms/asterisk18/PLIST
    pathrev: pkgsrc/comms/asterisk18/PLIST@1.12
    type: modified
  - new: '1.18'
    old: '1.17'
    path: pkgsrc/comms/asterisk18/distinfo
    pathrev: pkgsrc/comms/asterisk18/distinfo@1.18
    type: modified
  id: 20120128T191136Z.3b87f0d07296fb2fbdf5520df66431e7561b5f5f
  log: |
    Update to Asterisk 1.8.9.0:

    The Asterisk Development Team is pleased to announce the release of
    Asterisk 1.8.9.0. This release is available for immediate download at
    http://downloads.asterisk.org/pub/telephony/asterisk/

    The release of Asterisk 1.8.9.0 resolves several issues reported by the
    community and would have not been possible without your participation.
    Thank you!

    The following is a sample of the issues resolved in this release:

    * AST-2012-001: prevent crash when an SDP offer
      is received with an encrypted video stream when support for video
      is disabled and res_srtp is loaded.  (closes issue ASTERISK-19202)
      Reported by: Catalin Sanda

    * Handle AST_CONTROL_UPDATE_RTP_PEER frames in local bridge loop.  Failing
      to handle AST_CONTROL_UPDATE_RTP_PEER frames in the local bridge loop
      causes the loop to exit prematurely. This causes a variety of negative side
      effects, depending on when the loop exits. This patch handles the frame by
      essentially swallowing the frame in the local loop, as the current channel
      drivers expect the RTP bridge to handle the frame, and, in the case of the
      local bridge loop, no additional action is necessary.
      (closes issue ASTERISK-19095) Reported by: Stefan Schmidt Tested
      by: Matt Jordan

    * Fix timing source dependency issues with MOH.  Prior to this patch,
      res_musiconhold existed at the same module priority level as the timing
      sources that it depends on.  This would cause a problem when music on
      hold was reloaded, as the timing source could be changed after
      res_musiconhold was processed. This patch adds a new module priority
      level, AST_MODPRI_TIMING, that the various timing modules are now loaded
      at. This now occurs before loading other resource modules, such
      that the timing source is guaranteed to be set prior to resolving
      the timing source dependencies.
      (closes issue ASTERISK-17474) Reporter: Luke H Tested by: Luke H,
      Vladimir Mikhelson, zzsurf, Wes Van Tlghem, elguero, Thomas Arimont
      Patched by elguero

    * Fix RTP reference leak.  If a blind transfer were initiated using a
      REFER without a prior reINVITE to place the call on hold, AND if Asterisk
      were sending RTCP reports, then there was a reference leak for the
      RTP instance of the transferrer.
      (closes issue ASTERISK-19192) Reported by: Tyuta Vitali

    * Fix blind transfers from failing if an 'h' extension
      is present.  This prevents the 'h' extension from being run on the
      transferee channel when it is transferred via a native transfer
      mechanism such as SIP REFER.  (closes issue ASTERISK-19173) Reported
      by: Ross Beer Tested by: Kristjan Vrban Patches: ASTERISK-19173 by
      Mark Michelson (license 5049)

    * Restore call progress code for analog ports. Extracting sig_analog
      from chan_dahdi lost call progress detection functionality.  Fix
      analog ports from considering a call answered immediately after
      dialing has completed if the callprogress option is enabled.
      (closes issue ASTERISK-18841)
      Reported by: Richard Miller Patched by Richard Miller

    * Fix regression that 'rtp/rtcp set debup ip' only works when a port
      was also specified.
      (closes issue ASTERISK-18693) Reported by: Davide Dal Reviewed by:
      Walter Doekes

    For a full list of changes in this release candidate, please see the ChangeLog:

    http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-1.8.9.0

    Thank you for your continued support of Asterisk!
  module: pkgsrc
  subject: 'CVS commit: pkgsrc/comms/asterisk18'
  unixtime: '1327777896'
  user: jnemeth