[asterisk-users] Asterisk 1.8.9.0 Now Available

[Asterisk Development Team]

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!