[AsteriskBrasil] Fwd: [asterisk-dev] Asterisk 10.1.0 Now Available

[Denis Galvão - Gmail]

Denis at mobile.

Begin forwarded message:

> From: Asterisk Development Team <asteriskteam em digium.com>
> Date: 27 de janeiro de 2012 17:10:00 BRST
> To: asterisk-dev em lists.digium.com
> Subject: [asterisk-dev] Asterisk 10.1.0 Now Available
> Reply-To: Asterisk Developers Mailing List <asterisk-dev em lists.digium.com>
> 
> The Asterisk Development Team is pleased to announce the release of
> Asterisk 10.1.0. This release is available for immediate download at
> http://downloads.asterisk.org/pub/telephony/asterisk/
> 
> The release of Asterisk 10.1.0 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
> 
> * Allow playback of formats that don't support seeking.  ast_streamfile
>  previously did unconditional seeking on files that broke playback of
>  formats that don't support that functionality.  This patch avoids the
>  seek that was causing the problem.  
>  (closes issue ASTERISK-18994) Patched by: Timo Teras
> 
> * Add pjmedia probation concepts to res_rtp_asterisk's learning mode.  In
>  order to better handle RTP sources with strictrtp enabled (which is the
>  default setting in 10) using the learning mode to figure out new sources
>  when they change is handled by checking for a number of consecutive (by
>  sequence number) packets received to an rtp struct based on a new
>  configurable value called 'probation'.  Also, during learning mode instead
>  of liberally accepting all packets received, we now reject packets until a
>  clear source has been determined.
> 
> * 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-10.1.0
> 
> Thank you for your continued support of Asterisk!
> 
> 
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