[asterisk-dev] Asterisk 1.8.9.0-rc3 Now Available

Matthew Jordan mjordan at digium.com
Tue Jan 24 14:48:33 CST 2012


Apologies for the slightly malformed release announcement below.  I'll assume
that no one wants to be mail bombed with another one - if I'm wrong, I'll
resend it.

Sorry everyone!

Matthew Jordan
Digium, Inc. | Software Developer
445 Jan Davis Drive NW - Huntsville, AL 35806 - USA
Check us out at: http://digium.com & http://asterisk.org

----- Original Message -----
> From: "Asterisk Development Team" <asteriskteam at digium.com>
> To: asterisk-dev at lists.digium.com
> Subject: [asterisk-dev] Asterisk 1.8.9.0-rc3 Now Available
> 
> Asterisk 1.8.9.0. This release candidate is available for immediate
> download at
> http://downloads.asterisk.org/pub/telephony/asterisk/
> 
> The release of Asterisk 1.8.9.0-rc3 resolves several issues reported
> by the
> community and would have not been possible without your
> participation.
> Thank you!
> 
> The following are some of the issues resolved in this release
> candidate:
> 
> * 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
> 
> * Fix RTP reference leak.
> 
>   If a blind transfer were initiated using a REFER with 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 transferer.
>   (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: Kritjan Vrban
> 
> 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-rc3
> 
> Thank you for your continued support of Asterisk!lain;
> charset=ISO-8859-1
> Content-Transfer_encoding: 8bit
> 
> The Asterisk Development Team has announced the third release
> candidate of
> Asterisk 1.8.9.0. This release candidate is available for immediate
> download at
> http://downloads.asterisk.org/pub/telephony/asterisk/
> 
> The release of Asterisk 1.8.9.0-rc3 resolves several issues reported
> by the
> community and would have not been possible without your
> participation.
> Thank you!
> 
> The following are some of the issues resolved in this release
> candidate:
> 
> * 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
> 
> * Fix RTP reference leak.
> 
>   If a blind transfer were initiated using a REFER with 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 transferer.
>   (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: Kritjan Vrban
> 
> 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-rc3
> 
> Thank you for your continued support of Asterisk!
> 
> 
> --
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> 
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> To UNSUBSCRIBE or update options visit:
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