[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!
>
>
> --
> _____________________________________________________________________
> -- Bandwidth and Colocation Provided by http://www.api-digital.com --
>
> asterisk-dev mailing list
> To UNSUBSCRIBE or update options visit:
> http://lists.digium.com/mailman/listinfo/asterisk-dev
-------------- next part --------------
A non-text attachment was scrubbed...
Name: not available
Type: text/pthe
Size: 2931 bytes
Desc: not available
URL: <http://lists.digium.com/pipermail/asterisk-dev/attachments/20120124/b45bda4d/attachment.bin>
More information about the asterisk-dev
mailing list