[asterisk-dev] Asterisk 15.0.0-rc1 Now Available
Asterisk Development Team
asteriskteam at digium.com
Fri Sep 1 15:09:58 CDT 2017
The Asterisk Development Team would like to announce the first
release candidate of Asterisk 15.0.0.
This release candidate is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk
The release of Asterisk 15.0.0-rc1 resolves several issues reported by the
community and would have not been possible without your participation.
Thank you!
The following issues are resolved in this release candidate:
Bugs fixed in this release:
-----------------------------------
* ASTERISK-27152 - Sending a "tel" uri in a From or To header
in an unauthenticated message causes asterisk to crash
(Reported by Ross Beer)
* ASTERISK-27103 - core: ast_safe_system command injection
possible.
(Reported by Corey Farrell)
* ASTERISK-27013 - res_rtp_asterisk: Media can be hijacked even
with strict RTP enabled
(Reported by Joshua Colp)
* ASTERISK-27231 - res_rtp_asterisk: Allow remote SSRC to
change due to renegotiation
(Reported by Joshua Colp)
* ASTERISK-27222 - core: Don't queue up multiple video update
frames.
(Reported by Joshua Colp)
* ASTERISK-27212 - bridge_softmix: Quickly joining/leaving may
cause video stream to remain in SFU
(Reported by Richard
Mudgett)
* ASTERISK-27202 - If wget is not installed and "or" is not
available, external components (excluding pjsip) are not
installed
(Reported by Seán C. McCord)
* ASTERISK-27200 - manager: hook event is not being raised
(Reported by Kevin Harwell)
* ASTERISK-27179 - res_pjsip_session: Handling of 'msid' is
incorrect
(Reported by Kevin Harwell)
* ASTERISK-27182 - bridge: Crash when mapping streams
(Reported by Joshua Colp)
* ASTERISK-27189 - Make --with-pjproject-bundled the default
for Asterisk 15
(Reported by George Joseph)
* ASTERISK-27180 - channel: requester leaks joint_cap on
success.
(Reported by Corey Farrell)
* ASTERISK-27171 - Asterisk 15.0.0-Beta1 does not compile
(Reported by Ira Emus)
* ASTERISK-27119 - res_pjsip: parse/add msid attribute when
webrtc is enabled
(Reported by Kevin Harwell)
For a full list of changes in this release candidate, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.0.0-rc1
Thank you for your continued support of Asterisk!
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