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:
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Sending a “tel” uri in a From or To header in an unauthenticated message causes asterisk to crash (Reported by Ross Beer) |
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core: ast_safe_system command injection possible. (Reported by Corey Farrell) |
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res_rtp_asterisk: Media can be hijacked even with strict RTP enabled (Reported by Joshua Colp) |
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res_rtp_asterisk: Allow remote SSRC to change due to renegotiation (Reported by Joshua Colp) |
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core: Don’t queue up multiple video update frames. (Reported by Joshua Colp) |
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bridge_softmix: Quickly joining/leaving may cause video stream to remain in SFU (Reported by Richard Mudgett) |
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If wget is not installed and “or” is not available, external components (excluding pjsip) are not installed (Reported by Seán C. McCord) |
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manager: hook event is not being raised (Reported by Kevin Harwell) |
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res_pjsip_session: Handling of ‘msid’ is incorrect (Reported by Kevin Harwell) |
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bridge: Crash when mapping streams (Reported by Joshua Colp) |
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Make –with-pjproject-bundled the default for Asterisk 15 (Reported by George Joseph) |
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channel: requester leaks joint_cap on success. (Reported by Corey Farrell) |
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Asterisk 15.0.0-Beta1 does not compile (Reported by Ira Emus) |
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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!