The Asterisk Development Team would like to announce the release of Asterisk 13.24.0-rc1.
This release is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk
The release of Asterisk 13.24.0-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:
Security bugs fixed in this release:
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res_http_websocket: Crash when reading HTTP Upgrade requests (Reported by Sean Bright) |
New Features made in this release:
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add flag to allow CALLERID(num) to be placed in Contact header in chan_pjsip (Reported by Torrey Searle) |
Bugs fixed in this release:
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app_queue: Revert broken queue channel reference patch (Reported by lvl) |
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app_voicemail: MWI fails with mailboxes=##@device instead of mailboxes=##@default (Reported by Ronald Raikes) |
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Asterisk crashes when the res_pjsip_* modules unload (Reported by sungtae kim) |
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SIGABRT caused by stack corruption in hashkeys_read when no matching keys present (Reported by Michael Walton) |
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repeated segmentation faults (Reported by Eyal Hasson) |
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stasis: Filter messages at publishing to reduce work done (Reported by Joshua C. Colp) |
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Incorrect Behavior for rewrite_contact when Re-Invite omits routset (Reported by Torrey Searle) |
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Some conditions prevent running of el_end, break the terminal. (Reported by Corey Farrell) |
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need to reset DTMF last sequence number and timestamp on voice packet with marker bit (Reported by Alexei Gradinari) |
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rtp: Incorrect Packetization (Reported by Robert Cripps) |
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pbx_config: Only the first [globals] section is processed. (Reported by Corey Farrell) |
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Formatting error in documentation (Reported by Scott Griepentrog) |
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chan_sip: Asterisk 12+ chan_sip doesn’t report AST_CEL_PICKUP in handle_invite_replaces (Reported by Luit van Drongelen) |
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res_pjsip_notify: improve realtime performance on CLI completion on the endpoint (Reported by Alexei Gradinari) |
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Caller ID cannot be changed on Attended Transfer before dialing out (Reported by Alexei Gradinari) |
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function ast_sendtext() create RTP realtime packets with a trailing null byte in the payload (Reported by Emmanuel BUU) |
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bridging: Asterisk crashes when receiving an empty realtime text frame (Reported by Emmanuel BUU) |
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app_queue: QueueMemberStatus Event flooding AMI (Reported by Andrej) |
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res_pjsip: improve realtime performance on CLI ‘pjsip show contacts’ (Reported by Alexei Gradinari) |
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stasis: Playing MOH to bridge with ARI does not work (Reported by Cameron) |
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app_queue: Queue member considered inuse after immediately hanging up during dialing. (Reported by Cao Minh Hiep) |
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testsuite: Sniffer assumes pjmedia will use ports below 10000 (Reported by Joshua C. Colp) |
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res_odbc: missing SQL error diagnostic (Reported by Alexei Gradinari) |
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res_pjsip_mwi: Memory leak on reload (Reported by Sergej Kasumovic) |
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PJSIP: Update bundled PJPROJECT to version 2.8 (Reported by Joshua C. Colp) |
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chan_sip: SipNotify via AMI behaves differently to CLI (Reported by Peter Katzmann) |
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res_pjproject build failure (Reported by Jaco Kroon) |
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res_musiconhold : music on hold will not start if previous hold just reached end of file (Reported by Frederic LE FOLL) |
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Realtime queuemembers are not updated during retry phase (Reported by lvl) |
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alembic: PJSIP “mwi_subscribe_replaces_unsolicited” field is integer not boolean (Reported by Joshua C. Colp) |
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res_pjsip_transport_websocket: Properly set ‘received’ for IPv6 (Reported by Sean Bright) |
Improvements made in this release:
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New function PJSIP_PARSE_URI to parse an URI and return a specified part of the URI (Reported by Alexei Gradinari) |
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Allow the sip_to_pjsip script to be used in a pipe (Reported by Pascal Cadotte Michaud) |
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Remove stale nonoptreq references (Reported by Walter Doekes) |
For a full list of changes in this release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.24.0-rc1
Thank you for your continued support of Asterisk!