The Asterisk Development Team would like to announce the first release candidate of Asterisk 15.5.0.
This release candidate is available for immediate download at
http://downloads.asterisk.org/pub/telephony/asterisk
The release of Asterisk 15.5.0-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:
Security bugs fixed in this release:
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Username bruteforce is possible when using ACL with PJSIP (Reported by John) |
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iostreams: Potential DoS when client connection closed prematurely (Reported by Sean Bright) |
Bugs fixed in this release:
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res_pjsip_pubsub: apparent crash on shutdown (Reported by Kevin Harwell) |
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app_confbridge: Conference bridge and announcer channels are not removed if conference is ended as soon as it starts (Reported by Robert Mordec) |
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AMI: Action SendText needs to use the correct thread. (Reported by Richard Mudgett) |
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res_pjsip_messaging doesn’t accept application/* content-types. (Reported by George Joseph) |
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cdr: Deadlock with submit_scheduled_batch and submit_unscheduled_batch (Reported by Denis Lebedev) |
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res_pjsip_session doesn’t update media when a 200 comes in with a different port than a 183 (Reported by George Joseph) |
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pbx_dundi: Asterisk crashes when unloading module pbx_dundi.so with dundi peers (Reported by Kirsty Tyerman) |
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uuid: Enable UUID in Solaris 11. (Reported by Alexander Traud) |
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channels: CHECK_BLOCKING is ineffective (Reported by Corey Farrell) |
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BuildSystem: Enable ./configure in Solaris 11. (Reported by Alexander Traud) |
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bootstrap.sh: find -maxdepth is not POSIX compatible. (Reported by Alexander Traud) |
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menuselect: GCC 8: restrict-qualified parameter passed and aliased. (Reported by Alexander Traud) |
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tests/test_utils: Repair ./configure –with-ssl=PATH. (Reported by Alexander Traud) |
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chan_iax2: Stops listening for traffic (Reported by Kirsty Tyerman) |
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crypto.h: Repair ./configure –with-ssl=PATH. (Reported by Alexander Traud) |
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res_srtp: Repair ./configure –with-ssl=PATH. (Reported by Alexander Traud) |
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SQL fetch error on query which return 0 columns (Reported by Alexei Gradinari) |
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chan_pjsip isn’t updating hangupcause on 4XX responses (Reported by George Joseph) |
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ooh323c: GCC 8: output truncated before terminating nul. (Reported by Alexander Traud) |
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res_pjsip: Modified qualify_frequency doesn’t effect until pjsip reload (Reported by Alexei Gradinari) |
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res_fax: Deadlock when using Local channels and fax gateway (Reported by David Brillert) |
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rtp: DTMF Breaks With telephony-event/16000 (Reported by Dominic) |
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Manager events for MeetMe have incorrectly documented key name ‘Usernum’ – should be ‘User’ (Reported by Francois Blackburn) |
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tcptls.h: Repair ./configure –with-ssl=PATH. (Reported by Alexander Traud) |
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tcptls: Allow OpenSSL configured with no-dh. (Reported by Alexander Traud) |
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tcptls: Allow OpenSSL 1.1.x configured with enable-ssl3-method no-deprecated. (Reported by Alexander Traud) |
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Codec-Change Re-INVITE during DTMF can cause marker bit error (Reported by Torrey Searle) |
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res_rtp_asterisk: Add support for abs-send-time RTP extension (Reported by Joshua Colp) |
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config/ast_destroy_realtime_fields: successful DELETE is treated as failed (Reported by Alexei Gradinari) |
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: tcptls: Repair ./configure –with-ssl=PATH. (Reported by Alexander Traud) |
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Asterisk ODBC Voicemail Prompt storage fails with recent MariaDB version. (Reported by Nic Colledge) |
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Incorrect error reported when leaving/retrieving a ODBC voicemail (Reported by Nic Colledge) |
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chan_mobile: presents incorrect inbound Caller-ID names (Reported by Brian) |
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res_pjsip_endpoint_identifier_ip: Unregister the module for headers. (Reported by Alexander Traud) |
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res_pjsip: Register pjsip_transport_management not externally but internally. (Reported by Alexander Traud) |
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cli: “manager show settings” mislabels HTTP timeout as being minutes. (Reported by Corey Farrell) |
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Fix issues exposed by GCC 8 (Reported by George Joseph) |
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rtp_engine: Allow Media Formats with add_static_payload(-1) on egress again. (Reported by Alexander Traud) |
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sip_to_pjsip: Enable python3 compatibility. (Reported by Alexander Traud) |
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digest over for manager (ami) over http fails on too long uris (Reported by Jaco Kroon) |
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Macro allows an infinite loop of dialplan inclusion resulting in a crash (Reported by Tzafrir Cohen) |
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Asterisk got stuck while enabling “ari set debug all on” (Reported by shaurya jain) |
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chan_sip: one way / no audio with srtp (Reported by Florian Kaiser) |
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One way audio when calling from Asterisk(sip trunk) to another number where both are connected to a SBC using TLS+SRTP (Reported by Artur Pires) |
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pjsip_options: rework to make more efficient (Reported by Kevin Harwell) |
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translate: interpolated frames are not passed through (Reported by Kevin Harwell) |
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When the ooh323 debug is on there is no ringing signal to incoming calls via H323 trunk. (Reported by Dimos) |
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No “alert” or “progress” in chan_ooh323 if debug is enabled only on the module (Reported by Marco Giordani) |
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BuildSystem: Enable IMAP storage on FreeBSD and DragonFly BSD. (Reported by Alexander Traud) |
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bridge_softmix / app_confbridge: Add support for combining REMB reports (Reported by Joshua Colp) |
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app_confbridge: “core show profile bridge” does not output “sfu” when video_mode is sfu (Reported by Carlos Chavez) |
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chan_vpb: Avoid GNU old-style field designator extension. (Reported by Alexander Traud) |
Improvements made in this release:
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BuildSystem: Enable autotools in Solaris 11. (Reported by Alexander Traud) |
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Ten seconds of silence after mp3 playback (Reported by Sam Wierema) |
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res_rtp_asterisk: Allow OpenSSL configured with no-deprecated. (Reported by Alexander Traud) |
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res_crypto: Allow OpenSSL configured with no-deprecated. (Reported by Alexander Traud) |
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app_confbridge: Add talking indicator for ConfBridgeList AMI response (Reported by William McCall) |
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documentation: Error on wiki description of Asterisk 13 “MeetmeMute” event (Reported by Alessandro Polidori) |
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ast_coredumper: Fix OUTPUT directory (Reported by Ted G) |
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libasteriskssl: Allow OpenSSL 1.0.2 configured with no-deprecated. (Reported by Alexander Traud) |
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res_hep: Allow create_address to resolve a provided hostname (Reported by Sebastian Gutierrez) |
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Add DragonFly BSD. (Reported by Alexander Traud) |
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cppcheck identifies redundant “if” (Reported by Ilya Shipitsin) |
For a full list of changes in this release candidate, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-15.5.0-rc1
Thank you for your continued support of Asterisk!