The release of Asterisk 13.22.0 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:
-----------------------------------
| Username bruteforce is possible when using ACL with PJSIP | (Reported by John) |
Bugs fixed in this release:
-----------------------------------
| res_pjsip_pubsub: apparent crash on shutdown | (Reported by Kevin Harwell) | |
| app_confbridge: Conference bridge and announcer channels are not removed if conference is ended as soon as it starts | (Reported by Robert Mordec) | |
| cdr: Deadlock with submit_scheduled_batch and submit_unscheduled_batch | (Reported by Denis Lebedev) | |
| pbx_dundi: Asterisk crashes when unloading module pbx_dundi.so with dundi peers | (Reported by Kirsty Tyerman) | |
| AMI: Action SendText needs to use the correct thread. | (Reported by Richard Mudgett) | |
| res_pjsip_messaging doesn't accept application/* content-types. | (Reported by George Joseph) | |
| res_pjsip_session doesn't update media when a 200 comes in with a different port than a 183 | (Reported by George Joseph) | |
| [patch] uuid: Enable UUID in Solaris 11. | (Reported by Alexander Traud) | |
| channels: CHECK_BLOCKING is ineffective | (Reported by Corey Farrell) | |
| [patch] BuildSystem: Enable ./configure in Solaris 11. | (Reported by Alexander Traud) | |
| [patch] bootstrap.sh: find -maxdepth is not POSIX compatible. | (Reported by Alexander Traud) | |
| menuselect: GCC 8: restrict-qualified parameter passed and aliased. | (Reported by Alexander Traud) | |
| [patch] tests/test_utils: Repair ./configure --with-ssl=PATH. | (Reported by Alexander Traud) | |
| chan_iax2: Stops listening for traffic | (Reported by Kirsty Tyerman) | |
| [patch] crypto.h: Repair ./configure --with-ssl=PATH. | (Reported by Alexander Traud) | |
| [patch] res_srtp: Repair ./configure --with-ssl=PATH. | (Reported by Alexander Traud) | |
| SQL fetch error on query which return 0 columns | (Reported by Alexei Gradinari) | |
| chan_pjsip isn't updating hangupcause on 4XX responses | (Reported by George Joseph) | |
| [patch] ooh323c: GCC 8: output truncated before terminating nul. | (Reported by Alexander Traud) | |
| res_fax: Deadlock when using Local channels and fax gateway | (Reported by David Brillert) | |
| Manager events for MeetMe have incorrectly documented key name 'Usernum' - should be 'User' | (Reported by Francois Blackburn) | |
| [patch] tcptls.h: Repair ./configure --with-ssl=PATH. | (Reported by Alexander Traud) | |
| res_pjsip: Modified qualify_frequency doesn't effect until pjsip reload | (Reported by Alexei Gradinari) | |
| [patch] tcptls: Allow OpenSSL configured with no-dh. | (Reported by Alexander Traud) | |
| [patch] tcptls: Allow OpenSSL 1.1.x configured with enable-ssl3-method no-deprecated. | (Reported by Alexander Traud) | |
| Codec-Change Re-INVITE during DTMF can cause marker bit error | (Reported by Torrey Searle) | |
| config/ast_destroy_realtime_fields: successful DELETE is treated as failed | (Reported by Alexei Gradinari) | |
| [patch]: tcptls: Repair ./configure --with-ssl=PATH. | (Reported by Alexander Traud) | |
| Incorrect error reported when leaving/retrieving a ODBC voicemail | (Reported by Nic Colledge) | |
| chan_mobile: presents incorrect inbound Caller-ID names | (Reported by Brian) | |
| [patch] res_pjsip_endpoint_identifier_ip: Unregister the module for headers. | (Reported by Alexander Traud) | |
| [patch] res_pjsip: Register pjsip_transport_management not externally but internally. | (Reported by Alexander Traud) | |
| Asterisk ODBC Voicemail Prompt storage fails with recent MariaDB version. | (Reported by Nic Colledge) | |
| cli: "manager show settings" mislabels HTTP timeout as being minutes. | (Reported by Corey Farrell) | |
| Fix issues exposed by GCC 8 | (Reported by George Joseph) | |
| [patch] sip_to_pjsip: Enable python3 compatibility. | (Reported by Alexander Traud) | |
| digest over for manager (ami) over http fails on too long uris | (Reported by Jaco Kroon) | |
| Macro allows an infinite loop of dialplan inclusion resulting in a crash | (Reported by Tzafrir Cohen) | |
| Asterisk got stuck while enabling "ari set debug all on" | (Reported by shaurya jain) | |
| pjsip_options: rework to make more efficient | (Reported by Kevin Harwell) | |
| translate: interpolated frames are not passed through | (Reported by Kevin Harwell) | |
| When the ooh323 debug is on there is no ringing signal to incoming calls via H323 trunk. | (Reported by Dimos) | |
| No "alert" or "progress" in chan_ooh323 if debug is enabled only on the module | (Reported by Marco Giordani) | |
| [patch] BuildSystem: Enable IMAP storage on FreeBSD and DragonFly BSD. | (Reported by Alexander Traud) | |
| [patch] chan_vpb: Avoid GNU old-style field designator extension. | (Reported by Alexander Traud) |
Improvements made in this release:
-----------------------------------
| [patch] BuildSystem: Enable autotools in Solaris 11. | (Reported by Alexander Traud) | |
| Ten seconds of silence after mp3 playback | (Reported by Sam Wierema) | |
| [patch] res_rtp_asterisk: Allow OpenSSL configured with no-deprecated. | (Reported by Alexander Traud) | |
| [patch] res_crypto: Allow OpenSSL configured with no-deprecated. | (Reported by Alexander Traud) | |
| app_confbridge: Add talking indicator for ConfBridgeList AMI response | (Reported by William McCall) | |
| documentation: Error on wiki description of Asterisk 13 "MeetmeMute" event | (Reported by Alessandro Polidori) | |
| ast_coredumper: Fix OUTPUT directory | (Reported by Ted G) | |
| [patch] libasteriskssl: Allow OpenSSL 1.0.2 configured with no-deprecated. | (Reported by Alexander Traud) | |
| res_hep: Allow create_address to resolve a provided hostname | (Reported by Sebastian Gutierrez) | |
| [patch] Add DragonFly BSD. | (Reported by Alexander Traud) | |
| cppcheck identifies redundant "if" | (Reported by Ilya Shipitsin) |
For a full list of changes in this release, please see the ChangeLog:
http://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-13.22.0
Thank you for your continued support of Asterisk!
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