The release of Asterisk 18.10.0 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:
New Features made in this release:
-----------------------------------
| cdr: allow disabling CDR by default | (Reported by N A) | |
| ami: Add AMI event for Wink | (Reported by N A) | |
| app_sf: Add full tech-agnostic SF support | (Reported by N A) | |
| app_sendtext: Add ReceiveText application | (Reported by N A) | |
| func_json: Add JSON parsing function | (Reported by N A) |
Bugs fixed in this release:
-----------------------------------
| res_pjsip_outbound_authenticator_digest: ABRT attempting to clean up auth_sess | (Reported by George Joseph) | |
| res_tonedetect: fix logic errors in code | (Reported by N A) | |
| func_frame_drop: fix buffer usage typo | (Reported by N A) | |
| rtp sequence number can skip after DTMF under certain bridges | (Reported by Torrey Searle) | |
| gethostbyname_r is misdetected on NetBSD and causes a build failure | (Reported by MichaÅ? Górny) | |
| Segfault if sorcery object_lifetime_maximum and qualify_frequency the same value | (Reported by Alexei Gradinari) | |
| rdtsc is not enabled (stubbed out) on NetBSD | (Reported by MichaÅ? Górny) | |
| make_version uses GNU-ism that break git-svn-id parsing on NetBSD | (Reported by MichaÅ? Górny) | |
| ast_get_tid() not implemented for NetBSD | (Reported by MichaÅ? Górny) | |
| Build failure on NetBSD due to hmac function collision | (Reported by MichaÅ? Górny) | |
| res_rtp_asterisk: Invalid comparison creates unreachable code | (Reported by N A) | |
| configure fails if libsrtp dev files are not installed | (Reported by Sean Bright) | |
| res_pjsip_session doesn't support multipart message bodies | (Reported by George Joseph) | |
| Regression: Using external pjproject not working after "hack" commit | (Reported by George Joseph) | |
| VoiceMailMain() fails when encountering non-numeric CALLERID(num) | (Reported by Mark Murawski) | |
| pbx_variables: ASTSBINDIR is missing | (Reported by N A) | |
| It's hard to make changes to bundled pjproject | (Reported by George Joseph) | |
| SAY.CONF wrong logic when converting 24hour time to say 12 hour am/pm | (Reported by Vincent Dubois) | |
| PJSIP processing token with % incorrectly | (Reported by Dan Cropp) | |
| Support for Nordic language syntax in Queues | (Reported by Mark Petersen) | |
| app_queue: QueueSummary and QueueStatus events don't exist in documentation | (Reported by Luke Escude) | |
| tcptls.c: TCP client connect fails due to interrupt | (Reported by Kevin Harwell) | |
| app_queue: extension state incorrect | (Reported by Steve Davies) | |
| SAY_DTMF_INTERRUPT channel variable is not honored | (Reported by Sean Bright) | |
| The ast_rtp_codecs_payloads functions don't preserve order | (Reported by George Joseph) | |
| res_pjsip_sdp_rtp: Codec preference order of remote is not correct on unhold | (Reported by Ross Beer) | |
| Deadlock in bridge_channel_internal_join() on local channels. | (Reported by Krzysztof Trempala) | |
| test_timezone_watch breaks during DST to ST transition | (Reported by Josh Soref) | |
| bundled_pjproject: sip_inv is missing multipart support in some cases | (Reported by George Joseph) | |
| ast_coredumper does not delete results when requested and a specific output dir is set | (Reported by Frederic Van Espen) | |
| pbx_variables: cp4 variables is used uninitialized | (Reported by N A) | |
| pbx_variables: MSet truncates sets after 24 variables | (Reported by N A) | |
| chan_sip: ${CHANNEL(ruri)} in Dial/Queue b(test,s,1) cause a coredump | (Reported by Mark Petersen) | |
| xmldoc: Dump invalid to XML DTD: XSLT | (Reported by Alexander Traud) | |
| xmldoc: Dump invalid to XML DTD: ACO Matchfield | (Reported by Alexander Traud) | |
| documentation: Doxygen site is no longer being updated | (Reported by Joshua C. Colp) | |
| [patch] Update Doxygen Configuration for make progdocs | (Reported by Andrew Latham) | |
| res_pjsip_sdp_rtp: Warns on every offered crypto suite | (Reported by Alexander Traud) | |
| Infinite loop when out of ports and rtpstart value is odd | (Reported by Thomas Guebels) | |
| chan_pjsip: Wrong or missing Q.850 reason in CANCEL | (Reported by Simone Lazzaris) | |
| res: Fix for Doxygen | (Reported by Alexander Traud) | |
| main: Fix for Doxygen | (Reported by Alexander Traud) |
Improvements made in this release:
-----------------------------------
| Enable pickup on channel after having received 183 Progress | (Reported by Mark Petersen) | |
| res_pjsip_sdp_rtp: Keepalive not supported for video streams | (Reported by Luke Escude) | |
| Queue don't play "thank-you" when here is no hold time announcements | (Reported by Mark Petersen) | |
| frame.h: fix CNG documentation typo | (Reported by N A) | |
| documentation: Document special system and channel variables | (Reported by N A) | |
| utils.c: Remove all usages of ast_gethostbyname() | (Reported by Sean Bright) | |
| dsp: Define magic number as macro | (Reported by N A) | |
| cli: add module refresh command | (Reported by N A) | |
| app_mp3: Throw warning if attempting to play a nonexistent stream | (Reported by N A) | |
| Documentation is missing for a few AMI Events - Including CDR and events triggered after the QueueStatus action | (Reported by Dafi Ni) | |
| DIALEDPEERNUMBER not set on destination channel for Queue calls | (Reported by Mark Petersen) | |
| app.c: Throw warnings for nonexistent options | (Reported by N A) | |
| Support for Danish language syntax in VM | (Reported by Mark Petersen) | |
| strings: Fix misusage in comment examples | (Reported by N A) | |
| configs: Minor updates to sample configs | (Reported by N A) | |
| pbx: Add public API for more elegant variable substitution with extensions | (Reported by N A) | |
| Incompatibility with newer spandsp releases (3.0.0+) | (Reported by Dustin Marquess) |
For a full list of changes in this release, please see the ChangeLog:
https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-18.10.0
Thank you for your continued support of Asterisk!
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