The release of Asterisk 16.26.0 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:
-----------------------------------
| res_stir_shaken: Blind SSRF vulnerabilities | (Reported by Clint Ruoho) | |
| ${SQL_ESC()} not correctly escaping a terminating \ | (Reported by Leandro Dardini) | |
| res_stir_shaken: Resource exhaustion with large files | (Reported by Benjamin Keith Ford) |
New Features made in this release:
-----------------------------------
| Option to allow a user to not hear the join sound on enter but everyone else can | (Reported by Michael Cargile) | |
| func_db: Add a function to return cardinality of keys at prefix | (Reported by N A) | |
| Hint-like extension value lookup function without device state | (Reported by N A) | |
| chan_pjsip: Add ability to send flash events | (Reported by N A) | |
| cli: Add command to evaluate a function | (Reported by N A) | |
| app_queue: Add music on hold option | (Reported by N A) |
Bugs fixed in this release:
-----------------------------------
| chan_dahdi: Caller ID FSK Erroneously Sent when Picking Up Dahdi Call On Hold | (Reported by Josh Alberts) | |
| chan_dahdi: adding ring cadences is not idempotent on dahdi restart | (Reported by N A) | |
| chan_iax2: Prevent crashes due to attempted encryption with missing secrets | (Reported by N A) | |
| menuselect: Disabled by default modules that are enabled are always recompiled | (Reported by N A) | |
| app_meetme: Don't erroneously set global variables when channel is NULL | (Reported by N A) | |
| chan_dahdi: Round robin array size is too small for max number of groups | (Reported by N A) | |
| Asterisk's "T" flag is ignored when used with "r" or "R" flags. (documentation bug) | (Reported by Rusty Newton) | |
| Asterisk seems to ignore the "n" parameter for "disable console colorization" | (Reported by Sebastian Gutierrez) | |
| Session timers get removed on UPDATE | (Reported by Mark Petersen) | |
| file.c: seeking to negative file offset is not prevented | (Reported by N A) | |
| chan_sip: SIP route header is missing on UPDATE | (Reported by Mark Petersen) | |
| Do not change 180 Ringing to 183 Progress even if early_media already enabled | (Reported by Mark Petersen) | |
| iostream: Infinite TCP timeout writing data | (Reported by N A) | |
| Incorrect bridging on transfer | (Reported by Yury Kirsanov) | |
| Failed to sign STIR/SHAKEN payload with functionality not enabled | (Reported by Claude Diderich) | |
| res_pjsip: UDP transport does not work when async_operations is greater than 1 | (Reported by Ross Beer) | |
| res_pjsip_session: No video to caller if no camera available | (Reported by Michael Auracher) | |
| res_pjsip_session: No video after early media | (Reported by Michael Auracher) | |
| pjsip / WebRTC: Chrome creating large number of SDP attributes | (Reported by Josh Hogan) | |
| ast_variable_list_replace_variable uses variable with new keyword | (Reported by Jasper Hafkenscheid) | |
| cdr_adaptive_odbc: does not support DATETIME database columns | (Reported by Gregory Massel) | |
| Crash in pjsip_msg_find_hdr_by_name | (Reported by LA) | |
| Segmentation fault in libasteriskpj.so.2 | (Reported by Daniel Bonazzi) | |
| pbx: Only up to 127 includes in a dialplan context (AST_PBX_MAX_STACK - 1) | (Reported by Tzafrir Cohen) | |
| REGRESSION: The build process is requiring xmllint or xmlstarlet ro be installed when it shouldn't | (Reported by George Joseph) | |
| build: Asterisk 18.11.0 doesn't compile when wget isn't available | (Reported by Stefan Ruijsenaars) | |
| chan_iax2: Fix misaligned spacing in iax2 show netstats printout | (Reported by N A) | |
| agi: Fix xmldoc bug with set music | (Reported by N A) | |
| documentation: AGICommand_set+music documentation arguments displayed incorreclty | (Reported by Jonathan Harris) | |
| chan_iax2: "iax2 show registry" shows host for perceived | (Reported by David Herselman) | |
| res_pjsip_sdp_rtp: 183 Session in Progress. Disconnecting channel for lack of RTP activity | (Reported by Dmitriy Serov) | |
| res_pjsip_sdp_rtp: Disconnecting channel for lack of RTP activity in one way sessions | (Reported by Boris P. Korzun) | |
| Adjust for 64bit time_t | (Reported by Andre Heider) | |
| RLS: domain part of 'uri' list attribute mismatch with SUBSCRIBE request | (Reported by Alexei Gradinari) | |
| SayNumber can handle '01' to '07', but not '08' or '09' | (Reported by Jim Van Meggelen) | |
| logging messages truncated when using MUSL runtime | (Reported by Philip Prindeville) | |
| ari: Retrieving stored recording can returns wrong file | (Reported by Arix) |
Improvements made in this release:
-----------------------------------
| Missing documentation for chan_dahdi dial string ring cadences | (Reported by Scott Griepentrog) | |
| general: Add since tags to xmldocs | (Reported by N A) | |
| app_mf, app_sf: Return -1 on hangup | (Reported by N A) | |
| app_meetme: Emit warning if conference not found | (Reported by N A) | |
| Qualify pjproject 2.12 for Asterisk | (Reported by George Joseph) | |
| app_mf: Allow reading a maximum number of digits | (Reported by N A) | |
| Should Readme include information about install_prereq script? | (Reported by Marcel Wagner) | |
| Use pkg-config to find libxml2 headers and libraries | (Reported by Hugh McMaster) | |
| Documentation: Document explanations and examples for possible values of DIALSTATUS | (Reported by Rusty Newton) | |
| build: External binary modules don't use https | (Reported by INVADE International Ltd.) | |
| pbx_builtins: Add missing documentation | (Reported by N A) |
For a full list of changes in this release, please see the ChangeLog:
https://downloads.asterisk.org/pub/telephony/asterisk/ChangeLog-16.26.0
Thank you for your continued support of Asterisk!
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