From: Asterisk Development Team Date: Thu, 30 Oct 2025 16:15:07 +0000 (+0000) Subject: Update for 21.12.0-rc1 X-Git-Tag: 21.12.0-rc1^0 X-Git-Url: http://git.ipfire.org/gitweb.cgi?a=commitdiff_plain;h=refs%2Fheads%2Freleases%2F21;p=thirdparty%2Fasterisk.git Update for 21.12.0-rc1 --- diff --git a/.version b/.version index b570734337..2b05b778d5 100644 --- a/.version +++ b/.version @@ -1 +1 @@ -21.11.0 +21.12.0-rc1 diff --git a/CHANGES.html b/CHANGES.html index cd74cb4ebb..deb30b84c5 120000 --- a/CHANGES.html +++ b/CHANGES.html @@ -1 +1 @@ -ChangeLogs/ChangeLog-21.11.0.html \ No newline at end of file +ChangeLogs/ChangeLog-21.12.0-rc1.html \ No newline at end of file diff --git a/CHANGES.md b/CHANGES.md index 8a17bbb41f..5f02b48f54 120000 --- a/CHANGES.md +++ b/CHANGES.md @@ -1 +1 @@ -ChangeLogs/ChangeLog-21.11.0.md \ No newline at end of file +ChangeLogs/ChangeLog-21.12.0-rc1.md \ No newline at end of file diff --git a/ChangeLogs/ChangeLog-21.12.0-rc1.html b/ChangeLogs/ChangeLog-21.12.0-rc1.html new file mode 100644 index 0000000000..5374b12e8d --- /dev/null +++ b/ChangeLogs/ChangeLog-21.12.0-rc1.html @@ -0,0 +1,309 @@ +ChangeLog for asterisk-21.12.0-rc1 +

Change Log for Release asterisk-21.12.0-rc1

+

Links:

+ +

Summary:

+ +

User Notes:

+ +

Upgrade Notes:

+ +

Developer Notes:

+

Commit Authors:

+ +

Issue and Commit Detail:

+

Closed Issues:

+ +

Commits By Author:

+ +

Commit List:

+ +

Commit Details:

+

safe_asterisk: Resolve a POSIX sh problem and restore globbing behavior.

+

Author: Sean Bright + Date: 2025-10-22

+ +

Resolves: #1554

+

safe_asterisk: Fix logging and sorting issue.

+

Author: George Joseph + Date: 2025-10-17

+

Re-enabled "TTY=9" which was erroneously disabled as part of a recent + security fix and removed another logging "fix" that was added.

+

Also added a sort to the "find" that enumerates the scripts to be sourced so + they're sourced in the correct order.

+

Resolves: #1539

+

res_audiosocket: add message types for all slin sample rates

+

Author: Sven Kube + Date: 2025-10-10

+

Extend audiosocket messages with types 0x11 - 0x18 to create asterisk + frames in slin12, slin16, slin24, slin32, slin44, slin48, slin96, and + slin192 format, enabling the transmission of audio at a higher sample + rates. For audiosocket messages sent by Asterisk, the message kind is + determined by the format of the originating asterisk frame.

+

UpgradeNote: New audiosocket message types 0x11 - 0x18 has been added + for slin12, slin16, slin24, slin32, slin44, slin48, slin96, and + slin192 audio. External applications using audiosocket may need to be + updated to support these message types if the audiosocket channel is + created with one of these audio formats.

+

chan_websocket.c: Change payload references to command instead.

+

Author: George Joseph + Date: 2025-10-08

+

Some of the tests in process_text_message() were still comparing to the + websocket message payload instead of the "command" string.

+

Resolves: #1525

+

func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()

+

Author: Igor Goncharovsky + Date: 2025-09-04

+

As soon as SIP call may end with several Reason headers, we + want to make all of them available through the HAGUPCAUSE() function. + This implementation uses the same ao2 hash for cause codes storage + and adds a flag to make difference between last processed sip + message and content of reason headers.

+

UserNote: Added a new option to HANGUPCAUSE to access additional + information about hangup reason. Reason headers from pjsip + could be read using 'tech_extended' cause type.

+

channelstorage_cpp_map_name_id: Add read locking around retrievals.

+

Author: George Joseph + Date: 2025-10-01

+

When we retrieve a channel from a C++ map, we actually get back a wrapper + object that points to the channel then right after we retrieve it, we bump its + reference count. There's a tiny chance however that between those two + statements a delete and/or unref might happen which would cause the wrapper + object or the channel itself to become invalid resulting in a SEGV. To avoid + this we now perform a read lock on the driver around those statements.

+

Resolves: #1491

+

res_pjsip_geolocation: Add support for Geolocation loc-src parameter

+

Author: Max Grobecker + Date: 2025-09-21

+

This adds support for the Geolocation 'loc-src' parameter to res_pjsip_geolocation. + The already existing config option 'location_source` in res_geolocation is documented to add a 'loc-src' parameter containing a user-defined FQDN to the 'Geolocation:' header, + but that option had no effect as it was not implemented by res_pjsip_geolocation.

+

If the location_source configuration option is not set or invalid, that parameter will not be added (this is already checked by res_geolocation).

+

This commits adds already documented functionality.

+

stasis_channels.c: Make protocol_id optional to enable blind transfer via ari

+

Author: Sven Kube + Date: 2025-09-22

+

When handling SIP transfers via ARI, there is no protocol_id in case of + a blind transfer.

+

Resolves: #1467

+

Fix some doxygen, typos and whitespace

+

Author: Bastian Triller + Date: 2025-09-21

+

stasis_channels.c: Add null check for referred_by in ast_ari_transfer_message_..

+

Author: Sven Kube + Date: 2025-09-18

+

When handling SIP transfers via ARI, the referred_by field in + transfer_ari_state may be null, since SIP REFER requests are not + required to include a Referred-By header. Without this check, a null + value caused the transfer to fail and triggered a NOTIFY with a 500 + Internal Server Error.

+

app_queue: Add NULL pointer checks in app_queue

+

Author: phoneben + Date: 2025-09-11

+

Add NULL check for word_list before calling word_in_list() + Add NULL checks for channel snapshots from ast_multi_channel_blob_get_channel()

+

Resolves: #1425

+

app_externalivr: Prevent out-of-bounds read during argument processing.

+

Author: Sean Bright + Date: 2025-09-17

+

Resolves: #1422

+

chan_dahdi: Add DAHDI_CHANNEL function.

+

Author: Naveen Albert + Date: 2025-09-11

+

Add a dialplan function that can be used to get/set properties of + DAHDI channels (as opposed to Asterisk channels). This exposes + properties that were not previously available, allowing for certain + operations to now be performed in the dialplan.

+

Resolves: #1455

+

UserNote: The DAHDI_CHANNEL function allows for getting/setting + certain properties about DAHDI channels from the dialplan.

+

app_adsiprog: Fix possible NULL dereference.

+

Author: Naveen Albert + Date: 2025-09-10

+

get_token can return NULL, but process_token uses this result without + checking for NULL; as elsewhere, check for a NULL result to avoid + possible NULL dereference.

+

Resolves: #1419

+

manager.c: Fix presencestate object leak

+

Author: Nathan Monfils + Date: 2025-09-08

+

ast_presence_state allocates subtype and message. We straightforwardly + need to clean those up.

+

audiohook.c: Ensure correct AO2 reference is dereffed.

+

Author: Sean Bright + Date: 2025-09-10

+

Part of #1440.

+

res_cliexec: Remove unnecessary casts to char*.

+

Author: Naveen Albert + Date: 2025-09-09

+

Resolves: #1436

+

rtp_engine.c: Add exception for comfort noise payload.

+

Author: Ben Ford + Date: 2025-09-09

+

In a previous commit, a change was made to + ast_rtp_codecs_payload_code_tx_sample_rate to check for differing sample + rates. This ended up returning an invalid payload int for comfort noise. + A check has been added that returns early if the payload is in fact + supposed to be comfort noise.

+

Fixes: #1340

+

pbx_variables.c: Create real channel for "dialplan eval function".

+

Author: Naveen Albert + Date: 2025-09-09

+

"dialplan eval function" has been using a dummy channel for function + evaluation, much like many of the unit tests. However, sometimes, this + can cause issues for functions that are not expecting dummy channels. + As an example, ast_channel_tech(chan) is NULL on such channels, and + ast_channel_tech(chan)->type consequently results in a NULL dereference. + Normally, functions do not worry about this since channels executing + dialplan aren't dummy channels.

+

While some functions are better about checking for these sorts of edge + cases, use a real channel with a dummy technology to make this CLI + command inherently safe for any dialplan function that could be evaluated + from the CLI.

+

Resolves: #1434

+ diff --git a/ChangeLogs/ChangeLog-21.12.0-rc1.md b/ChangeLogs/ChangeLog-21.12.0-rc1.md new file mode 100644 index 0000000000..5d9d24104d --- /dev/null +++ b/ChangeLogs/ChangeLog-21.12.0-rc1.md @@ -0,0 +1,336 @@ + +## Change Log for Release asterisk-21.12.0-rc1 + +### Links: + + - [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.12.0-rc1.html) + - [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.11.0...21.12.0-rc1) + - [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.12.0-rc1.tar.gz) + - [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) + +### Summary: + +- Commits: 19 +- Commit Authors: 10 +- Issues Resolved: 12 +- Security Advisories Resolved: 0 + +### User Notes: + +- #### func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE() + Added a new option to HANGUPCAUSE to access additional + information about hangup reason. Reason headers from pjsip + could be read using 'tech_extended' cause type. + +- #### chan_dahdi: Add DAHDI_CHANNEL function. + The DAHDI_CHANNEL function allows for getting/setting + certain properties about DAHDI channels from the dialplan. + + +### Upgrade Notes: + +- #### res_audiosocket: add message types for all slin sample rates + New audiosocket message types 0x11 - 0x18 has been added + for slin12, slin16, slin24, slin32, slin44, slin48, slin96, and + slin192 audio. External applications using audiosocket may need to be + updated to support these message types if the audiosocket channel is + created with one of these audio formats. + + +### Developer Notes: + + +### Commit Authors: + +- Bastian Triller: (1) +- Ben Ford: (1) +- George Joseph: (3) +- Igor Goncharovsky: (1) +- Max Grobecker: (1) +- Nathan Monfils: (1) +- Naveen Albert: (4) +- Phoneben: (1) +- Sean Bright: (3) +- Sven Kube: (3) + +## Issue and Commit Detail: + +### Closed Issues: + + - 1340: [bug]: comfort noise packet corrupted + - 1419: [bug]: static code analysis issues in app_adsiprog.c + - 1422: [bug]: static code analysis issues in apps/app_externalivr.c + - 1425: [bug]: static code analysis issues in apps/app_queue.c + - 1434: [improvement]: pbx_variables: Create real channel for dialplan eval CLI command + - 1436: [improvement]: res_cliexec: Avoid unnecessary cast to char* + - 1455: [new-feature]: chan_dahdi: Add DAHDI_CHANNEL function + - 1467: [bug]: Crash in res_pjsip_refer during REFER progress teardown with PJSIP_TRANSFER_HANDLING(ari-only) + - 1491: [bug]: Segfault: `channelstorage_cpp` fast lookup without lock (`get_by_name_exact`/`get_by_uniqueid`) leads to UAF during hangup + - 1525: [bug]: chan_websocket: fix use of raw payload variable for string comparison in process_text_message + - 1539: [bug]: safe_asterisk without TTY doesn't log to file + - 1554: [bug]: safe_asterisk recurses into subdirectories of startup.d after f97361 + +### Commits By Author: + +- #### Bastian Triller (1): + - Fix some doxygen, typos and whitespace + +- #### Ben Ford (1): + - rtp_engine.c: Add exception for comfort noise payload. + +- #### George Joseph (3): + - channelstorage_cpp_map_name_id: Add read locking around retrievals. + - chan_websocket.c: Change payload references to command instead. + - safe_asterisk: Fix logging and sorting issue. + +- #### Igor Goncharovsky (1): + - func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE() + +- #### Max Grobecker (1): + - res_pjsip_geolocation: Add support for Geolocation loc-src parameter + +- #### Nathan Monfils (1): + - manager.c: Fix presencestate object leak + +- #### Naveen Albert (4): + - pbx_variables.c: Create real channel for "dialplan eval function". + - res_cliexec: Remove unnecessary casts to char*. + - app_adsiprog: Fix possible NULL dereference. + - chan_dahdi: Add DAHDI_CHANNEL function. + +- #### Sean Bright (3): + - audiohook.c: Ensure correct AO2 reference is dereffed. + - app_externalivr: Prevent out-of-bounds read during argument processing. + - safe_asterisk: Resolve a POSIX sh problem and restore globbing behavior. + +- #### Sven Kube (3): + - stasis_channels.c: Add null check for referred_by in ast_ari_transfer_message_.. + - stasis_channels.c: Make protocol_id optional to enable blind transfer via ari + - res_audiosocket: add message types for all slin sample rates + +- #### phoneben (1): + - app_queue: Add NULL pointer checks in app_queue + + +### Commit List: + +- safe_asterisk: Resolve a POSIX sh problem and restore globbing behavior. +- safe_asterisk: Fix logging and sorting issue. +- res_audiosocket: add message types for all slin sample rates +- chan_websocket.c: Change payload references to command instead. +- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE() +- channelstorage_cpp_map_name_id: Add read locking around retrievals. +- res_pjsip_geolocation: Add support for Geolocation loc-src parameter +- stasis_channels.c: Make protocol_id optional to enable blind transfer via ari +- Fix some doxygen, typos and whitespace +- app_queue: Add NULL pointer checks in app_queue +- app_externalivr: Prevent out-of-bounds read during argument processing. +- chan_dahdi: Add DAHDI_CHANNEL function. +- app_adsiprog: Fix possible NULL dereference. +- manager.c: Fix presencestate object leak +- audiohook.c: Ensure correct AO2 reference is dereffed. +- res_cliexec: Remove unnecessary casts to char*. +- rtp_engine.c: Add exception for comfort noise payload. +- pbx_variables.c: Create real channel for "dialplan eval function". + +### Commit Details: + +#### safe_asterisk: Resolve a POSIX sh problem and restore globbing behavior. + Author: Sean Bright + Date: 2025-10-22 + + * Using `==` with the POSIX sh `test` utility is UB. + * Switch back to using globs instead of using `$(find … | sort)`. + * Fix a missing redirect when checking for the OS type. + + Resolves: #1554 + +#### safe_asterisk: Fix logging and sorting issue. + Author: George Joseph + Date: 2025-10-17 + + Re-enabled "TTY=9" which was erroneously disabled as part of a recent + security fix and removed another logging "fix" that was added. + + Also added a sort to the "find" that enumerates the scripts to be sourced so + they're sourced in the correct order. + + Resolves: #1539 + +#### res_audiosocket: add message types for all slin sample rates + Author: Sven Kube + Date: 2025-10-10 + + Extend audiosocket messages with types 0x11 - 0x18 to create asterisk + frames in slin12, slin16, slin24, slin32, slin44, slin48, slin96, and + slin192 format, enabling the transmission of audio at a higher sample + rates. For audiosocket messages sent by Asterisk, the message kind is + determined by the format of the originating asterisk frame. + + UpgradeNote: New audiosocket message types 0x11 - 0x18 has been added + for slin12, slin16, slin24, slin32, slin44, slin48, slin96, and + slin192 audio. External applications using audiosocket may need to be + updated to support these message types if the audiosocket channel is + created with one of these audio formats. + +#### chan_websocket.c: Change payload references to command instead. + Author: George Joseph + Date: 2025-10-08 + + Some of the tests in process_text_message() were still comparing to the + websocket message payload instead of the "command" string. + + Resolves: #1525 + +#### func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE() + Author: Igor Goncharovsky + Date: 2025-09-04 + + As soon as SIP call may end with several Reason headers, we + want to make all of them available through the HAGUPCAUSE() function. + This implementation uses the same ao2 hash for cause codes storage + and adds a flag to make difference between last processed sip + message and content of reason headers. + + UserNote: Added a new option to HANGUPCAUSE to access additional + information about hangup reason. Reason headers from pjsip + could be read using 'tech_extended' cause type. + +#### channelstorage_cpp_map_name_id: Add read locking around retrievals. + Author: George Joseph + Date: 2025-10-01 + + When we retrieve a channel from a C++ map, we actually get back a wrapper + object that points to the channel then right after we retrieve it, we bump its + reference count. There's a tiny chance however that between those two + statements a delete and/or unref might happen which would cause the wrapper + object or the channel itself to become invalid resulting in a SEGV. To avoid + this we now perform a read lock on the driver around those statements. + + Resolves: #1491 + +#### res_pjsip_geolocation: Add support for Geolocation loc-src parameter + Author: Max Grobecker + Date: 2025-09-21 + + This adds support for the Geolocation 'loc-src' parameter to res_pjsip_geolocation. + The already existing config option 'location_source` in res_geolocation is documented to add a 'loc-src' parameter containing a user-defined FQDN to the 'Geolocation:' header, + but that option had no effect as it was not implemented by res_pjsip_geolocation. + + If the `location_source` configuration option is not set or invalid, that parameter will not be added (this is already checked by res_geolocation). + + This commits adds already documented functionality. + +#### stasis_channels.c: Make protocol_id optional to enable blind transfer via ari + Author: Sven Kube + Date: 2025-09-22 + + When handling SIP transfers via ARI, there is no protocol_id in case of + a blind transfer. + + Resolves: #1467 + +#### Fix some doxygen, typos and whitespace + Author: Bastian Triller + Date: 2025-09-21 + + +#### stasis_channels.c: Add null check for referred_by in ast_ari_transfer_message_.. + Author: Sven Kube + Date: 2025-09-18 + + When handling SIP transfers via ARI, the `referred_by` field in + `transfer_ari_state` may be null, since SIP REFER requests are not + required to include a `Referred-By` header. Without this check, a null + value caused the transfer to fail and triggered a NOTIFY with a 500 + Internal Server Error. + +#### app_queue: Add NULL pointer checks in app_queue + Author: phoneben + Date: 2025-09-11 + + Add NULL check for word_list before calling word_in_list() + Add NULL checks for channel snapshots from ast_multi_channel_blob_get_channel() + + Resolves: #1425 + +#### app_externalivr: Prevent out-of-bounds read during argument processing. + Author: Sean Bright + Date: 2025-09-17 + + Resolves: #1422 + +#### chan_dahdi: Add DAHDI_CHANNEL function. + Author: Naveen Albert + Date: 2025-09-11 + + Add a dialplan function that can be used to get/set properties of + DAHDI channels (as opposed to Asterisk channels). This exposes + properties that were not previously available, allowing for certain + operations to now be performed in the dialplan. + + Resolves: #1455 + + UserNote: The DAHDI_CHANNEL function allows for getting/setting + certain properties about DAHDI channels from the dialplan. + +#### app_adsiprog: Fix possible NULL dereference. + Author: Naveen Albert + Date: 2025-09-10 + + get_token can return NULL, but process_token uses this result without + checking for NULL; as elsewhere, check for a NULL result to avoid + possible NULL dereference. + + Resolves: #1419 + +#### manager.c: Fix presencestate object leak + Author: Nathan Monfils + Date: 2025-09-08 + + ast_presence_state allocates subtype and message. We straightforwardly + need to clean those up. + +#### audiohook.c: Ensure correct AO2 reference is dereffed. + Author: Sean Bright + Date: 2025-09-10 + + Part of #1440. + +#### res_cliexec: Remove unnecessary casts to char*. + Author: Naveen Albert + Date: 2025-09-09 + + Resolves: #1436 + +#### rtp_engine.c: Add exception for comfort noise payload. + Author: Ben Ford + Date: 2025-09-09 + + In a previous commit, a change was made to + ast_rtp_codecs_payload_code_tx_sample_rate to check for differing sample + rates. This ended up returning an invalid payload int for comfort noise. + A check has been added that returns early if the payload is in fact + supposed to be comfort noise. + + Fixes: #1340 + +#### pbx_variables.c: Create real channel for "dialplan eval function". + Author: Naveen Albert + Date: 2025-09-09 + + "dialplan eval function" has been using a dummy channel for function + evaluation, much like many of the unit tests. However, sometimes, this + can cause issues for functions that are not expecting dummy channels. + As an example, ast_channel_tech(chan) is NULL on such channels, and + ast_channel_tech(chan)->type consequently results in a NULL dereference. + Normally, functions do not worry about this since channels executing + dialplan aren't dummy channels. + + While some functions are better about checking for these sorts of edge + cases, use a real channel with a dummy technology to make this CLI + command inherently safe for any dialplan function that could be evaluated + from the CLI. + + Resolves: #1434 + diff --git a/README.html b/README.html index 5a4a9fcb89..67fb32f684 100644 --- a/README.html +++ b/README.html @@ -1,4 +1,4 @@ -Readme for asterisk-21.11.0 +Readme for asterisk-21.12.0-rc1

The Asterisk(R) Open Source PBX

By Mark Spencer <markster@digium.com> and the Asterisk.org developer community.
 Copyright (C) 2001-2025 Sangoma Technologies Corporation and other copyright holders.
@@ -37,7 +37,7 @@ hardware.

 
If you are updating from a previous version of Asterisk, make sure you
 read the Change Logs.

 
-
Change Logs

+
Change Logs

 
 
 
NEW INSTALLATIONS

diff --git a/README.md b/README.md
index cf78772940..ca103c6d73 100644
--- a/README.md
+++ b/README.md
@@ -55,7 +55,7 @@ If you are updating from a previous version of Asterisk, make sure you
 read the Change Logs.
 
 
-[Change Logs](ChangeLogs/ChangeLog-21.11.0.html)
+[Change Logs](ChangeLogs/ChangeLog-21.12.0-rc1.html)
 
 
 ### NEW INSTALLATIONS