From 2c73433fb83d39674292440f195b7b3adc68c550 Mon Sep 17 00:00:00 2001 From: Asterisk Development Team Date: Thu, 30 Oct 2025 16:16:44 +0000 Subject: [PATCH] Update for 22.7.0-rc1 --- .version | 2 +- CHANGES.html | 2 +- CHANGES.md | 2 +- ChangeLogs/ChangeLog-22.7.0-rc1.html | 902 ++++++++++++++++++++++++ ChangeLogs/ChangeLog-22.7.0-rc1.md | 993 +++++++++++++++++++++++++++ README.html | 4 +- README.md | 2 +- 7 files changed, 1901 insertions(+), 6 deletions(-) create mode 100644 ChangeLogs/ChangeLog-22.7.0-rc1.html create mode 100644 ChangeLogs/ChangeLog-22.7.0-rc1.md diff --git a/.version b/.version index dc5f6a52b1..93211c7d24 100644 --- a/.version +++ b/.version @@ -1 +1 @@ -22.6.0 +22.7.0-rc1 diff --git a/CHANGES.html b/CHANGES.html index ee9daa44a0..07dc4dfb1c 120000 --- a/CHANGES.html +++ b/CHANGES.html @@ -1 +1 @@ -ChangeLogs/ChangeLog-22.6.0.html \ No newline at end of file +ChangeLogs/ChangeLog-22.7.0-rc1.html \ No newline at end of file diff --git a/CHANGES.md b/CHANGES.md index fb349a835c..719239a71c 120000 --- a/CHANGES.md +++ b/CHANGES.md @@ -1 +1 @@ -ChangeLogs/ChangeLog-22.6.0.md \ No newline at end of file +ChangeLogs/ChangeLog-22.7.0-rc1.md \ No newline at end of file diff --git a/ChangeLogs/ChangeLog-22.7.0-rc1.html b/ChangeLogs/ChangeLog-22.7.0-rc1.html new file mode 100644 index 0000000000..b99864872c --- /dev/null +++ b/ChangeLogs/ChangeLog-22.7.0-rc1.html @@ -0,0 +1,902 @@ +ChangeLog for asterisk-22.7.0-rc1 +

Change Log for Release asterisk-22.7.0-rc1

+

Links:

+ +

Summary:

+ +

User Notes:

+ +

Upgrade Notes:

+ +

Developer Notes:

+ +

Commit Authors:

+ +

Issue and Commit Detail:

+

Closed Issues:

+ +

Commits By Author:

+ +

Commit List:

+ +

Commit Details:

+

res_audiosocket: fix temporarily unavailable

+

Author: Roman Pertsev + Date: 2025-10-07

+

Operations on non-blocking sockets may return a resource temporarily unavailable error (EAGAIN or EWOULDBLOCK). This is not a fatal error but a normal condition indicating that the operation would block.

+

This patch corrects the handling of this case. Instead of incorrectly treating it as a reason to terminate the connection, the code now waits for data to arrive on the socket.

+

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

+

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

+ +

Resolves: #1554

+

res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function.

+

Author: George Joseph + Date: 2025-10-24

+

Also...

+ +

Resolves: #781

+

UserNote: The STIR_SHAKEN_ATTESTATION dialplan function has been added + which will allow suppressing attestation on a call-by-call basis + regardless of the profile attached to the outgoing endpoint.

+

Revert "pjsip: Move from threadpool to taskpool"

+

Author: Joshua C. Colp + Date: 2025-10-27

+

This reverts commit bb6b76c2d8239b2665223dcbf6d507aa9aa4534e.

+

iostream.c: Handle TLS handshake attacks in order to resolve the issue of exce..

+

Author: Tinet-mucw + Date: 2025-10-26

+

The TCP three-way handshake completes, but if the server is under a TLS handshake attack, asterisk will get stuck at SSL_do_handshake(). + In this case, a timeout mechanism should be set for the SSL/TLS handshake process to prevent indefinite waiting during the SSL handshake.

+

Resolves: #1559

+

chan_pjsip: Disable SSRC change for WebRTC endpoints.

+

Author: George Joseph + Date: 2025-10-21

+

Commit b333ee3b introduced a fix to chan_pjsip that addressed RTP issues with + blind transfers and some SBCs. Unfortunately, the fix broke some WebRTC + clients that are sensitive to SSRC changes and non-monotonic timestamps so + the fix is now disabled for endpoints with the "bundle" parameter set to true.

+

Resolves: #1535

+

chan_websocket: Add channel_id to MEDIA_START, DRIVER_STATUS and DTMF_END even..

+

Author: gauravs456 + Date: 2025-10-21

+

Resolves: #1544

+

pjsip: Move from threadpool to taskpool

+

Author: Joshua C. Colp + Date: 2025-09-23

+

This change moves the PJSIP module from the threadpool API + to the taskpool API. PJSIP-specific implementations for + task usage have been removed and replaced with calls to + the optimized taskpool implementations instead. The need + for a pool of serializers has also been removed as + taskpool inherently provides this. The default settings + have also been changed to be more realistic for common + usage.

+

UpgradeNote: The threadpool_* options in pjsip.conf have now + been deprecated though they continue to be read and used. + They have been replaced with taskpool options that give greater + control over the underlying taskpool used for PJSIP. An alembic + upgrade script has been added to add these options to realtime + as well.

+

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

+

Fix Endianness detection in utils.h for non-Linux

+

Author: Christoph Moench-Tegeder + Date: 2025-10-19

+

Commit 43bf8a4ded7a65203b766b91eaf8331a600e9d8d introduced endian + dependend byte-swapping code in include/asterisk/utils.h, where the + endianness was detected using the __BYTE_ORDER macro. This macro + lives in endian.h, which on Linux is included implicitely (by the + network-related headers, I think), but on FreeBSD the headers are + laid out differently and we do not get __BYTE_ORDER the implicit way.

+

Instead, this makes the usage of endian.h explicit by including it + where we need it, and switches the BYTE_ORDER/*ENDIAN macros to the + POSIX-defined ones (see + https://pubs.opengroup.org/onlinepubs/9799919799/basedefs/endian.h.html + for standard compliance). Additionally, this adds a compile-time check + for the endianness-logic: compilation will fail if neither big nor + little endian can be detected.

+

Fixes: #1536

+

app_queue.c: Fix error in Queue parameter documentation.

+

Author: Ben Ford + Date: 2025-10-20

+

When macro was removed in Asterisk 21, the parameter documentation in + code was not updated to reflect the correct numerization for gosub. It + still stated that it was the seventh parameter, but got shifted to the + sixth due to the removal of macro. This has been updated to correctly + reflect the parameter order, and a note has been added to the XML that + states this was done after the initial commit.

+

Fixes: #1534

+

UpgradeNote: As part of Asterisk 21, macros were removed from Asterisk. + This resulted in argument order changing for the Queue dialplan + application since the macro argument was removed. Upgrade notice was + missed when this was done, so this upgrade note has been added to + provide a record of such and a notice to users who may have not upgraded + yet.

+

devicestate: Don't publish redundant device state messages.

+

Author: Joshua C. Colp + Date: 2025-10-17

+

When publishing device state check the local cache for the + existing device state. If the new device state is unchanged + from the prior one, don't bother publishing the update. This + can reduce the work done by consumers of device state, such + as hints and app_queue, by not publishing a message to them.

+

These messages would most often occur with devices that are + seeing numerous simultaneous channels. The underlying device + state would remain as in use throughout, but an update would + be published as channels are created and hung up.

+

chan_pjsip: Add technology-specific off-nominal hangup cause to events.

+

Author: George Joseph + Date: 2025-10-14

+

Although the ISDN/Q.850/Q.931 hangup cause code is already part of the ARI + and AMI hangup and channel destroyed events, it can be helpful to know what + the actual channel technology code was if the call was unsuccessful. + For PJSIP, it's the SIP response code.

+ +

DeveloperNote: A "tech_cause" parameter has been added to the + ChannelHangupRequest and ChannelDestroyed ARI event messages and a "TechCause" + parameter has been added to the HangupRequest, SoftHangupRequest and Hangup + AMI event messages. For chan_pjsip, these will be set to the last SIP + response status code for off-nominally terminated calls. The parameter is + suppressed for nominal termination.

+

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.

+

res_fax.c: lower FAXOPT read warning to debug level

+

Author: phoneben + Date: 2025-10-03

+

Reading ${FAXOPT()} before a fax session is common in dialplans to check fax state. + Currently this logs an error even when no fax datastore exists, creating excessive noise. + Change these messages to ast_debug(3, …) so they appear only with debug enabled.

+

Resolves: #1509

+

endpoints: Remove need for stasis subscription.

+

Author: Joshua C. Colp + Date: 2025-10-10

+

When an endpoint is created in the core of Asterisk a subscription + was previously created alongside it to monitor any channels being + destroyed that were related to it. This was done by receiving all + channel snapshot updates for every channel and only reacting when + it was indicated that the channel was dead.

+

This change removes this logic and instead provides an API call + for directly removing a channel from an endpoint. This is called + when channels are destroyed. This operation is fast, so blocking + the calling thread for a short period of time doesn't have any + noticeable impact.

+

app_queue: Allow stasis message filtering to work.

+

Author: Joshua C. Colp + Date: 2025-10-10

+

The app_queue module subscribes on a per-dialed agent basis to both + the bridge all and channel all topics to keep apprised of things going + on involving them. This subscription has associated state that must + be cleaned up when the subscription ends. This was done by setting + a default router callback that only had logic to handle the case + where the subscription ends. By using the default router callback + all filtering for the subscription was disabled, causing unrelated + messages to get published and handled by it.

+

This change makes it so that an explicit route is added for the + message type used for the message indicating the subscription has + ended and removes the default router callback. This allows message + filtering to occur on publishing reducing the messages to app_queue + to only those it is interested in.

+

taskpool: Fix some references to threadpool that should be taskpool.

+

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

+

Resolves: #1478

+

Update contact information for anthm

+

Author: Anthony Minessale + Date: 2025-10-10

+

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_callerid: Document limitation of DNID fields.

+

Author: Naveen Albert + Date: 2025-10-06

+

The Dial() application does not propagate DNID fields, which is counter + to the behavior of the other Caller ID fields. This behavior is likely + intentional since the use of Dial theoretically suggests a new dialed + number, but document this caveat to inform users of it.

+

Resolves: #1519

+

func_channel: Allow R/W of ADSI CPE capability setting.

+

Author: Naveen Albert + Date: 2025-10-06

+

Allow retrieving and setting the channel's ADSI capability from the + dialplan.

+

Resolves: #1514

+

UserNote: CHANNEL(adsicpe) can now be read or written to change + the channels' ADSI CPE capability setting.

+

core_unreal: Preserve ADSI capability when dialing Local channels.

+

Author: Naveen Albert + Date: 2025-10-06

+

Dial() already preserves the ADSI capability by copying it to the new + channel, but since Local channel pairs consist of two channels, we + also need to copy the capability to the second channel.

+

Resolves: #1517

+

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.

+

sig_analog: Allow '#' to end the inter-digit timeout when dialing.

+

Author: Naveen Albert + Date: 2025-10-03

+

It is customary to allow # to terminate digit collection immediately + when there would normally be a timeout. However, currently, users are + forced to wait for the timeout to expire when dialing numbers that + are prefixes of other valid matches, and there is no way to end the + timeout early. Customarily, # terminates the timeout, but at the moment, + this is just rejected unless there happens to be a matching extension + ending in #.

+

Allow # to terminate the timeout in cases where there is no dialplan + match. This ensures that the dialplan is always respected, but if a + valid extension has been dialed that happens to prefix other valid + matches, # can be used to dial it immediately.

+

Resolves: #1510

+

func_math: Add DIGIT_SUM function.

+

Author: Naveen Albert + Date: 2025-10-01

+

Add a function (DIGIT_SUM) which returns the digit sum of a number.

+

Resolves: #1499

+

UserNote: The DIGIT_SUM function can be used to return the digit sum of + a number.

+

app_sf: Add post-digit timer option to ReceiveSF.

+

Author: Naveen Albert + Date: 2025-10-01

+

Add a sorely needed option to set a timeout between digits, rather than + for receiving the entire number. This is needed if the number of digits + being sent is unknown by the receiver in advance. Previously, we had + to wait for the entire timer to expire.

+

Resolves: #1493

+

UserNote: The 't' option for ReceiveSF now allows for a timer since + the last digit received, in addition to the number-wide timeout.

+

codec_builtin.c: Adjust some of the quality scores to reflect reality.

+

Author: Naveen Albert + Date: 2025-10-02

+

Among the lower-quality voice codecs, some of the quality scores did + not make sense relative to each other.

+

For instance, quality-wise, G.729 > G.723 > PLC10. + However, current scores do not uphold these relationships.

+

Tweak the scores slightly to reflect more accurate relationships.

+

Resolves: #1501

+

res_tonedetect: Fix formatting of XML documentation.

+

Author: Naveen Albert + Date: 2025-10-02

+

Fix the indentation in the documentation for the variable list.

+

Resolves: #1507

+

res_fax: Add XML documentation for channel variables.

+

Author: Naveen Albert + Date: 2025-10-02

+

Document the channel variables currently set by SendFAX and ReceiveFAX.

+

Resolves: #1505

+

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

+

app_dial: Allow fractional seconds for dial timeouts.

+

Author: Naveen Albert + Date: 2025-09-30

+

Even though Dial() internally uses milliseconds for its dial timeouts, + this capability has been mostly obscured from users as the argument is + only parsed as an integer, thus forcing the use of whole seconds for + timeouts.

+

Parse it as a decimal instead so that timeouts can now truly have + millisecond precision.

+

Resolves: #1487

+

UserNote: The answer and progress dial timeouts now have millisecond + precision, instead of having to be whole numbers.

+

dsp.c: Make minor fixes to debug log messages.

+

Author: Naveen Albert + Date: 2025-10-01

+

Commit dc8e3eeaaf094a3d16991289934093d5e7127680 improved the debug log + messages in dsp.c. This makes two minor corrections to it:

+ +

Resolves: #1496

+

config_options.c: Improve misleading warning.

+

Author: Naveen Albert + Date: 2025-09-30

+

When running "config show help ", if no XML documentation exists + for the specified module, "Module not found." is returned, + which is misleading if the module is loaded but simply has no XML + documentation for its config. Improve the message to clarify that the + module may simply have no config documentation.

+

Resolves: #1489

+

func_scramble: Add example to XML documentation.

+

Author: Naveen Albert + Date: 2025-09-29

+

The previous lack of an example made it ambiguous if the arguments went + inside the function arguments or were part of the right-hand value.

+

Resolves: #1485

+

sig_analog: Eliminate potential timeout with Last Number Redial.

+

Author: Naveen Albert + Date: 2025-09-29

+

If Last Number Redial is used to redial, ensure that we do not wait + for further digits. This was possible if the number that was last + dialed is a prefix of another possible dialplan match. Since all we + did is copy the number into the extension buffer, if other matches + are now possible, there would thus be a timeout before the call went + through. We now complete redialed calls immediaetly in all cases.

+

Resolves: #1483

+

ARI: The bridges play and record APIs now handle sample rates > 8K correctly.

+

Author: George Joseph + Date: 2025-09-25

+

The bridge play and record APIs were forcing the Announcer/Recorder channel + to slin8 which meant that if you played or recorded audio with a sample + rate > 8K, it was downsampled to 8K limiting the bandwidth.

+ +

Resolves: #1479

+

DeveloperNote: The ARI /bridges/play and /bridges/record REST APIs have new + parameters that allow the caller to specify the format to be used on the + "Announcer" and "Recorder" channels respecitvely.

+

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.

+

sorcery: Move from threadpool to taskpool.

+

Author: Joshua C. Colp + Date: 2025-09-23

+

This change moves observer invocation from the use of + a threadpool to a taskpool. The taskpool options have also + been adjusted to ensure that at least one taskprocessor + remains available at all times.

+

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.

+

taskpool: Update versions for taskpool stasis options.

+

Author: Joshua C. Colp + Date: 2025-09-16

+

taskpool: Add taskpool API, switch Stasis to using it.

+

Author: Joshua C. Colp + Date: 2025-08-06

+

This change introduces a new API called taskpool. This is a pool + of taskprocessors. It provides the following functionality:

+
    +
  1. Task pushing to a pool of taskprocessors
    2. +
  2. Synchronous tasks
    3. +
  3. Serializers for execution ordering of tasks
    4. +
  4. Growing/shrinking of number of taskprocessors in pool
    5. +
+

This functionality already exists through the combination of + threadpool+taskprocessors but through investigating I determined + that this carries substantial overhead for short to medium duration + tasks. The threadpool uses a single queue of work, and for management + of threads it involves additional tasks.

+

I wrote taskpool to eliminate the extra overhead and management + as much as possible. Instead of a single queue of work each + taskprocessor has its own queue and at push time a selector chooses + the taskprocessor to queue the task to. Each taskprocessor also + has its own thread like normal. This spreads out the tasks immediately + and reduces contention on shared resources.

+

Using the included efficiency tests the number of tasks that can be + executed per second in a taskpool is 6-12 times more than an equivalent + threadpool+taskprocessor setup.

+

Stasis has been moved over to using this new API as it is a heavy consumer + of threadpool+taskprocessors and produces a lot of tasks.

+

UpgradeNote: The threadpool_* options in stasis.conf have now been deprecated + though they continue to be read and used. They have been replaced with taskpool + options that give greater control over the underlying taskpool used for stasis.

+

DeveloperNote: The taskpool API has been added for common usage of a + pool of taskprocessors. It is suggested to use this API instead of the + threadpool+taskprocessor approach.

+

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-22.7.0-rc1.md b/ChangeLogs/ChangeLog-22.7.0-rc1.md new file mode 100644 index 0000000000..e3933365cf --- /dev/null +++ b/ChangeLogs/ChangeLog-22.7.0-rc1.md @@ -0,0 +1,993 @@ + +## Change Log for Release asterisk-22.7.0-rc1 + +### Links: + + - [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.7.0-rc1.html) + - [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.6.0...22.7.0-rc1) + - [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.7.0-rc1.tar.gz) + - [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) + +### Summary: + +- Commits: 53 +- Commit Authors: 16 +- Issues Resolved: 35 +- Security Advisories Resolved: 0 + +### User Notes: + +- #### res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function. + The STIR_SHAKEN_ATTESTATION dialplan function has been added + which will allow suppressing attestation on a call-by-call basis + regardless of the profile attached to the outgoing endpoint. + +- #### func_channel: Allow R/W of ADSI CPE capability setting. + CHANNEL(adsicpe) can now be read or written to change + the channels' ADSI CPE capability setting. + +- #### 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. + +- #### func_math: Add DIGIT_SUM function. + The DIGIT_SUM function can be used to return the digit sum of + a number. + +- #### app_sf: Add post-digit timer option to ReceiveSF. + The 't' option for ReceiveSF now allows for a timer since + the last digit received, in addition to the number-wide timeout. + +- #### app_dial: Allow fractional seconds for dial timeouts. + The answer and progress dial timeouts now have millisecond + precision, instead of having to be whole numbers. + +- #### chan_dahdi: Add DAHDI_CHANNEL function. + The DAHDI_CHANNEL function allows for getting/setting + certain properties about DAHDI channels from the dialplan. + + +### Upgrade Notes: + +- #### pjsip: Move from threadpool to taskpool + The threadpool_* options in pjsip.conf have now + been deprecated though they continue to be read and used. + They have been replaced with taskpool options that give greater + control over the underlying taskpool used for PJSIP. An alembic + upgrade script has been added to add these options to realtime + as well. + +- #### app_queue.c: Fix error in Queue parameter documentation. + As part of Asterisk 21, macros were removed from Asterisk. + This resulted in argument order changing for the Queue dialplan + application since the macro argument was removed. Upgrade notice was + missed when this was done, so this upgrade note has been added to + provide a record of such and a notice to users who may have not upgraded + yet. + +- #### 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. + +- #### taskpool: Add taskpool API, switch Stasis to using it. + The threadpool_* options in stasis.conf have now been deprecated + though they continue to be read and used. They have been replaced with taskpool + options that give greater control over the underlying taskpool used for stasis. + + +### Developer Notes: + +- #### chan_pjsip: Add technology-specific off-nominal hangup cause to events. + A "tech_cause" parameter has been added to the + ChannelHangupRequest and ChannelDestroyed ARI event messages and a "TechCause" + parameter has been added to the HangupRequest, SoftHangupRequest and Hangup + AMI event messages. For chan_pjsip, these will be set to the last SIP + response status code for off-nominally terminated calls. The parameter is + suppressed for nominal termination. + +- #### ARI: The bridges play and record APIs now handle sample rates > 8K correctly. + The ARI /bridges/play and /bridges/record REST APIs have new + parameters that allow the caller to specify the format to be used on the + "Announcer" and "Recorder" channels respecitvely. + +- #### taskpool: Add taskpool API, switch Stasis to using it. + The taskpool API has been added for common usage of a + pool of taskprocessors. It is suggested to use this API instead of the + threadpool+taskprocessor approach. + + +### Commit Authors: + +- Anthony Minessale: (1) +- Bastian Triller: (1) +- Ben Ford: (2) +- Christoph Moench-Tegeder: (1) +- Gauravs456: (1) +- George Joseph: (8) +- Igor Goncharovsky: (1) +- Joshua C. Colp: (8) +- Max Grobecker: (1) +- Nathan Monfils: (1) +- Naveen Albert: (18) +- Phoneben: (2) +- Roman Pertsev: (1) +- Sean Bright: (3) +- Sven Kube: (3) +- Tinet-Mucw: (1) + +## Issue and Commit Detail: + +### Closed Issues: + + - 781: [improvement]: Allow call by call disabling Stir/Shaken header inclusion + - 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) + - 1478: [improvement]: Stasis threadpool -> taskpool + - 1479: [bug]: The ARI bridge play and record APIs limit audio bandwidth by forcing the slin8 format. + - 1483: [improvement]: sig_analog: Eliminate possible timeout for Last Number Redial + - 1485: [improvement]: func_scramble: Add example to XML documentation. + - 1487: [improvement]: app_dial: Allow partial seconds to be used for dial timeouts + - 1489: [improvement]: config_options.c: Improve misleading error message + - 1491: [bug]: Segfault: `channelstorage_cpp` fast lookup without lock (`get_by_name_exact`/`get_by_uniqueid`) leads to UAF during hangup + - 1493: [new-feature]: app_sf: Add post-digit timer option + - 1496: [improvement]: dsp.c: Minor fixes to debug log messages + - 1499: [new-feature]: func_math: Add function to return the digit sum + - 1501: [improvement]: codec_builtin: Fix some inaccurate quality weights. + - 1505: [improvement]: res_fax: Add XML documentation for channel variables + - 1507: [improvement]: res_tonedetect: Minor formatting issue in documentation + - 1509: [improvement]: res_fax.c — log debug error as debug, not regular log + - 1510: [new-feature]: sig_analog: Allow '#' to end the inter-digit timeout when dialing. + - 1514: [improvement]: func_channel: Allow R/W of ADSI CPE capability setting. + - 1517: [improvement]: core_unreal: Preserve ADSI capability when dialing Local channels + - 1519: [improvement]: app_dial / func_callerid: DNIS information is not propagated by Dial + - 1525: [bug]: chan_websocket: fix use of raw payload variable for string comparison in process_text_message + - 1534: [bug]: app_queue when using gosub breaks dialplan when going from 20 to 21, What's new in 21 doesn't mention it's a breaking change, + - 1535: [bug]: chan_pjsip changes SSRC on WebRTC channels, which is unsupported by some browsers + - 1536: [bug]: asterisk -rx connects to console instead of executing a command + - 1539: [bug]: safe_asterisk without TTY doesn't log to file + - 1544: [improvement]: While Receiving the MediaConnect Message Using External Media Over websocket ChannelID is Details are missing + - 1554: [bug]: safe_asterisk recurses into subdirectories of startup.d after f97361 + - 1559: [improvement]: Handle TLS handshake attacks in order to resolve the issue of exceeding the maximum number of HTTPS sessions. + +### Commits By Author: + +- #### Anthony Minessale (1): + - Update contact information for anthm + +- #### Bastian Triller (1): + - Fix some doxygen, typos and whitespace + +- #### Ben Ford (2): + - rtp_engine.c: Add exception for comfort noise payload. + - app_queue.c: Fix error in Queue parameter documentation. + +- #### Christoph Moench-Tegeder (1): + - Fix Endianness detection in utils.h for non-Linux + +- #### George Joseph (8): + - ARI: The bridges play and record APIs now handle sample rates > 8K correctly. + - channelstorage_cpp_map_name_id: Add read locking around retrievals. + - chan_websocket.c: Change payload references to command instead. + - taskpool: Fix some references to threadpool that should be taskpool. + - chan_pjsip: Add technology-specific off-nominal hangup cause to events. + - safe_asterisk: Fix logging and sorting issue. + - chan_pjsip: Disable SSRC change for WebRTC endpoints. + - res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function. + +- #### Igor Goncharovsky (1): + - func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE() + +- #### Joshua C. Colp (8): + - taskpool: Add taskpool API, switch Stasis to using it. + - taskpool: Update versions for taskpool stasis options. + - sorcery: Move from threadpool to taskpool. + - app_queue: Allow stasis message filtering to work. + - endpoints: Remove need for stasis subscription. + - devicestate: Don't publish redundant device state messages. + - pjsip: Move from threadpool to taskpool + - Revert "pjsip: Move from threadpool to taskpool" + +- #### Max Grobecker (1): + - res_pjsip_geolocation: Add support for Geolocation loc-src parameter + +- #### Nathan Monfils (1): + - manager.c: Fix presencestate object leak + +- #### Naveen Albert (18): + - 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. + - sig_analog: Eliminate potential timeout with Last Number Redial. + - func_scramble: Add example to XML documentation. + - config_options.c: Improve misleading warning. + - dsp.c: Make minor fixes to debug log messages. + - app_dial: Allow fractional seconds for dial timeouts. + - res_fax: Add XML documentation for channel variables. + - res_tonedetect: Fix formatting of XML documentation. + - codec_builtin.c: Adjust some of the quality scores to reflect reality. + - app_sf: Add post-digit timer option to ReceiveSF. + - func_math: Add DIGIT_SUM function. + - sig_analog: Allow '#' to end the inter-digit timeout when dialing. + - core_unreal: Preserve ADSI capability when dialing Local channels. + - func_channel: Allow R/W of ADSI CPE capability setting. + - func_callerid: Document limitation of DNID fields. + +- #### Roman Pertsev (1): + - res_audiosocket: fix temporarily unavailable + +- #### 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 + +- #### Tinet-mucw (1): + - iostream.c: Handle TLS handshake attacks in order to resolve the issue of exce.. + +- #### gauravs456 (1): + - chan_websocket: Add channel_id to MEDIA_START, DRIVER_STATUS and DTMF_END even.. + +- #### phoneben (2): + - app_queue: Add NULL pointer checks in app_queue + - res_fax.c: lower FAXOPT read warning to debug level + + +### Commit List: + +- res_audiosocket: fix temporarily unavailable +- safe_asterisk: Resolve a POSIX sh problem and restore globbing behavior. +- res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function. +- Revert "pjsip: Move from threadpool to taskpool" +- chan_pjsip: Disable SSRC change for WebRTC endpoints. +- pjsip: Move from threadpool to taskpool +- safe_asterisk: Fix logging and sorting issue. +- Fix Endianness detection in utils.h for non-Linux +- app_queue.c: Fix error in Queue parameter documentation. +- devicestate: Don't publish redundant device state messages. +- chan_pjsip: Add technology-specific off-nominal hangup cause to events. +- res_audiosocket: add message types for all slin sample rates +- res_fax.c: lower FAXOPT read warning to debug level +- endpoints: Remove need for stasis subscription. +- app_queue: Allow stasis message filtering to work. +- taskpool: Fix some references to threadpool that should be taskpool. +- Update contact information for anthm +- chan_websocket.c: Change payload references to command instead. +- func_callerid: Document limitation of DNID fields. +- func_channel: Allow R/W of ADSI CPE capability setting. +- core_unreal: Preserve ADSI capability when dialing Local channels. +- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE() +- sig_analog: Allow '#' to end the inter-digit timeout when dialing. +- func_math: Add DIGIT_SUM function. +- app_sf: Add post-digit timer option to ReceiveSF. +- codec_builtin.c: Adjust some of the quality scores to reflect reality. +- res_tonedetect: Fix formatting of XML documentation. +- res_fax: Add XML documentation for channel variables. +- channelstorage_cpp_map_name_id: Add read locking around retrievals. +- app_dial: Allow fractional seconds for dial timeouts. +- dsp.c: Make minor fixes to debug log messages. +- config_options.c: Improve misleading warning. +- func_scramble: Add example to XML documentation. +- sig_analog: Eliminate potential timeout with Last Number Redial. +- ARI: The bridges play and record APIs now handle sample rates > 8K correctly. +- res_pjsip_geolocation: Add support for Geolocation loc-src parameter +- sorcery: Move from threadpool to taskpool. +- 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. +- taskpool: Update versions for taskpool stasis options. +- taskpool: Add taskpool API, switch Stasis to using it. +- 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: + +#### res_audiosocket: fix temporarily unavailable + Author: Roman Pertsev + Date: 2025-10-07 + + Operations on non-blocking sockets may return a resource temporarily unavailable error (EAGAIN or EWOULDBLOCK). This is not a fatal error but a normal condition indicating that the operation would block. + + This patch corrects the handling of this case. Instead of incorrectly treating it as a reason to terminate the connection, the code now waits for data to arrive on the socket. + +#### 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 + +#### res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function. + Author: George Joseph + Date: 2025-10-24 + + Also... + + * Refactored the verification datastore process so instead of having + a separate channel datastore for each verification result, there's only + one channel datastore with a vector of results. + + * Refactored some log messages to include channel name and removed + some that would be redundant if a memory allocation failed. + + Resolves: #781 + + UserNote: The STIR_SHAKEN_ATTESTATION dialplan function has been added + which will allow suppressing attestation on a call-by-call basis + regardless of the profile attached to the outgoing endpoint. + +#### Revert "pjsip: Move from threadpool to taskpool" + Author: Joshua C. Colp + Date: 2025-10-27 + + This reverts commit bb6b76c2d8239b2665223dcbf6d507aa9aa4534e. + +#### iostream.c: Handle TLS handshake attacks in order to resolve the issue of exce.. + Author: Tinet-mucw + Date: 2025-10-26 + + The TCP three-way handshake completes, but if the server is under a TLS handshake attack, asterisk will get stuck at SSL_do_handshake(). + In this case, a timeout mechanism should be set for the SSL/TLS handshake process to prevent indefinite waiting during the SSL handshake. + + Resolves: #1559 + +#### chan_pjsip: Disable SSRC change for WebRTC endpoints. + Author: George Joseph + Date: 2025-10-21 + + Commit b333ee3b introduced a fix to chan_pjsip that addressed RTP issues with + blind transfers and some SBCs. Unfortunately, the fix broke some WebRTC + clients that are sensitive to SSRC changes and non-monotonic timestamps so + the fix is now disabled for endpoints with the "bundle" parameter set to true. + + Resolves: #1535 + +#### chan_websocket: Add channel_id to MEDIA_START, DRIVER_STATUS and DTMF_END even.. + Author: gauravs456 + Date: 2025-10-21 + + Resolves: #1544 + +#### pjsip: Move from threadpool to taskpool + Author: Joshua C. Colp + Date: 2025-09-23 + + This change moves the PJSIP module from the threadpool API + to the taskpool API. PJSIP-specific implementations for + task usage have been removed and replaced with calls to + the optimized taskpool implementations instead. The need + for a pool of serializers has also been removed as + taskpool inherently provides this. The default settings + have also been changed to be more realistic for common + usage. + + UpgradeNote: The threadpool_* options in pjsip.conf have now + been deprecated though they continue to be read and used. + They have been replaced with taskpool options that give greater + control over the underlying taskpool used for PJSIP. An alembic + upgrade script has been added to add these options to realtime + as well. + +#### 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 + +#### Fix Endianness detection in utils.h for non-Linux + Author: Christoph Moench-Tegeder + Date: 2025-10-19 + + Commit 43bf8a4ded7a65203b766b91eaf8331a600e9d8d introduced endian + dependend byte-swapping code in include/asterisk/utils.h, where the + endianness was detected using the __BYTE_ORDER macro. This macro + lives in endian.h, which on Linux is included implicitely (by the + network-related headers, I think), but on FreeBSD the headers are + laid out differently and we do not get __BYTE_ORDER the implicit way. + + Instead, this makes the usage of endian.h explicit by including it + where we need it, and switches the BYTE_ORDER/*ENDIAN macros to the + POSIX-defined ones (see + https://pubs.opengroup.org/onlinepubs/9799919799/basedefs/endian.h.html + for standard compliance). Additionally, this adds a compile-time check + for the endianness-logic: compilation will fail if neither big nor + little endian can be detected. + + Fixes: #1536 + +#### app_queue.c: Fix error in Queue parameter documentation. + Author: Ben Ford + Date: 2025-10-20 + + When macro was removed in Asterisk 21, the parameter documentation in + code was not updated to reflect the correct numerization for gosub. It + still stated that it was the seventh parameter, but got shifted to the + sixth due to the removal of macro. This has been updated to correctly + reflect the parameter order, and a note has been added to the XML that + states this was done after the initial commit. + + Fixes: #1534 + + UpgradeNote: As part of Asterisk 21, macros were removed from Asterisk. + This resulted in argument order changing for the Queue dialplan + application since the macro argument was removed. Upgrade notice was + missed when this was done, so this upgrade note has been added to + provide a record of such and a notice to users who may have not upgraded + yet. + +#### devicestate: Don't publish redundant device state messages. + Author: Joshua C. Colp + Date: 2025-10-17 + + When publishing device state check the local cache for the + existing device state. If the new device state is unchanged + from the prior one, don't bother publishing the update. This + can reduce the work done by consumers of device state, such + as hints and app_queue, by not publishing a message to them. + + These messages would most often occur with devices that are + seeing numerous simultaneous channels. The underlying device + state would remain as in use throughout, but an update would + be published as channels are created and hung up. + +#### chan_pjsip: Add technology-specific off-nominal hangup cause to events. + Author: George Joseph + Date: 2025-10-14 + + Although the ISDN/Q.850/Q.931 hangup cause code is already part of the ARI + and AMI hangup and channel destroyed events, it can be helpful to know what + the actual channel technology code was if the call was unsuccessful. + For PJSIP, it's the SIP response code. + + * A new "tech_hangupcause" field was added to the ast_channel structure along + with ast_channel_tech_hangupcause() and ast_channel_tech_hangupcause_set() + functions. It should only be set for off-nominal terminations. + + * chan_pjsip was modified to set the tech hangup cause in the + chan_pjsip_hangup() and chan_pjsip_session_end() functions. This is a bit + tricky because these two functions aren't always called in the same order. + The channel that hangs up first will get chan_pjsip_session_end() called + first which will trigger the core to call chan_pjsip_hangup() on itself, + then call chan_pjsip_hangup() on the other channel. The other channel's + chan_pjsip_session_end() function will get called last. Unfortunately, + the other channel's HangupRequest events are sent before chan_pjsip has had a + chance to set the tech hangupcause code so the HangupRequest events for that + channel won't have the cause code set. The ChannelDestroyed and Hangup + events however will have the code set for both channels. + + * A new "tech_cause" field was added to the ast_channel_snapshot_hangup + structure. This is a public structure so a bit of refactoring was needed to + preserve ABI compatibility. + + * The ARI ChannelHangupRequest and ChannelDestroyed events were modified to + include the "tech_cause" parameter in the JSON for off-nominal terminations. + The parameter is suppressed for nominal termination. + + * The AMI SoftHangupRequest, HangupRequest and Hangup events were modified to + include the "TechCause" parameter for off-nominal terminations. Like their ARI + counterparts, the parameter is suppressed for nominal termination. + + DeveloperNote: A "tech_cause" parameter has been added to the + ChannelHangupRequest and ChannelDestroyed ARI event messages and a "TechCause" + parameter has been added to the HangupRequest, SoftHangupRequest and Hangup + AMI event messages. For chan_pjsip, these will be set to the last SIP + response status code for off-nominally terminated calls. The parameter is + suppressed for nominal termination. + +#### 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. + +#### res_fax.c: lower FAXOPT read warning to debug level + Author: phoneben + Date: 2025-10-03 + + Reading ${FAXOPT()} before a fax session is common in dialplans to check fax state. + Currently this logs an error even when no fax datastore exists, creating excessive noise. + Change these messages to ast_debug(3, …) so they appear only with debug enabled. + + Resolves: #1509 + +#### endpoints: Remove need for stasis subscription. + Author: Joshua C. Colp + Date: 2025-10-10 + + When an endpoint is created in the core of Asterisk a subscription + was previously created alongside it to monitor any channels being + destroyed that were related to it. This was done by receiving all + channel snapshot updates for every channel and only reacting when + it was indicated that the channel was dead. + + This change removes this logic and instead provides an API call + for directly removing a channel from an endpoint. This is called + when channels are destroyed. This operation is fast, so blocking + the calling thread for a short period of time doesn't have any + noticeable impact. + +#### app_queue: Allow stasis message filtering to work. + Author: Joshua C. Colp + Date: 2025-10-10 + + The app_queue module subscribes on a per-dialed agent basis to both + the bridge all and channel all topics to keep apprised of things going + on involving them. This subscription has associated state that must + be cleaned up when the subscription ends. This was done by setting + a default router callback that only had logic to handle the case + where the subscription ends. By using the default router callback + all filtering for the subscription was disabled, causing unrelated + messages to get published and handled by it. + + This change makes it so that an explicit route is added for the + message type used for the message indicating the subscription has + ended and removes the default router callback. This allows message + filtering to occur on publishing reducing the messages to app_queue + to only those it is interested in. + +#### taskpool: Fix some references to threadpool that should be taskpool. + Author: George Joseph + Date: 2025-10-10 + + Resolves: #1478 + +#### Update contact information for anthm + Author: Anthony Minessale + Date: 2025-10-10 + + +#### 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_callerid: Document limitation of DNID fields. + Author: Naveen Albert + Date: 2025-10-06 + + The Dial() application does not propagate DNID fields, which is counter + to the behavior of the other Caller ID fields. This behavior is likely + intentional since the use of Dial theoretically suggests a new dialed + number, but document this caveat to inform users of it. + + Resolves: #1519 + +#### func_channel: Allow R/W of ADSI CPE capability setting. + Author: Naveen Albert + Date: 2025-10-06 + + Allow retrieving and setting the channel's ADSI capability from the + dialplan. + + Resolves: #1514 + + UserNote: CHANNEL(adsicpe) can now be read or written to change + the channels' ADSI CPE capability setting. + +#### core_unreal: Preserve ADSI capability when dialing Local channels. + Author: Naveen Albert + Date: 2025-10-06 + + Dial() already preserves the ADSI capability by copying it to the new + channel, but since Local channel pairs consist of two channels, we + also need to copy the capability to the second channel. + + Resolves: #1517 + +#### 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. + +#### sig_analog: Allow '#' to end the inter-digit timeout when dialing. + Author: Naveen Albert + Date: 2025-10-03 + + It is customary to allow # to terminate digit collection immediately + when there would normally be a timeout. However, currently, users are + forced to wait for the timeout to expire when dialing numbers that + are prefixes of other valid matches, and there is no way to end the + timeout early. Customarily, # terminates the timeout, but at the moment, + this is just rejected unless there happens to be a matching extension + ending in #. + + Allow # to terminate the timeout in cases where there is no dialplan + match. This ensures that the dialplan is always respected, but if a + valid extension has been dialed that happens to prefix other valid + matches, # can be used to dial it immediately. + + Resolves: #1510 + +#### func_math: Add DIGIT_SUM function. + Author: Naveen Albert + Date: 2025-10-01 + + Add a function (DIGIT_SUM) which returns the digit sum of a number. + + Resolves: #1499 + + UserNote: The DIGIT_SUM function can be used to return the digit sum of + a number. + +#### app_sf: Add post-digit timer option to ReceiveSF. + Author: Naveen Albert + Date: 2025-10-01 + + Add a sorely needed option to set a timeout between digits, rather than + for receiving the entire number. This is needed if the number of digits + being sent is unknown by the receiver in advance. Previously, we had + to wait for the entire timer to expire. + + Resolves: #1493 + + UserNote: The 't' option for ReceiveSF now allows for a timer since + the last digit received, in addition to the number-wide timeout. + +#### codec_builtin.c: Adjust some of the quality scores to reflect reality. + Author: Naveen Albert + Date: 2025-10-02 + + Among the lower-quality voice codecs, some of the quality scores did + not make sense relative to each other. + + For instance, quality-wise, G.729 > G.723 > PLC10. + However, current scores do not uphold these relationships. + + Tweak the scores slightly to reflect more accurate relationships. + + Resolves: #1501 + +#### res_tonedetect: Fix formatting of XML documentation. + Author: Naveen Albert + Date: 2025-10-02 + + Fix the indentation in the documentation for the variable list. + + Resolves: #1507 + +#### res_fax: Add XML documentation for channel variables. + Author: Naveen Albert + Date: 2025-10-02 + + Document the channel variables currently set by SendFAX and ReceiveFAX. + + Resolves: #1505 + +#### 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 + +#### app_dial: Allow fractional seconds for dial timeouts. + Author: Naveen Albert + Date: 2025-09-30 + + Even though Dial() internally uses milliseconds for its dial timeouts, + this capability has been mostly obscured from users as the argument is + only parsed as an integer, thus forcing the use of whole seconds for + timeouts. + + Parse it as a decimal instead so that timeouts can now truly have + millisecond precision. + + Resolves: #1487 + + UserNote: The answer and progress dial timeouts now have millisecond + precision, instead of having to be whole numbers. + +#### dsp.c: Make minor fixes to debug log messages. + Author: Naveen Albert + Date: 2025-10-01 + + Commit dc8e3eeaaf094a3d16991289934093d5e7127680 improved the debug log + messages in dsp.c. This makes two minor corrections to it: + + * Properly guard an added log statement in a conditional. + * Don't add one to the hit count if there was no hit (however, we do + still want to do this for the case where this is one). + + Resolves: #1496 + +#### config_options.c: Improve misleading warning. + Author: Naveen Albert + Date: 2025-09-30 + + When running "config show help ", if no XML documentation exists + for the specified module, "Module not found." is returned, + which is misleading if the module is loaded but simply has no XML + documentation for its config. Improve the message to clarify that the + module may simply have no config documentation. + + Resolves: #1489 + +#### func_scramble: Add example to XML documentation. + Author: Naveen Albert + Date: 2025-09-29 + + The previous lack of an example made it ambiguous if the arguments went + inside the function arguments or were part of the right-hand value. + + Resolves: #1485 + +#### sig_analog: Eliminate potential timeout with Last Number Redial. + Author: Naveen Albert + Date: 2025-09-29 + + If Last Number Redial is used to redial, ensure that we do not wait + for further digits. This was possible if the number that was last + dialed is a prefix of another possible dialplan match. Since all we + did is copy the number into the extension buffer, if other matches + are now possible, there would thus be a timeout before the call went + through. We now complete redialed calls immediaetly in all cases. + + Resolves: #1483 + +#### ARI: The bridges play and record APIs now handle sample rates > 8K correctly. + Author: George Joseph + Date: 2025-09-25 + + The bridge play and record APIs were forcing the Announcer/Recorder channel + to slin8 which meant that if you played or recorded audio with a sample + rate > 8K, it was downsampled to 8K limiting the bandwidth. + + * The /bridges/play REST APIs have a new "announcer_format" parameter that + allows the caller to explicitly set the format on the "Announcer" channel + through which the audio is played into the bridge. If not specified, the + default depends on how many channels are currently in the bridge. If + a single channel is in the bridge, then the Announcer channel's format + will be set to the same as that channel's. If multiple channels are in the + bridge, the channels will be scanned to find the one with the highest + sample rate and the Announcer channel's format will be set to the slin + format that has an equal to or greater than sample rate. + + * The /bridges/record REST API has a new "recorder_format" parameter that + allows the caller to explicitly set the format on the "Recorder" channel + from which audio is retrieved to write to the file. If not specified, + the Recorder channel's format will be set to the format that was requested + to save the audio in. + + Resolves: #1479 + + DeveloperNote: The ARI /bridges/play and /bridges/record REST APIs have new + parameters that allow the caller to specify the format to be used on the + "Announcer" and "Recorder" channels respecitvely. + +#### 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. + +#### sorcery: Move from threadpool to taskpool. + Author: Joshua C. Colp + Date: 2025-09-23 + + This change moves observer invocation from the use of + a threadpool to a taskpool. The taskpool options have also + been adjusted to ensure that at least one taskprocessor + remains available at all times. + +#### 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. + +#### taskpool: Update versions for taskpool stasis options. + Author: Joshua C. Colp + Date: 2025-09-16 + + +#### taskpool: Add taskpool API, switch Stasis to using it. + Author: Joshua C. Colp + Date: 2025-08-06 + + This change introduces a new API called taskpool. This is a pool + of taskprocessors. It provides the following functionality: + + 1. Task pushing to a pool of taskprocessors + 2. Synchronous tasks + 3. Serializers for execution ordering of tasks + 4. Growing/shrinking of number of taskprocessors in pool + + This functionality already exists through the combination of + threadpool+taskprocessors but through investigating I determined + that this carries substantial overhead for short to medium duration + tasks. The threadpool uses a single queue of work, and for management + of threads it involves additional tasks. + + I wrote taskpool to eliminate the extra overhead and management + as much as possible. Instead of a single queue of work each + taskprocessor has its own queue and at push time a selector chooses + the taskprocessor to queue the task to. Each taskprocessor also + has its own thread like normal. This spreads out the tasks immediately + and reduces contention on shared resources. + + Using the included efficiency tests the number of tasks that can be + executed per second in a taskpool is 6-12 times more than an equivalent + threadpool+taskprocessor setup. + + Stasis has been moved over to using this new API as it is a heavy consumer + of threadpool+taskprocessors and produces a lot of tasks. + + UpgradeNote: The threadpool_* options in stasis.conf have now been deprecated + though they continue to be read and used. They have been replaced with taskpool + options that give greater control over the underlying taskpool used for stasis. + + DeveloperNote: The taskpool API has been added for common usage of a + pool of taskprocessors. It is suggested to use this API instead of the + threadpool+taskprocessor approach. + +#### 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 7241dc9533..8b17449690 100644 --- a/README.html +++ b/README.html @@ -1,4 +1,4 @@ -Readme for asterisk-22.6.0 +Readme for asterisk-22.7.0-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 5495ad98dc..f9a8459e98 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-22.6.0.html)
+[Change Logs](ChangeLogs/ChangeLog-22.7.0-rc1.html)
 
 
 ### NEW INSTALLATIONS