From 6fed73515728d0d74b7f3045c9d44f023b3e6c47 Mon Sep 17 00:00:00 2001 From: Asterisk Development Team Date: Thu, 20 Mar 2025 18:34:12 +0000 Subject: [PATCH] Update for 22.3.0-rc1 --- .version | 2 +- CHANGES.html | 1 + CHANGES.md | 2 +- ChangeLogs/ChangeLog-22.3.0-rc1.html | 562 ++++++++++++++++++++++++++ ChangeLogs/ChangeLog-22.3.0-rc1.md | 573 +++++++++++++++++++++++++++ README.html | 174 ++++++++ README.md | 2 +- 7 files changed, 1313 insertions(+), 3 deletions(-) create mode 120000 CHANGES.html create mode 100644 ChangeLogs/ChangeLog-22.3.0-rc1.html create mode 100644 ChangeLogs/ChangeLog-22.3.0-rc1.md create mode 100644 README.html diff --git a/.version b/.version index 44e031ba9b..23df23f9f2 100644 --- a/.version +++ b/.version @@ -1 +1 @@ -22.2.0 +22.3.0-rc1 diff --git a/CHANGES.html b/CHANGES.html new file mode 120000 index 0000000000..4c929289ff --- /dev/null +++ b/CHANGES.html @@ -0,0 +1 @@ +ChangeLogs/ChangeLog-22.3.0-rc1.html \ No newline at end of file diff --git a/CHANGES.md b/CHANGES.md index a8523e56cc..5d1fd63d36 120000 --- a/CHANGES.md +++ b/CHANGES.md @@ -1 +1 @@ -ChangeLogs/ChangeLog-22.2.0.md \ No newline at end of file +ChangeLogs/ChangeLog-22.3.0-rc1.md \ No newline at end of file diff --git a/ChangeLogs/ChangeLog-22.3.0-rc1.html b/ChangeLogs/ChangeLog-22.3.0-rc1.html new file mode 100644 index 0000000000..82c6f93bbf --- /dev/null +++ b/ChangeLogs/ChangeLog-22.3.0-rc1.html @@ -0,0 +1,562 @@ +ChangeLog for asterisk-22.3.0-rc1 +

Change Log for Release asterisk-22.3.0-rc1

+

Links:

+ +

Summary:

+ +

User Notes:

+ +

Upgrade Notes:

+

Commit Authors:

+ +

Issue and Commit Detail:

+

Closed Issues:

+ +

Commits By Author:

+ +

Commit List:

+ +

Commit Details:

+

documentation: Update Gosub, Goto, and add new documentationtype.

+

Author: Ben Ford + Date: 2025-03-14

+

Gosub and Goto were not displaying their syntax correctly on the docs + site. This change adds a new way to specify an optional context, an + optional extension, and a required priority that the xml stylesheet can + parse without having to know which optional parameters come in which + order. In Asterisk, it looks like this:

+
parameter name="context" documentationtype="dialplan_context"
+parameter name="extension" documentationtype="dialplan_extension"
+parameter name="priority" documentationtype="dialplan_priority" required="true"
+
+

The stylesheet will ignore the context and extension parameters, but for + priority, it will automatically inject the following:

+
[[context,]extension,]priority
+
+

This is the correct oder for applications such as Gosub and Goto.

+

res_config_curl.c: Remove unnecessary warnings.

+

Author: Sean Bright + Date: 2025-03-17

+

Resolves: #1164

+

README.md: Updates and Fixes

+

Author: George Joseph + Date: 2025-03-05

+ +

Going forward, the release process will create HTML versions of the README + and change log and will update the link in the README to the current + change log for the branch...

+ +

Resolves: #1131

+

res_rtp_asterisk.c: Don't truncate spec-compliant ice-ufrag or ice-pwd.

+

Author: Sean Bright + Date: 2025-03-07

+

RFC 8839[1] indicates that the ice-ufrag and ice-pwd attributes + can be up to 256 bytes long. While we don't generate values of that + size, we should be able to accomodate them without truncating.

+
    +
  1. https://www.rfc-editor.org/rfc/rfc8839#name-ice-ufrag-and-ice-pwd-attri
    2. +
+

fix: Correct default flag for tcp_keepalive_enable option

+

Author: Joshua Elson + Date: 2025-03-06

+

Resolves an issue where the tcp_keepalive_enable option was not properly enabled in the sample configuration due to an incorrect default flag setting.

+

Fixes: #1149

+

docs: AMI documentation fixes.

+

Author: Sean Bright + Date: 2025-02-18

+

Most of this patch is adding missing PJSIP-related event + documentation, but the one functional change was adding a sorcery + to-string handler for endpoint's redirect_method which was not + showing up in the AMI event details or pjsip show endpoint + <endpoint> output.

+

The rest of the changes are summarized below:

+ +

config.c: #include of non-existent file should not crash

+

Author: Allan Nathanson + Date: 2025-03-03

+

Corrects a segmentation fault when a configuration file has a #include + statement that referenced a file that does not exist.

+

Resolves: https://github.com/asterisk/asterisk/issues/1139

+

manager.c: Check for restricted file in action_createconfig.

+

Author: George Joseph + Date: 2025-03-03

+

The CreateConfig manager action now ensures that a config file can + only be created in the AST_CONFIG_DIR unless live_dangerously is set.

+

Resolves: #1122

+

swagger_model.py: Fix invalid escape sequence in get_list_parameter_type().

+

Author: George Joseph + Date: 2025-03-04

+

Recent python versions complain when backslashes in strings create invalid + escape sequences. This causes issues for strings used as regex patterns like + '^List\[(.*)\]$' where you want the regex parser to treat [ and ] + as literals. Double-backslashing is one way to fix it but simply converting + the string to a raw string re.match(r'^List\[(.*)\]$', text) is easier + and less error prone.

+

Revert "res_rtp_asterisk.c: Set Mark on rtp when timestamp skew is too big"

+

Author: Maximilian Fridrich + Date: 2025-02-28

+

This reverts commit f30ad96b3f467739c38ff415e80bffc4afff1da7.

+

The original change was not RFC compliant and caused issues because it + set the RTP marker bit in cases when it shouldn't be set. See the + linked issue #1135 for a detailed explanation.

+

Fixes: #1135.

+

res_rtp_asterisk.c: Use correct timeout value for T.140 RED timer.

+

Author: Sean Bright + Date: 2025-02-24

+

Found while reviewing #1128

+

docs: Fix typos in cdr/ Found via codespell

+

Author: Luz Paz + Date: 2025-02-12

+

docs: Fix various typos in channels/ Found via `codespell -q 3 -S "./CREDITS,*..

+

Author: Luz Paz + Date: 2025-02-04

+

docs: Fix various typos in main/ Found via `codespell -q 3 -S "./CREDITS" -L a..

+

Author: Luz Paz + Date: 2025-02-04

+

bridging: Fix multiple bridging issues causing SEGVs and FRACKs.

+

Author: George Joseph + Date: 2025-01-22

+

Issues:

+ +

Changes:

+ +

Resolves: #211

+

bridge_channel: don't set cause code on channel during bridge delete if alread..

+

Author: Mike Bradeen + Date: 2025-02-18

+

Due to a potential race condition via ARI when hanging up a channel hangup with cause + while also deleting a bridge containing that channel, the bridge delete can over-write + the hangup cause code resulting in Normal Call Clearing instead of the set value.

+

With this change, bridge deletion will only set the hangup code if it hasn't been + previously set.

+

Resolves: #1124

+

res_config_pgsql: Fix regression that removed dbname config.

+

Author: George Joseph + Date: 2025-02-11

+

A recent commit accidentally removed the code that sets dbname. + This commit adds it back in.

+

Resolves: #1119

+

res_stir_shaken: Allow missing or anonymous CID to continue to the dialplan.

+

Author: George Joseph + Date: 2025-02-05

+

The verification check for missing or anonymous callerid was happening before + the endpoint's profile was retrieved which meant that the failure_action + parameter wasn't available. Therefore, if verification was enabled and there + was no callerid or it was "anonymous", the call was immediately terminated + instead of giving the dialplan the ability to decide what to do with the call.

+ +

Resolves: #1112

+

resource_channels.c: Fix memory leak in ast_ari_channels_external_media.

+

Author: George Joseph + Date: 2025-02-04

+

Between ast_ari_channels_external_media(), external_media_rtp_udp(), + and external_media_audiosocket_tcp(), the variables structure being passed + around wasn't being cleaned up properly when there was a failure.

+ +

Resolves: #1109

+

ari/pjsip: Make it possible to control transfers through ARI

+

Author: Holger Hans Peter Freyther + Date: 2024-06-15

+

Introduce a ChannelTransfer event and the ability to notify progress to + ARI. Implement emitting this event from the PJSIP channel instead of + handling the transfer in Asterisk when configured.

+

Introduce a dialplan function to the PJSIP channel to switch between the + "core" and "ari-only" behavior.

+

UserNote: Call transfers on the PJSIP channel can now be controlled by + ARI. This can be enabled by using the PJSIP_TRANSFER_HANDLING(ari-only) + dialplan function.

+

channel.c: Remove dead AST_GENERATOR_FD code.

+

Author: Sean Bright + Date: 2025-02-06

+

Nothing ever sets the AST_GENERATOR_FD, so this block of code will + never execute. It also is the only place where the generate callback + is called with the channel lock held which made it difficult to reason + about the thread safety of ast_generators.

+

In passing, also note that AST_AGENT_FD isn't used either.

+

func_strings.c: Prevent SEGV in HASH single-argument mode.

+

Author: George Joseph + Date: 2025-01-30

+

When in single-argument mode (very rarely used), a malformation of a column + name (also very rare) could cause a NULL to be returned when retrieving the + channel variable for that column. Passing that to strncat causes a SEGV. We + now check for the NULL and print a warning message.

+

Resolves: #1101

+

docs: Add version information to AGI command XML elements.

+

Author: George Joseph + Date: 2025-01-24

+

This process was a bit different than the others because everything + is in the same file, there's an array that contains the command + names and their handler functions, and the last command was created + over 15 years ago.

+ +

docs: Fix minor typo in MixMonitor AMI action

+

Author: Jeremy Lainé + Date: 2025-01-28

+

The Options argument was erroneously documented as lowercase + options.

+

utils: Disable old style definition warnings for libdb.

+

Author: Naveen Albert + Date: 2025-01-23

+

Newer versions of gcc now warn about old style definitions, such + as those in libdb, which causes compilation failure with DEVMODE + enabled. Ignore these warnings for libdb.

+

Resolves: #1085

+

rtp.conf.sample: Correct stunaddr example.

+

Author: fabriziopicconi + Date: 2024-09-25

+

docs: Add version information to ARI resources and methods.

+

Author: George Joseph + Date: 2025-01-27

+ +

docs: Indent tags.

+

Author: Sean Bright + Date: 2025-01-23

+

Also updates the 'since' of applications/functions that existed before + XML documentation was introduced (1.6.2.0).

+ diff --git a/ChangeLogs/ChangeLog-22.3.0-rc1.md b/ChangeLogs/ChangeLog-22.3.0-rc1.md new file mode 100644 index 0000000000..fa4cc0e85c --- /dev/null +++ b/ChangeLogs/ChangeLog-22.3.0-rc1.md @@ -0,0 +1,573 @@ + +## Change Log for Release asterisk-22.3.0-rc1 + +### Links: + + - [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-22.3.0-rc1.html) + - [GitHub Diff](https://github.com/asterisk/asterisk/compare/22.2.0...22.3.0-rc1) + - [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-22.3.0-rc1.tar.gz) + - [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) + +### Summary: + +- Commits: 28 +- Commit Authors: 12 +- Issues Resolved: 12 +- Security Advisories Resolved: 0 + +### User Notes: + +- #### ari/pjsip: Make it possible to control transfers through ARI + Call transfers on the PJSIP channel can now be controlled by + ARI. This can be enabled by using the PJSIP_TRANSFER_HANDLING(ari-only) + dialplan function. + + +### Upgrade Notes: + + +### Commit Authors: + +- Allan Nathanson: (1) +- Ben Ford: (1) +- Fabriziopicconi: (1) +- George Joseph: (10) +- Holger Hans Peter Freyther: (1) +- Jeremy Lainé: (1) +- Joshua Elson: (1) +- Luz Paz: (3) +- Maximilian Fridrich: (1) +- Mike Bradeen: (1) +- Naveen Albert: (1) +- Sean Bright: (6) + +## Issue and Commit Detail: + +### Closed Issues: + + - 211: [bug]: stasis: Off-nominal channel leave causes bridge to be destroyed + - 1085: [bug]: utils: Compilation failure with DEVMODE due to old-style definitions + - 1101: [bug]: when setting a var with a double quotes and using Set(HASH) + - 1109: [bug]: Off nominal memory leak in res/ari/resource_channels.c + - 1112: [bug]: STIR/SHAKEN verification doesn't allow anonymous callerid to be passed to the dialplan. + - 1119: [bug]: Realtime database not working after upgrade from 22.0.0 to 22.2.0 + - 1122: Need status on CVE-2024-57520 claim. + - 1124: [bug]: Race condition between bridge and channel delete can over-write cause code set in hangup. + - 1131: [bug]: CHANGES link broken in README.md + - 1135: [bug]: Problems with video decoding due to RTP marker bit set + - 1149: [bug]: res_pjsip: Mismatch in tcp_keepalive_enable causes not to enable + - 1164: [bug]: WARNING Message in messages.log for res_curl.conf [globals] + +### Commits By Author: + +- #### Allan Nathanson (1): + - config.c: #include of non-existent file should not crash + +- #### Ben Ford (1): + - documentation: Update Gosub, Goto, and add new documentationtype. + +- #### George Joseph (10): + - docs: Add version information to ARI resources and methods. + - docs: Add version information to AGI command XML elements. + - func_strings.c: Prevent SEGV in HASH single-argument mode. + - resource_channels.c: Fix memory leak in ast_ari_channels_external_media. + - res_stir_shaken: Allow missing or anonymous CID to continue to the dialplan. + - res_config_pgsql: Fix regression that removed dbname config. + - bridging: Fix multiple bridging issues causing SEGVs and FRACKs. + - swagger_model.py: Fix invalid escape sequence in get_list_parameter_type(). + - manager.c: Check for restricted file in action_createconfig. + - README.md: Updates and Fixes + +- #### Holger Hans Peter Freyther (1): + - ari/pjsip: Make it possible to control transfers through ARI + +- #### Jeremy Lainé (1): + - docs: Fix minor typo in MixMonitor AMI action + +- #### Joshua Elson (1): + - fix: Correct default flag for tcp_keepalive_enable option + +- #### Luz Paz (3): + - docs: Fix various typos in main/ Found via `codespell -q 3 -S "./CREDITS" -L a.. + - docs: Fix various typos in channels/ Found via `codespell -q 3 -S "./CREDITS,*.. + - docs: Fix typos in cdr/ Found via codespell + +- #### Maximilian Fridrich (1): + - Revert "res_rtp_asterisk.c: Set Mark on rtp when timestamp skew is too big" + +- #### Mike Bradeen (1): + - bridge_channel: don't set cause code on channel during bridge delete if alread.. + +- #### Naveen Albert (1): + - utils: Disable old style definition warnings for libdb. + +- #### Sean Bright (6): + - docs: Indent tags. + - channel.c: Remove dead AST_GENERATOR_FD code. + - res_rtp_asterisk.c: Use correct timeout value for T.140 RED timer. + - docs: AMI documentation fixes. + - res_rtp_asterisk.c: Don't truncate spec-compliant `ice-ufrag` or `ice-pwd`. + - res_config_curl.c: Remove unnecessary warnings. + +- #### fabriziopicconi (1): + - rtp.conf.sample: Correct stunaddr example. + + +### Commit List: + +- documentation: Update Gosub, Goto, and add new documentationtype. +- res_config_curl.c: Remove unnecessary warnings. +- README.md: Updates and Fixes +- res_rtp_asterisk.c: Don't truncate spec-compliant `ice-ufrag` or `ice-pwd`. +- fix: Correct default flag for tcp_keepalive_enable option +- docs: AMI documentation fixes. +- config.c: #include of non-existent file should not crash +- manager.c: Check for restricted file in action_createconfig. +- swagger_model.py: Fix invalid escape sequence in get_list_parameter_type(). +- Revert "res_rtp_asterisk.c: Set Mark on rtp when timestamp skew is too big" +- res_rtp_asterisk.c: Use correct timeout value for T.140 RED timer. +- docs: Fix typos in cdr/ Found via codespell +- bridging: Fix multiple bridging issues causing SEGVs and FRACKs. +- res_config_pgsql: Fix regression that removed dbname config. +- res_stir_shaken: Allow missing or anonymous CID to continue to the dialplan. +- resource_channels.c: Fix memory leak in ast_ari_channels_external_media. +- ari/pjsip: Make it possible to control transfers through ARI +- channel.c: Remove dead AST_GENERATOR_FD code. +- func_strings.c: Prevent SEGV in HASH single-argument mode. +- docs: Add version information to AGI command XML elements. +- docs: Fix minor typo in MixMonitor AMI action +- utils: Disable old style definition warnings for libdb. +- rtp.conf.sample: Correct stunaddr example. +- docs: Add version information to ARI resources and methods. +- docs: Indent tags. + +### Commit Details: + +#### documentation: Update Gosub, Goto, and add new documentationtype. + Author: Ben Ford + Date: 2025-03-14 + + Gosub and Goto were not displaying their syntax correctly on the docs + site. This change adds a new way to specify an optional context, an + optional extension, and a required priority that the xml stylesheet can + parse without having to know which optional parameters come in which + order. In Asterisk, it looks like this: + + parameter name="context" documentationtype="dialplan_context" + parameter name="extension" documentationtype="dialplan_extension" + parameter name="priority" documentationtype="dialplan_priority" required="true" + + The stylesheet will ignore the context and extension parameters, but for + priority, it will automatically inject the following: + + [[context,]extension,]priority + + This is the correct oder for applications such as Gosub and Goto. + + +#### res_config_curl.c: Remove unnecessary warnings. + Author: Sean Bright + Date: 2025-03-17 + + Resolves: #1164 + +#### README.md: Updates and Fixes + Author: George Joseph + Date: 2025-03-05 + + * Outdated information has been removed. + * New links added. + * Placeholder added for link to change logs. + + Going forward, the release process will create HTML versions of the README + and change log and will update the link in the README to the current + change log for the branch... + + * In the development branches, the link will always point to the current + release on GitHub. + * In the "releases/*" branches and the tarballs, the link will point to the + ChangeLogs/ChangeLog-.html file in the source directory. + * On the downloads website, the link will point to the + ChangeLog-.html file in the same directory. + + Resolves: #1131 + +#### res_rtp_asterisk.c: Don't truncate spec-compliant `ice-ufrag` or `ice-pwd`. + Author: Sean Bright + Date: 2025-03-07 + + RFC 8839[1] indicates that the `ice-ufrag` and `ice-pwd` attributes + can be up to 256 bytes long. While we don't generate values of that + size, we should be able to accomodate them without truncating. + + 1. https://www.rfc-editor.org/rfc/rfc8839#name-ice-ufrag-and-ice-pwd-attri + + +#### fix: Correct default flag for tcp_keepalive_enable option + Author: Joshua Elson + Date: 2025-03-06 + + Resolves an issue where the tcp_keepalive_enable option was not properly enabled in the sample configuration due to an incorrect default flag setting. + + Fixes: #1149 + +#### docs: AMI documentation fixes. + Author: Sean Bright + Date: 2025-02-18 + + Most of this patch is adding missing PJSIP-related event + documentation, but the one functional change was adding a sorcery + to-string handler for endpoint's `redirect_method` which was not + showing up in the AMI event details or `pjsip show endpoint + ` output. + + The rest of the changes are summarized below: + + * app_agent_pool.c: Typo fix Epoche -> Epoch. + * stasis_bridges.c: Add missing AttendedTransfer properties. + * stasis_channels.c: Add missing AgentLogoff properties. + * pjsip_manager.xml: + - Add missing AorList properties. + - Add missing AorDetail properties. + - Add missing ContactList properties. + - Add missing ContactStatusDetail properties. + - Add missing EventDetail properties. + - Add missing AuthList properties. + - Add missing AuthDetail properties. + - Add missing TransportDetail properties. + - Add missing EndpointList properties. + - Add missing IdentifyDetail properties. + * res_pjsip_registrar.c: Add missing InboundRegistrationDetail documentation. + * res_pjsip_pubsub.c: + - Add missing ResourceListDetail documentation. + - Add missing InboundSubscriptionDetail documentation. + - Add missing OutboundSubscriptionDetail documentation. + * res_pjsip_outbound_registration.c: Add missing OutboundRegistrationDetail documentation. + + +#### config.c: #include of non-existent file should not crash + Author: Allan Nathanson + Date: 2025-03-03 + + Corrects a segmentation fault when a configuration file has a #include + statement that referenced a file that does not exist. + + Resolves: https://github.com/asterisk/asterisk/issues/1139 + +#### manager.c: Check for restricted file in action_createconfig. + Author: George Joseph + Date: 2025-03-03 + + The `CreateConfig` manager action now ensures that a config file can + only be created in the AST_CONFIG_DIR unless `live_dangerously` is set. + + Resolves: #1122 + +#### swagger_model.py: Fix invalid escape sequence in get_list_parameter_type(). + Author: George Joseph + Date: 2025-03-04 + + Recent python versions complain when backslashes in strings create invalid + escape sequences. This causes issues for strings used as regex patterns like + `'^List\[(.*)\]$'` where you want the regex parser to treat `[` and `]` + as literals. Double-backslashing is one way to fix it but simply converting + the string to a raw string `re.match(r'^List\[(.*)\]$', text)` is easier + and less error prone. + + +#### Revert "res_rtp_asterisk.c: Set Mark on rtp when timestamp skew is too big" + Author: Maximilian Fridrich + Date: 2025-02-28 + + This reverts commit f30ad96b3f467739c38ff415e80bffc4afff1da7. + + The original change was not RFC compliant and caused issues because it + set the RTP marker bit in cases when it shouldn't be set. See the + linked issue #1135 for a detailed explanation. + + Fixes: #1135. + +#### res_rtp_asterisk.c: Use correct timeout value for T.140 RED timer. + Author: Sean Bright + Date: 2025-02-24 + + Found while reviewing #1128 + + +#### docs: Fix typos in cdr/ Found via codespell + Author: Luz Paz + Date: 2025-02-12 + + +#### docs: Fix various typos in channels/ Found via `codespell -q 3 -S "./CREDITS,*.. + Author: Luz Paz + Date: 2025-02-04 + + +#### docs: Fix various typos in main/ Found via `codespell -q 3 -S "./CREDITS" -L a.. + Author: Luz Paz + Date: 2025-02-04 + + +#### bridging: Fix multiple bridging issues causing SEGVs and FRACKs. + Author: George Joseph + Date: 2025-01-22 + + Issues: + + * The bridging core allowed multiple bridges to be created with the same + unique bridgeId at the same time. Only the last bridge created with the + duplicate name was actually saved to the core bridges container. + + * The bridging core was creating a stasis topic for the bridge and saving it + in the bridge->topic field but not increasing its reference count. In the + case where two bridges were created with the same uniqueid (which is also + the topic name), the second bridge would get the _existing_ topic the first + bridge created. When the first bridge was destroyed, it would take the + topic with it so when the second bridge attempted to publish a message to + it it either FRACKed or SEGVd. + + * The bridge destructor, which also destroys the bridge topic, is run from the + bridge manager thread not the caller's thread. This makes it possible for + an ARI developer to create a new one with the same uniqueid believing the + old one was destroyed when, in fact, the old one's destructor hadn't + completed. This could cause the new bridge to get the old one's topic just + before the topic was destroyed. When the new bridge attempted to publish + a message on that topic, asterisk could either FRACK or SEGV. + + * The ARI bridges resource also allowed multiple bridges to be created with + the same uniqueid but it kept the duplicate bridges in its app_bridges + container. This created a situation where if you added two bridges with + the same "bridge1" uniqueid, all operations on "bridge1" were performed on + the first bridge created and the second was basically orphaned. If you + attempted to delete what you thought was the second bridge, you actually + deleted the first one created. + + Changes: + + * A new API `ast_bridge_topic_exists(uniqueid)` was created to determine if + a topic already exists for a bridge. + + * `bridge_base_init()` in bridge.c and `ast_ari_bridges_create()` in + resource_bridges.c now call `ast_bridge_topic_exists(uniqueid)` to check + if a bridge with the requested uniqueid already exists and will fail if it + does. + + * `bridge_register()` in bridges.c now checks the core bridges container to + make sure a bridge doesn't already exist with the requested uniqueid. + Although most callers of `bridge_register()` will have already called + `bridge_base_init()`, which will now fail on duplicate bridges, there + is no guarantee of this so we must check again. + + * The core bridges container allocation was changed to reject duplicate + uniqueids instead of silently replacing an existing one. This is a "belt + and suspenders" check. + + * A global mutex was added to bridge.c to prevent concurrent calls to + `bridge_base_init()` and `bridge_register()`. + + * Even though you can no longer create multiple bridges with the same uniqueid + at the same time, it's still possible that the bridge topic might be + destroyed while a second bridge with the same uniqueid was trying to use + it. To address this, the bridging core now increments the reference count + on bridge->topic when a bridge is created and decrements it when the + bridge is destroyed. + + * `bridge_create_common()` in res_stasis.c now checks the stasis app_bridges + container to make sure a bridge with the requested uniqueid doesn't already + exist. This may seem like overkill but there are so many entrypoints to + bridge creation that we need to be safe and catch issues as soon in the + process as possible. + + * The stasis app_bridges container allocation was changed to reject duplicate + uniqueids instead of adding them. This is a "belt and suspenders" check. + + * The `bridge show all` CLI command now shows the bridge name as well as the + bridge id. + + * Response code 409 "Conflict" was added as a possible response from the ARI + bridge create resources to signal that a bridge with the requested uniqueid + already exists. + + * Additional debugging was added to multiple bridging and stasis files. + + Resolves: #211 + +#### bridge_channel: don't set cause code on channel during bridge delete if alread.. + Author: Mike Bradeen + Date: 2025-02-18 + + Due to a potential race condition via ARI when hanging up a channel hangup with cause + while also deleting a bridge containing that channel, the bridge delete can over-write + the hangup cause code resulting in Normal Call Clearing instead of the set value. + + With this change, bridge deletion will only set the hangup code if it hasn't been + previously set. + + Resolves: #1124 + +#### res_config_pgsql: Fix regression that removed dbname config. + Author: George Joseph + Date: 2025-02-11 + + A recent commit accidentally removed the code that sets dbname. + This commit adds it back in. + + Resolves: #1119 + +#### res_stir_shaken: Allow missing or anonymous CID to continue to the dialplan. + Author: George Joseph + Date: 2025-02-05 + + The verification check for missing or anonymous callerid was happening before + the endpoint's profile was retrieved which meant that the failure_action + parameter wasn't available. Therefore, if verification was enabled and there + was no callerid or it was "anonymous", the call was immediately terminated + instead of giving the dialplan the ability to decide what to do with the call. + + * The callerid check now happens after the verification context is created and + the endpoint's stir_shaken_profile is available. + + * The check now processes the callerid failure just as it does for other + verification failures and respects the failure_action parameter. If set + to "continue" or "continue_return_reason", `STIR_SHAKEN(0,verify_result)` + in the dialplan will return "invalid_or_no_callerid". + + * If the endpoint's failure_action is "reject_request", the call will be + rejected with `433 "Anonymity Disallowed"`. + + * If the endpoint's failure_action is "continue_return_reason", the call will + continue but a `Reason: STIR; cause=433; text="Anonymity Disallowed"` + header will be added to the next provisional or final response. + + Resolves: #1112 + +#### resource_channels.c: Fix memory leak in ast_ari_channels_external_media. + Author: George Joseph + Date: 2025-02-04 + + Between ast_ari_channels_external_media(), external_media_rtp_udp(), + and external_media_audiosocket_tcp(), the `variables` structure being passed + around wasn't being cleaned up properly when there was a failure. + + * In ast_ari_channels_external_media(), the `variables` structure is now + defined with RAII_VAR to ensure it always gets cleaned up. + + * The ast_variables_destroy() call was removed from external_media_rtp_udp(). + + * The ast_variables_destroy() call was removed from + external_media_audiosocket_tcp(), its `endpoint` allocation was changed to + to use ast_asprintf() as external_media_rtp_udp() does, and it now + returns an error on failure. + + * ast_ari_channels_external_media() now checks the new return code from + external_media_audiosocket_tcp() and sets the appropriate error response. + + Resolves: #1109 + +#### ari/pjsip: Make it possible to control transfers through ARI + Author: Holger Hans Peter Freyther + Date: 2024-06-15 + + Introduce a ChannelTransfer event and the ability to notify progress to + ARI. Implement emitting this event from the PJSIP channel instead of + handling the transfer in Asterisk when configured. + + Introduce a dialplan function to the PJSIP channel to switch between the + "core" and "ari-only" behavior. + + UserNote: Call transfers on the PJSIP channel can now be controlled by + ARI. This can be enabled by using the PJSIP_TRANSFER_HANDLING(ari-only) + dialplan function. + + +#### channel.c: Remove dead AST_GENERATOR_FD code. + Author: Sean Bright + Date: 2025-02-06 + + Nothing ever sets the `AST_GENERATOR_FD`, so this block of code will + never execute. It also is the only place where the `generate` callback + is called with the channel lock held which made it difficult to reason + about the thread safety of `ast_generator`s. + + In passing, also note that `AST_AGENT_FD` isn't used either. + + +#### func_strings.c: Prevent SEGV in HASH single-argument mode. + Author: George Joseph + Date: 2025-01-30 + + When in single-argument mode (very rarely used), a malformation of a column + name (also very rare) could cause a NULL to be returned when retrieving the + channel variable for that column. Passing that to strncat causes a SEGV. We + now check for the NULL and print a warning message. + + Resolves: #1101 + +#### docs: Add version information to AGI command XML elements. + Author: George Joseph + Date: 2025-01-24 + + This process was a bit different than the others because everything + is in the same file, there's an array that contains the command + names and their handler functions, and the last command was created + over 15 years ago. + + * Dump a `git blame` of res/res_agi.c from BEFORE the handle_* prototypes + were changed. + * Create a command <> handler function xref by parsing the the agi_command + array. + * For each entry, grep the function definition line "static int handle_*" + from the git blame output and capture the commit. This will be the + commit the command was created in. + * Do a `git tag --contains | sort -V | head -1` to get the + tag the function was created in. + * Add a single since/version element to the command XML. Multiple versions + aren't supported here because the branching and tagging scheme changed + several times in the 2000's. + + +#### docs: Fix minor typo in MixMonitor AMI action + Author: Jeremy Lainé + Date: 2025-01-28 + + The `Options` argument was erroneously documented as lowercase + `options`. + + +#### utils: Disable old style definition warnings for libdb. + Author: Naveen Albert + Date: 2025-01-23 + + Newer versions of gcc now warn about old style definitions, such + as those in libdb, which causes compilation failure with DEVMODE + enabled. Ignore these warnings for libdb. + + Resolves: #1085 + +#### rtp.conf.sample: Correct stunaddr example. + Author: fabriziopicconi + Date: 2024-09-25 + + +#### docs: Add version information to ARI resources and methods. + Author: George Joseph + Date: 2025-01-27 + + * Dump a git blame of each file in rest-api/api-docs. + + * Get the commit for each "resourcePath" and "httpMethod" entry. + + * Find the tags for each commit (same as other processes). + + * Insert a "since" array after each "resourcePath" and "httpMethod" entry. + + +#### docs: Indent tags. + Author: Sean Bright + Date: 2025-01-23 + + Also updates the 'since' of applications/functions that existed before + XML documentation was introduced (1.6.2.0). + + diff --git a/README.html b/README.html new file mode 100644 index 0000000000..3c36920d6d --- /dev/null +++ b/README.html @@ -0,0 +1,174 @@ +Readme for asterisk-22.3.0-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.
+
+

SECURITY

+

It is imperative that you read and fully understand the contents of +the security information document before you attempt to configure and run +an Asterisk server.

+

See Important Security Considerations for more information.

+

WHAT IS ASTERISK ?

+

Asterisk is an Open Source PBX and telephony toolkit. It is, in a +sense, middleware between Internet and telephony channels on the bottom, +and Internet and telephony applications at the top. However, Asterisk supports +more telephony interfaces than just Internet telephony. Asterisk also has a +vast amount of support for traditional PSTN telephony, as well.

+

For more information on the project itself, please visit the Asterisk +Home Page and the official +Asterisk Documentation.

+

SUPPORTED OPERATING SYSTEMS

+

Linux

+

The Asterisk Open Source PBX is developed and tested primarily on the +GNU/Linux operating system, and is supported on every major GNU/Linux +distribution.

+

Others

+

Asterisk has also been 'ported' and reportedly runs properly on other +operating systems as well, Apple's Mac OS X, and the BSD variants.

+

GETTING STARTED

+

Most users are using VoIP/SIP exclusively these days but if you need to +interface to TDM or analog services or devices, be sure you've got supported +hardware.

+

Supported telephony hardware includes: +* All Analog and Digital Interface cards from Sangoma +* Any full duplex sound card supported by PortAudio +* The Xorcom Astribank channel bank

+

UPGRADING FROM AN EARLIER VERSION

+

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

+ +

Change Logs

+ + +

NEW INSTALLATIONS

+

Ensure that your system contains a compatible compiler and development +libraries. Asterisk requires either the GNU Compiler Collection (GCC) version +4.1 or higher, or a compiler that supports the C99 specification and some of +the gcc language extensions. In addition, your system needs to have the C +library headers available, and the headers and libraries for ncurses.

+

There are many modules that have additional dependencies. To see what +libraries are being looked for, see ./configure --help, or run +make menuselect to view the dependencies for specific modules.

+

On many distributions, these dependencies are installed by packages with names +like 'glibc-devel', 'ncurses-devel', 'openssl-devel' and 'zlib-devel' +or similar. The contrib/scripts/install_prereq script can be used to install +the dependencies for most Debian and Redhat based Linux distributions. +The script also handles SUSE, Arch, Gentoo, FreeBSD, NetBSD and OpenBSD but +those distributions mightnoit have complete support or they might be out of date.

+

So, let's proceed:

+
    +
  1. +

    Read the documentation.
    +The Asterisk Documentation website has full +information for building, installing, configuring and running Asterisk.

    +
    2. +
  2. +

    Run ./configure
    +Execute the configure script to guess values for system-dependent +variables used during compilation. If the script indicates that some required +components are missing, you can run ./contrib/scripts/install_prereq install +to install the necessary components. Note that this will install all dependencies +for every functionality of Asterisk. After running the script, you will need +to rerun ./configure.

    +
    3. +
  3. +

    Run make menuselect
    +This is needed if you want to select the modules that will be compiled and to +check dependencies for various optional modules.

    +
    4. +
  4. +

    Run make
    +Assuming the build completes successfully:

    +
    5. +
  5. +

    Run make install
    +If this is your first time working with Asterisk, you may wish to install +the sample PBX, with demonstration extensions, etc. If so, run:

    +
    6. +
  6. +

    Run make samples
    +Doing so will overwrite any existing configuration files you have installed.

    +
    7. +
  7. +

    Finally, you can launch Asterisk in the foreground mode (not a daemon) with +asterisk -vvvc
    +You'll see a bunch of verbose messages fly by your screen as Asterisk +initializes (that's the "very very verbose" mode). When it's ready, if +you specified the "c" then you'll get a command line console, that looks +like this:
    +*CLI>
    +You can type core show help at any time to get help with the system. For help +with a specific command, type core show help <command>.

    +
    8. +
+

man asterisk at the Unix/Linux command prompt will give you detailed +information on how to start and stop Asterisk, as well as all the command +line options for starting Asterisk.

+

ABOUT CONFIGURATION FILES

+

All Asterisk configuration files share a common format. Comments are +delimited by ; (since # of course, being a DTMF digit, may occur in +many places). A configuration file is divided into sections whose names +appear in []'s. Each section typically contains statements in the form +variable = value although you may see variable => value in older samples.

+

SPECIAL NOTE ON TIME

+

Those using SIP phones should be aware that Asterisk is sensitive to +large jumps in time. Manually changing the system time using date(1) +(or other similar commands) may cause SIP registrations and other +internal processes to fail. For this reason, you should always use +a time synchronization package to keep your system time accurate. +All OS/distributions make one or more of the following packages +available:

+
    +
  • ntpd/ntpsec
    • +
  • chronyd
    • +
  • systemd-timesyncd
    • +
+

Be sure to install and configure one (and only one) of them.

+

FILE DESCRIPTORS

+

Depending on the size of your system and your configuration, +Asterisk can consume a large number of file descriptors. In UNIX, +file descriptors are used for more than just files on disk. File +descriptors are also used for handling network communication +(e.g. SIP, IAX2, or H.323 calls) and hardware access (e.g. analog and +digital trunk hardware). Asterisk accesses many on-disk files for +everything from configuration information to voicemail storage.

+

Most systems limit the number of file descriptors that Asterisk can +have open at one time. This can limit the number of simultaneous +calls that your system can handle. For example, if the limit is set +at 1024 (a common default value) Asterisk can handle approximately 150 +SIP calls simultaneously. To change the number of file descriptors +follow the instructions for your system below:

+

PAM-BASED LINUX SYSTEM

+

If your system uses PAM (Pluggable Authentication Modules) edit +/etc/security/limits.conf. Add these lines to the bottom of the file:

+
root            soft    nofile          4096
+root            hard    nofile          8196
+asterisk        soft    nofile          4096
+asterisk        hard    nofile          8196
+
+

(adjust the numbers to taste). You may need to reboot the system for +these changes to take effect.

+

GENERIC UNIX SYSTEM

+

If there are no instructions specifically adapted to your system +above you can try adding the command ulimit -n 8192 to the script +that starts Asterisk.

+

MORE INFORMATION

+

Visit the Asterisk Documentation website +for more documentation on various features and please read all the +configuration samples that include documentation on the configuration options.

+

Finally, you may wish to join the +Asterisk Community Forums

+

Welcome to the growing worldwide community of Asterisk users!

+
        Mark Spencer, and the Asterisk.org development community
+
+
+

Asterisk is a trademark of Sangoma Technologies Corporation

+

[Sangoma] +[Home Page] +[Support] +[Documentation] +[Community Forums] +[Release Notes] +[Security] +[Mailing List Archive]

+ diff --git a/README.md b/README.md index 063db527e9..e65182c641 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](https://downloads.asterisk.org/pub/telephony/asterisk) +[Change Logs](ChangeLogs/ChangeLog-22.3.0-rc1.html) ### NEW INSTALLATIONS