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Update for 23.1.0

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2025-11-20 18:13:00 +00:00
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23.1.0-rc2 23.1.0

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<html><head><title>ChangeLog for asterisk-23.1.0-rc2</title></head><body>
<h2>Change Log for Release asterisk-23.1.0-rc2</h2>
<h3>Links:</h3>
<ul>
<li><a href="https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.1.0-rc2.html">Full ChangeLog</a> </li>
<li><a href="https://github.com/asterisk/asterisk/compare/23.1.0-rc1...23.1.0-rc2">GitHub Diff</a> </li>
<li><a href="https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.1.0-rc2.tar.gz">Tarball</a> </li>
<li><a href="https://downloads.asterisk.org/pub/telephony/asterisk">Downloads</a> </li>
</ul>
<h3>Summary:</h3>
<ul>
<li>Commits: 1</li>
<li>Commit Authors: 1</li>
<li>Issues Resolved: 1</li>
<li>Security Advisories Resolved: 0</li>
</ul>
<h3>User Notes:</h3>
<h3>Upgrade Notes:</h3>
<h3>Developer Notes:</h3>
<h3>Commit Authors:</h3>
<ul>
<li>George Joseph: (1)</li>
</ul>
<h2>Issue and Commit Detail:</h2>
<h3>Closed Issues:</h3>
<ul>
<li>1578: [bug]: Deadlock with externalMedia custom channel id and cpp map channel backend</li>
</ul>
<h3>Commits By Author:</h3>
<ul>
<li>
<h4>George Joseph (1):</h4>
</li>
</ul>
<h3>Commit List:</h3>
<ul>
<li>channelstorage: Allow storage driver read locking to be skipped.</li>
</ul>
<h3>Commit Details:</h3>
<h4>channelstorage: Allow storage driver read locking to be skipped.</h4>
<p>Author: George Joseph
Date: 2025-11-06</p>
<p>After PR #1498 added read locking to channelstorage_cpp_map_name_id, if ARI
channels/externalMedia was called with a custom channel id AND the
cpp_map_name_id channel storage backend is in use, a deadlock can occur when
hanging up the channel. It's actually triggered in
channel.c:__ast_channel_alloc_ap() when it gets a write lock on the
channelstorage driver then subsequently does a lookup for channel uniqueid
which now does a read lock. This is an invalid operation and causes the lock
state to get "bad". When the channels try to hang up, a write lock is
attempted again which hangs and causes the deadlock.</p>
<p>Now instead of the cpp_map_name_id channelstorage driver "get" APIs
automatically performing a read lock, they take a "lock" parameter which
allows a caller who already has a write lock to indicate that the "get" API
must not attempt its own lock. This prevents the state from getting mesed up.</p>
<p>The ao2_legacy driver uses the ao2 container's recursive mutex so doesn't
have this issue but since it also implements the common channelstorage API,
it needed its "get" implementations updated to take the lock parameter. They
just don't use it.</p>
<p>Resolves: #1578</p>
</body></html>

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## Change Log for Release asterisk-23.1.0-rc2
### Links:
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.1.0-rc2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.1.0-rc1...23.1.0-rc2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.1.0-rc2.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
### Summary:
- Commits: 1
- Commit Authors: 1
- Issues Resolved: 1
- Security Advisories Resolved: 0
### User Notes:
### Upgrade Notes:
### Developer Notes:
### Commit Authors:
- George Joseph: (1)
## Issue and Commit Detail:
### Closed Issues:
- 1578: [bug]: Deadlock with externalMedia custom channel id and cpp map channel backend
### Commits By Author:
- #### George Joseph (1):
### Commit List:
- channelstorage: Allow storage driver read locking to be skipped.
### Commit Details:
#### channelstorage: Allow storage driver read locking to be skipped.
Author: George Joseph
Date: 2025-11-06
After PR #1498 added read locking to channelstorage_cpp_map_name_id, if ARI
channels/externalMedia was called with a custom channel id AND the
cpp_map_name_id channel storage backend is in use, a deadlock can occur when
hanging up the channel. It's actually triggered in
channel.c:__ast_channel_alloc_ap() when it gets a write lock on the
channelstorage driver then subsequently does a lookup for channel uniqueid
which now does a read lock. This is an invalid operation and causes the lock
state to get "bad". When the channels try to hang up, a write lock is
attempted again which hangs and causes the deadlock.
Now instead of the cpp_map_name_id channelstorage driver "get" APIs
automatically performing a read lock, they take a "lock" parameter which
allows a caller who already has a write lock to indicate that the "get" API
must not attempt its own lock. This prevents the state from getting mesed up.
The ao2_legacy driver uses the ao2 container's recursive mutex so doesn't
have this issue but since it also implements the common channelstorage API,
it needed its "get" implementations updated to take the lock parameter. They
just don't use it.
Resolves: #1578

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@@ -1,17 +1,17 @@
<html><head><title>ChangeLog for asterisk-23.1.0-rc1</title></head><body> <html><head><title>ChangeLog for asterisk-23.1.0</title></head><body>
<h2>Change Log for Release asterisk-23.1.0-rc1</h2> <h2>Change Log for Release asterisk-23.1.0</h2>
<h3>Links:</h3> <h3>Links:</h3>
<ul> <ul>
<li><a href="https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.1.0-rc1.html">Full ChangeLog</a> </li> <li><a href="https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.1.0.html">Full ChangeLog</a> </li>
<li><a href="https://github.com/asterisk/asterisk/compare/23.0.0...23.1.0-rc1">GitHub Diff</a> </li> <li><a href="https://github.com/asterisk/asterisk/compare/23.0.0...23.1.0">GitHub Diff</a> </li>
<li><a href="https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.1.0-rc1.tar.gz">Tarball</a> </li> <li><a href="https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.1.0.tar.gz">Tarball</a> </li>
<li><a href="https://downloads.asterisk.org/pub/telephony/asterisk">Downloads</a> </li> <li><a href="https://downloads.asterisk.org/pub/telephony/asterisk">Downloads</a> </li>
</ul> </ul>
<h3>Summary:</h3> <h3>Summary:</h3>
<ul> <ul>
<li>Commits: 54</li> <li>Commits: 53</li>
<li>Commit Authors: 17</li> <li>Commit Authors: 17</li>
<li>Issues Resolved: 36</li> <li>Issues Resolved: 37</li>
<li>Security Advisories Resolved: 0</li> <li>Security Advisories Resolved: 0</li>
</ul> </ul>
<h3>User Notes:</h3> <h3>User Notes:</h3>
@@ -57,15 +57,6 @@
<h3>Upgrade Notes:</h3> <h3>Upgrade Notes:</h3>
<ul> <ul>
<li> <li>
<h4>pjsip: Move from threadpool to taskpool</h4>
<p>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.</p>
</li>
<li>
<h4>app_queue.c: Fix error in Queue parameter documentation.</h4> <h4>app_queue.c: Fix error in Queue parameter documentation.</h4>
<p>As part of Asterisk 21, macros were removed from Asterisk. <p>As part of Asterisk 21, macros were removed from Asterisk.
This resulted in argument order changing for the Queue dialplan This resulted in argument order changing for the Queue dialplan
@@ -120,18 +111,18 @@
<li>Bastian Triller: (1)</li> <li>Bastian Triller: (1)</li>
<li>Ben Ford: (2)</li> <li>Ben Ford: (2)</li>
<li>Christoph Moench-Tegeder: (1)</li> <li>Christoph Moench-Tegeder: (1)</li>
<li>Gauravs456: (1)</li> <li>George Joseph: (9)</li>
<li>George Joseph: (8)</li>
<li>Igor Goncharovsky: (1)</li> <li>Igor Goncharovsky: (1)</li>
<li>Joshua C. Colp: (8)</li> <li>Joshua C. Colp: (6)</li>
<li>Max Grobecker: (1)</li> <li>Max Grobecker: (1)</li>
<li>Nathan Monfils: (1)</li> <li>Nathan Monfils: (1)</li>
<li>Naveen Albert: (18)</li> <li>Naveen Albert: (18)</li>
<li>Phoneben: (2)</li>
<li>Roman Pertsev: (1)</li> <li>Roman Pertsev: (1)</li>
<li>Sean Bright: (3)</li> <li>Sean Bright: (3)</li>
<li>Sven Kube: (3)</li> <li>Sven Kube: (3)</li>
<li>Tinet-Mucw: (1)</li> <li>Tinet-mucw: (1)</li>
<li>gauravs456: (1)</li>
<li>phoneben: (2)</li>
</ul> </ul>
<h2>Issue and Commit Detail:</h2> <h2>Issue and Commit Detail:</h2>
<h3>Closed Issues:</h3> <h3>Closed Issues:</h3>
@@ -172,6 +163,7 @@
<li>1544: [improvement]: While Receiving the MediaConnect Message Using External Media Over websocket ChannelID is Details are missing</li> <li>1544: [improvement]: While Receiving the MediaConnect Message Using External Media Over websocket ChannelID is Details are missing</li>
<li>1554: [bug]: safe_asterisk recurses into subdirectories of startup.d after f97361</li> <li>1554: [bug]: safe_asterisk recurses into subdirectories of startup.d after f97361</li>
<li>1559: [improvement]: Handle TLS handshake attacks in order to resolve the issue of exceeding the maximum number of HTTPS sessions.</li> <li>1559: [improvement]: Handle TLS handshake attacks in order to resolve the issue of exceeding the maximum number of HTTPS sessions.</li>
<li>1578: [bug]: Deadlock with externalMedia custom channel id and cpp map channel backend</li>
</ul> </ul>
<h3>Commits By Author:</h3> <h3>Commits By Author:</h3>
<ul> <ul>
@@ -196,9 +188,9 @@
<li> <li>
<h4>Ben Ford (2):</h4> <h4>Ben Ford (2):</h4>
</li> </li>
<li>rtp_engine.c: Add exception for comfort noise payload.</li> <li>app_queue.c: Fix error in Queue parameter documentation.</li>
<li> <li>
<p>app_queue.c: Fix error in Queue parameter documentation.</p> <p>rtp_engine.c: Add exception for comfort noise payload.</p>
</li> </li>
<li> <li>
<h4>Christoph Moench-Tegeder (1):</h4> <h4>Christoph Moench-Tegeder (1):</h4>
@@ -207,17 +199,18 @@
<p>Fix Endianness detection in utils.h for non-Linux</p> <p>Fix Endianness detection in utils.h for non-Linux</p>
</li> </li>
<li> <li>
<h4>George Joseph (8):</h4> <h4>George Joseph (9):</h4>
</li> </li>
<li>ARI: The bridges play and record APIs now handle sample rates &gt; 8K correctly.</li> <li>channelstorage: Allow storage driver read locking to be skipped.</li>
<li>channelstorage_cpp_map_name_id: Add read locking around retrievals.</li> <li>res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function.</li>
<li>chan_websocket.c: Change payload references to command instead.</li>
<li>taskpool: Fix some references to threadpool that should be taskpool.</li>
<li>chan_pjsip: Add technology-specific off-nominal hangup cause to events.</li>
<li>safe_asterisk: Fix logging and sorting issue.</li>
<li>chan_pjsip: Disable SSRC change for WebRTC endpoints.</li> <li>chan_pjsip: Disable SSRC change for WebRTC endpoints.</li>
<li>safe_asterisk: Fix logging and sorting issue.</li>
<li>chan_pjsip: Add technology-specific off-nominal hangup cause to events.</li>
<li>taskpool: Fix some references to threadpool that should be taskpool.</li>
<li>chan_websocket.c: Change payload references to command instead.</li>
<li>channelstorage_cpp_map_name_id: Add read locking around retrievals.</li>
<li> <li>
<p>res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function.</p> <p>ARI: The bridges play and record APIs now handle sample rates &gt; 8K correctly.</p>
</li> </li>
<li> <li>
<h4>Igor Goncharovsky (1):</h4> <h4>Igor Goncharovsky (1):</h4>
@@ -226,17 +219,15 @@
<p>func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()</p> <p>func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()</p>
</li> </li>
<li> <li>
<h4>Joshua C. Colp (8):</h4> <h4>Joshua C. Colp (6):</h4>
</li> </li>
<li>taskpool: Add taskpool API, switch Stasis to using it.</li>
<li>taskpool: Update versions for taskpool stasis options.</li>
<li>sorcery: Move from threadpool to taskpool.</li>
<li>app_queue: Allow stasis message filtering to work.</li>
<li>endpoints: Remove need for stasis subscription.</li>
<li>devicestate: Don't publish redundant device state messages.</li> <li>devicestate: Don't publish redundant device state messages.</li>
<li>pjsip: Move from threadpool to taskpool</li> <li>endpoints: Remove need for stasis subscription.</li>
<li>app_queue: Allow stasis message filtering to work.</li>
<li>sorcery: Move from threadpool to taskpool.</li>
<li>taskpool: Update versions for taskpool stasis options.</li>
<li> <li>
<p>Revert "pjsip: Move from threadpool to taskpool"</p> <p>taskpool: Add taskpool API, switch Stasis to using it.</p>
</li> </li>
<li> <li>
<h4>Max Grobecker (1):</h4> <h4>Max Grobecker (1):</h4>
@@ -253,25 +244,25 @@
<li> <li>
<h4>Naveen Albert (18):</h4> <h4>Naveen Albert (18):</h4>
</li> </li>
<li>pbx_variables.c: Create real channel for "dialplan eval function".</li> <li>func_callerid: Document limitation of DNID fields.</li>
<li>res_cliexec: Remove unnecessary casts to char*.</li>
<li>app_adsiprog: Fix possible NULL dereference.</li>
<li>chan_dahdi: Add DAHDI_CHANNEL function.</li>
<li>sig_analog: Eliminate potential timeout with Last Number Redial.</li>
<li>func_scramble: Add example to XML documentation.</li>
<li>config_options.c: Improve misleading warning.</li>
<li>dsp.c: Make minor fixes to debug log messages.</li>
<li>app_dial: Allow fractional seconds for dial timeouts.</li>
<li>res_fax: Add XML documentation for channel variables.</li>
<li>res_tonedetect: Fix formatting of XML documentation.</li>
<li>codec_builtin.c: Adjust some of the quality scores to reflect reality.</li>
<li>app_sf: Add post-digit timer option to ReceiveSF.</li>
<li>func_math: Add DIGIT_SUM function.</li>
<li>sig_analog: Allow '#' to end the inter-digit timeout when dialing.</li>
<li>core_unreal: Preserve ADSI capability when dialing Local channels.</li>
<li>func_channel: Allow R/W of ADSI CPE capability setting.</li> <li>func_channel: Allow R/W of ADSI CPE capability setting.</li>
<li>core_unreal: Preserve ADSI capability when dialing Local channels.</li>
<li>sig_analog: Allow '#' to end the inter-digit timeout when dialing.</li>
<li>func_math: Add DIGIT_SUM function.</li>
<li>app_sf: Add post-digit timer option to ReceiveSF.</li>
<li>codec_builtin.c: Adjust some of the quality scores to reflect reality.</li>
<li>res_tonedetect: Fix formatting of XML documentation.</li>
<li>res_fax: Add XML documentation for channel variables.</li>
<li>app_dial: Allow fractional seconds for dial timeouts.</li>
<li>dsp.c: Make minor fixes to debug log messages.</li>
<li>config_options.c: Improve misleading warning.</li>
<li>func_scramble: Add example to XML documentation.</li>
<li>sig_analog: Eliminate potential timeout with Last Number Redial.</li>
<li>chan_dahdi: Add DAHDI_CHANNEL function.</li>
<li>app_adsiprog: Fix possible NULL dereference.</li>
<li>res_cliexec: Remove unnecessary casts to char*.</li>
<li> <li>
<p>func_callerid: Document limitation of DNID fields.</p> <p>pbx_variables.c: Create real channel for "dialplan eval function".</p>
</li> </li>
<li> <li>
<h4>Roman Pertsev (1):</h4> <h4>Roman Pertsev (1):</h4>
@@ -282,45 +273,46 @@
<li> <li>
<h4>Sean Bright (3):</h4> <h4>Sean Bright (3):</h4>
</li> </li>
<li>audiohook.c: Ensure correct AO2 reference is dereffed.</li> <li>safe_asterisk: Resolve a POSIX sh problem and restore globbing behavior.</li>
<li>app_externalivr: Prevent out-of-bounds read during argument processing.</li> <li>app_externalivr: Prevent out-of-bounds read during argument processing.</li>
<li> <li>
<p>safe_asterisk: Resolve a POSIX sh problem and restore globbing behavior.</p> <p>audiohook.c: Ensure correct AO2 reference is dereffed.</p>
</li> </li>
<li> <li>
<h4>Sven Kube (3):</h4> <h4>Sven Kube (3):</h4>
</li> </li>
<li>stasis_channels.c: Add null check for referred_by in ast_ari_transfer_message_..</li> <li>res_audiosocket: add message types for all slin sample rates</li>
<li>stasis_channels.c: Make protocol_id optional to enable blind transfer via ari</li> <li>stasis_channels.c: Make protocol_id optional to enable blind transfer via ari</li>
<li> <li>
<p>res_audiosocket: add message types for all slin sample rates</p> <p>stasis_channels.c: Add null check for referred_by in ast_ari_transfer_message_create</p>
</li> </li>
<li> <li>
<h4>Tinet-mucw (1):</h4> <h4>Tinet-mucw (1):</h4>
</li> </li>
<li> <li>
<p>iostream.c: Handle TLS handshake attacks in order to resolve the issue of exce..</p> <p>iostream.c: Handle TLS handshake attacks in order to resolve the issue of exceeding the maximum number of HTTPS sessions.</p>
</li> </li>
<li> <li>
<h4>gauravs456 (1):</h4> <h4>gauravs456 (1):</h4>
</li> </li>
<li> <li>
<p>chan_websocket: Add channel_id to MEDIA_START, DRIVER_STATUS and DTMF_END even..</p> <p>chan_websocket: Add channel_id to MEDIA_START, DRIVER_STATUS and DTMF_END events.</p>
</li> </li>
<li> <li>
<h4>phoneben (2):</h4> <h4>phoneben (2):</h4>
</li> </li>
<li>app_queue: Add NULL pointer checks in app_queue</li>
<li>res_fax.c: lower FAXOPT read warning to debug level</li> <li>res_fax.c: lower FAXOPT read warning to debug level</li>
<li>app_queue: Add NULL pointer checks in app_queue</li>
</ul> </ul>
<h3>Commit List:</h3> <h3>Commit List:</h3>
<ul> <ul>
<li>channelstorage: Allow storage driver read locking to be skipped.</li>
<li>res_audiosocket: fix temporarily unavailable</li> <li>res_audiosocket: fix temporarily unavailable</li>
<li>safe_asterisk: Resolve a POSIX sh problem and restore globbing behavior.</li> <li>safe_asterisk: Resolve a POSIX sh problem and restore globbing behavior.</li>
<li>res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function.</li> <li>res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function.</li>
<li>Revert "pjsip: Move from threadpool to taskpool"</li> <li>iostream.c: Handle TLS handshake attacks in order to resolve the issue of exceeding the maximum number of HTTPS sessions.</li>
<li>chan_pjsip: Disable SSRC change for WebRTC endpoints.</li> <li>chan_pjsip: Disable SSRC change for WebRTC endpoints.</li>
<li>pjsip: Move from threadpool to taskpool</li> <li>chan_websocket: Add channel_id to MEDIA_START, DRIVER_STATUS and DTMF_END events.</li>
<li>safe_asterisk: Fix logging and sorting issue.</li> <li>safe_asterisk: Fix logging and sorting issue.</li>
<li>Fix Endianness detection in utils.h for non-Linux</li> <li>Fix Endianness detection in utils.h for non-Linux</li>
<li>app_queue.c: Fix error in Queue parameter documentation.</li> <li>app_queue.c: Fix error in Queue parameter documentation.</li>
@@ -355,6 +347,7 @@
<li>stasis_channels.c: Make protocol_id optional to enable blind transfer via ari</li> <li>stasis_channels.c: Make protocol_id optional to enable blind transfer via ari</li>
<li>config.c: fix saving of deep/wide template configurations</li> <li>config.c: fix saving of deep/wide template configurations</li>
<li>Fix some doxygen, typos and whitespace</li> <li>Fix some doxygen, typos and whitespace</li>
<li>stasis_channels.c: Add null check for referred_by in ast_ari_transfer_message_create</li>
<li>app_queue: Add NULL pointer checks in app_queue</li> <li>app_queue: Add NULL pointer checks in app_queue</li>
<li>app_externalivr: Prevent out-of-bounds read during argument processing.</li> <li>app_externalivr: Prevent out-of-bounds read during argument processing.</li>
<li>chan_dahdi: Add DAHDI_CHANNEL function.</li> <li>chan_dahdi: Add DAHDI_CHANNEL function.</li>
@@ -368,6 +361,27 @@
<li>pbx_variables.c: Create real channel for "dialplan eval function".</li> <li>pbx_variables.c: Create real channel for "dialplan eval function".</li>
</ul> </ul>
<h3>Commit Details:</h3> <h3>Commit Details:</h3>
<h4>channelstorage: Allow storage driver read locking to be skipped.</h4>
<p>Author: George Joseph
Date: 2025-11-06</p>
<p>After PR #1498 added read locking to channelstorage_cpp_map_name_id, if ARI
channels/externalMedia was called with a custom channel id AND the
cpp_map_name_id channel storage backend is in use, a deadlock can occur when
hanging up the channel. It's actually triggered in
channel.c:__ast_channel_alloc_ap() when it gets a write lock on the
channelstorage driver then subsequently does a lookup for channel uniqueid
which now does a read lock. This is an invalid operation and causes the lock
state to get "bad". When the channels try to hang up, a write lock is
attempted again which hangs and causes the deadlock.</p>
<p>Now instead of the cpp_map_name_id channelstorage driver "get" APIs
automatically performing a read lock, they take a "lock" parameter which
allows a caller who already has a write lock to indicate that the "get" API
must not attempt its own lock. This prevents the state from getting mesed up.</p>
<p>The ao2_legacy driver uses the ao2 container's recursive mutex so doesn't
have this issue but since it also implements the common channelstorage API,
it needed its "get" implementations updated to take the lock parameter. They
just don't use it.</p>
<p>Resolves: #1578</p>
<h4>res_audiosocket: fix temporarily unavailable</h4> <h4>res_audiosocket: fix temporarily unavailable</h4>
<p>Author: Roman Pertsev <p>Author: Roman Pertsev
Date: 2025-10-07</p> Date: 2025-10-07</p>
@@ -401,11 +415,7 @@
<p>UserNote: The STIR_SHAKEN_ATTESTATION dialplan function has been added <p>UserNote: The STIR_SHAKEN_ATTESTATION dialplan function has been added
which will allow suppressing attestation on a call-by-call basis which will allow suppressing attestation on a call-by-call basis
regardless of the profile attached to the outgoing endpoint.</p> regardless of the profile attached to the outgoing endpoint.</p>
<h4>Revert "pjsip: Move from threadpool to taskpool"</h4> <h4>iostream.c: Handle TLS handshake attacks in order to resolve the issue of exceeding the maximum number of HTTPS sessions.</h4>
<p>Author: Joshua C. Colp
Date: 2025-10-27</p>
<p>This reverts commit bb6b76c2d8239b2665223dcbf6d507aa9aa4534e.</p>
<h4>iostream.c: Handle TLS handshake attacks in order to resolve the issue of exce..</h4>
<p>Author: Tinet-mucw <p>Author: Tinet-mucw
Date: 2025-10-26</p> Date: 2025-10-26</p>
<p>The TCP three-way handshake completes, but if the server is under a TLS handshake attack, asterisk will get stuck at SSL_do_handshake(). <p>The TCP three-way handshake completes, but if the server is under a TLS handshake attack, asterisk will get stuck at SSL_do_handshake().
@@ -419,27 +429,10 @@
clients that are sensitive to SSRC changes and non-monotonic timestamps so 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.</p> the fix is now disabled for endpoints with the "bundle" parameter set to true.</p>
<p>Resolves: #1535</p> <p>Resolves: #1535</p>
<h4>chan_websocket: Add channel_id to MEDIA_START, DRIVER_STATUS and DTMF_END even..</h4> <h4>chan_websocket: Add channel_id to MEDIA_START, DRIVER_STATUS and DTMF_END events.</h4>
<p>Author: gauravs456 <p>Author: gauravs456
Date: 2025-10-21</p> Date: 2025-10-21</p>
<p>Resolves: #1544</p> <p>Resolves: #1544</p>
<h4>pjsip: Move from threadpool to taskpool</h4>
<p>Author: Joshua C. Colp
Date: 2025-09-23</p>
<p>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.</p>
<p>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.</p>
<h4>safe_asterisk: Fix logging and sorting issue.</h4> <h4>safe_asterisk: Fix logging and sorting issue.</h4>
<p>Author: George Joseph <p>Author: George Joseph
Date: 2025-10-17</p> Date: 2025-10-17</p>
@@ -814,7 +807,7 @@
<h4>Fix some doxygen, typos and whitespace</h4> <h4>Fix some doxygen, typos and whitespace</h4>
<p>Author: Bastian Triller <p>Author: Bastian Triller
Date: 2025-09-21</p> Date: 2025-09-21</p>
<h4>stasis_channels.c: Add null check for referred_by in ast_ari_transfer_message_..</h4> <h4>stasis_channels.c: Add null check for referred_by in ast_ari_transfer_message_create</h4>
<p>Author: Sven Kube <p>Author: Sven Kube
Date: 2025-09-18</p> Date: 2025-09-18</p>
<p>When handling SIP transfers via ARI, the <code>referred_by</code> field in <p>When handling SIP transfers via ARI, the <code>referred_by</code> field in

177
ChangeLogs/ChangeLog-23.1.0-rc1.md → ChangeLogs/ChangeLog-23.1.0.md
View File

@@ -1,18 +1,18 @@
## Change Log for Release asterisk-23.1.0-rc1 ## Change Log for Release asterisk-23.1.0
### Links: ### Links:
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.1.0-rc1.html) - [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-23.1.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.0.0...23.1.0-rc1) - [GitHub Diff](https://github.com/asterisk/asterisk/compare/23.0.0...23.1.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.1.0-rc1.tar.gz) - [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-23.1.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk) - [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
### Summary: ### Summary:
- Commits: 54 - Commits: 53
- Commit Authors: 17 - Commit Authors: 17
- Issues Resolved: 36 - Issues Resolved: 37
- Security Advisories Resolved: 0 - Security Advisories Resolved: 0
### User Notes: ### User Notes:
@@ -50,14 +50,6 @@
### Upgrade Notes: ### 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. - #### app_queue.c: Fix error in Queue parameter documentation.
As part of Asterisk 21, macros were removed from Asterisk. As part of Asterisk 21, macros were removed from Asterisk.
This resulted in argument order changing for the Queue dialplan This resulted in argument order changing for the Queue dialplan
@@ -107,18 +99,18 @@
- Bastian Triller: (1) - Bastian Triller: (1)
- Ben Ford: (2) - Ben Ford: (2)
- Christoph Moench-Tegeder: (1) - Christoph Moench-Tegeder: (1)
- Gauravs456: (1) - George Joseph: (9)
- George Joseph: (8)
- Igor Goncharovsky: (1) - Igor Goncharovsky: (1)
- Joshua C. Colp: (8) - Joshua C. Colp: (6)
- Max Grobecker: (1) - Max Grobecker: (1)
- Nathan Monfils: (1) - Nathan Monfils: (1)
- Naveen Albert: (18) - Naveen Albert: (18)
- Phoneben: (2)
- Roman Pertsev: (1) - Roman Pertsev: (1)
- Sean Bright: (3) - Sean Bright: (3)
- Sven Kube: (3) - Sven Kube: (3)
- Tinet-Mucw: (1) - Tinet-mucw: (1)
- gauravs456: (1)
- phoneben: (2)
## Issue and Commit Detail: ## Issue and Commit Detail:
@@ -160,6 +152,7 @@
- 1544: [improvement]: While Receiving the MediaConnect Message Using External Media Over websocket ChannelID is Details are missing - 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 - 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. - 1559: [improvement]: Handle TLS handshake attacks in order to resolve the issue of exceeding the maximum number of HTTPS sessions.
- 1578: [bug]: Deadlock with externalMedia custom channel id and cpp map channel backend
### Commits By Author: ### Commits By Author:
@@ -173,34 +166,33 @@
- Fix some doxygen, typos and whitespace - Fix some doxygen, typos and whitespace
- #### Ben Ford (2): - #### Ben Ford (2):
- rtp_engine.c: Add exception for comfort noise payload.
- app_queue.c: Fix error in Queue parameter documentation. - app_queue.c: Fix error in Queue parameter documentation.
- rtp_engine.c: Add exception for comfort noise payload.
- #### Christoph Moench-Tegeder (1): - #### Christoph Moench-Tegeder (1):
- Fix Endianness detection in utils.h for non-Linux - Fix Endianness detection in utils.h for non-Linux
- #### George Joseph (8): - #### George Joseph (9):
- ARI: The bridges play and record APIs now handle sample rates > 8K correctly. - channelstorage: Allow storage driver read locking to be skipped.
- 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. - res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function.
- chan_pjsip: Disable SSRC change for WebRTC endpoints.
- safe_asterisk: Fix logging and sorting issue.
- chan_pjsip: Add technology-specific off-nominal hangup cause to events.
- taskpool: Fix some references to threadpool that should be taskpool.
- chan_websocket.c: Change payload references to command instead.
- channelstorage_cpp_map_name_id: Add read locking around retrievals.
- ARI: The bridges play and record APIs now handle sample rates > 8K correctly.
- #### Igor Goncharovsky (1): - #### Igor Goncharovsky (1):
- func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE() - func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()
- #### Joshua C. Colp (8): - #### Joshua C. Colp (6):
- 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. - devicestate: Don't publish redundant device state messages.
- pjsip: Move from threadpool to taskpool - endpoints: Remove need for stasis subscription.
- Revert "pjsip: Move from threadpool to taskpool" - app_queue: Allow stasis message filtering to work.
- sorcery: Move from threadpool to taskpool.
- taskpool: Update versions for taskpool stasis options.
- taskpool: Add taskpool API, switch Stasis to using it.
- #### Max Grobecker (1): - #### Max Grobecker (1):
- res_pjsip_geolocation: Add support for Geolocation loc-src parameter - res_pjsip_geolocation: Add support for Geolocation loc-src parameter
@@ -209,57 +201,57 @@
- manager.c: Fix presencestate object leak - manager.c: Fix presencestate object leak
- #### Naveen Albert (18): - #### 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. - 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.
- 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.
- 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.
- chan_dahdi: Add DAHDI_CHANNEL function.
- app_adsiprog: Fix possible NULL dereference.
- res_cliexec: Remove unnecessary casts to char*.
- pbx_variables.c: Create real channel for "dialplan eval function".
- #### Roman Pertsev (1): - #### Roman Pertsev (1):
- res_audiosocket: fix temporarily unavailable - res_audiosocket: fix temporarily unavailable
- #### Sean Bright (3): - #### 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. - safe_asterisk: Resolve a POSIX sh problem and restore globbing behavior.
- app_externalivr: Prevent out-of-bounds read during argument processing.
- audiohook.c: Ensure correct AO2 reference is dereffed.
- #### Sven Kube (3): - #### 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 - res_audiosocket: add message types for all slin sample rates
- stasis_channels.c: Make protocol_id optional to enable blind transfer via ari
- stasis_channels.c: Add null check for referred_by in ast_ari_transfer_message_create
- #### Tinet-mucw (1): - #### Tinet-mucw (1):
- iostream.c: Handle TLS handshake attacks in order to resolve the issue of exce.. - iostream.c: Handle TLS handshake attacks in order to resolve the issue of exceeding the maximum number of HTTPS sessions.
- #### gauravs456 (1): - #### gauravs456 (1):
- chan_websocket: Add channel_id to MEDIA_START, DRIVER_STATUS and DTMF_END even.. - chan_websocket: Add channel_id to MEDIA_START, DRIVER_STATUS and DTMF_END events.
- #### phoneben (2): - #### phoneben (2):
- app_queue: Add NULL pointer checks in app_queue
- res_fax.c: lower FAXOPT read warning to debug level - res_fax.c: lower FAXOPT read warning to debug level
- app_queue: Add NULL pointer checks in app_queue
### Commit List: ### Commit List:
- channelstorage: Allow storage driver read locking to be skipped.
- res_audiosocket: fix temporarily unavailable - res_audiosocket: fix temporarily unavailable
- safe_asterisk: Resolve a POSIX sh problem and restore globbing behavior. - safe_asterisk: Resolve a POSIX sh problem and restore globbing behavior.
- res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function. - res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function.
- Revert "pjsip: Move from threadpool to taskpool" - iostream.c: Handle TLS handshake attacks in order to resolve the issue of exceeding the maximum number of HTTPS sessions.
- chan_pjsip: Disable SSRC change for WebRTC endpoints. - chan_pjsip: Disable SSRC change for WebRTC endpoints.
- pjsip: Move from threadpool to taskpool - chan_websocket: Add channel_id to MEDIA_START, DRIVER_STATUS and DTMF_END events.
- safe_asterisk: Fix logging and sorting issue. - safe_asterisk: Fix logging and sorting issue.
- Fix Endianness detection in utils.h for non-Linux - Fix Endianness detection in utils.h for non-Linux
- app_queue.c: Fix error in Queue parameter documentation. - app_queue.c: Fix error in Queue parameter documentation.
@@ -294,6 +286,7 @@
- stasis_channels.c: Make protocol_id optional to enable blind transfer via ari - stasis_channels.c: Make protocol_id optional to enable blind transfer via ari
- config.c: fix saving of deep/wide template configurations - config.c: fix saving of deep/wide template configurations
- Fix some doxygen, typos and whitespace - Fix some doxygen, typos and whitespace
- stasis_channels.c: Add null check for referred_by in ast_ari_transfer_message_create
- app_queue: Add NULL pointer checks in app_queue - app_queue: Add NULL pointer checks in app_queue
- app_externalivr: Prevent out-of-bounds read during argument processing. - app_externalivr: Prevent out-of-bounds read during argument processing.
- chan_dahdi: Add DAHDI_CHANNEL function. - chan_dahdi: Add DAHDI_CHANNEL function.
@@ -308,6 +301,32 @@
### Commit Details: ### Commit Details:
#### channelstorage: Allow storage driver read locking to be skipped.
Author: George Joseph
Date: 2025-11-06
After PR #1498 added read locking to channelstorage_cpp_map_name_id, if ARI
channels/externalMedia was called with a custom channel id AND the
cpp_map_name_id channel storage backend is in use, a deadlock can occur when
hanging up the channel. It's actually triggered in
channel.c:__ast_channel_alloc_ap() when it gets a write lock on the
channelstorage driver then subsequently does a lookup for channel uniqueid
which now does a read lock. This is an invalid operation and causes the lock
state to get "bad". When the channels try to hang up, a write lock is
attempted again which hangs and causes the deadlock.
Now instead of the cpp_map_name_id channelstorage driver "get" APIs
automatically performing a read lock, they take a "lock" parameter which
allows a caller who already has a write lock to indicate that the "get" API
must not attempt its own lock. This prevents the state from getting mesed up.
The ao2_legacy driver uses the ao2 container's recursive mutex so doesn't
have this issue but since it also implements the common channelstorage API,
it needed its "get" implementations updated to take the lock parameter. They
just don't use it.
Resolves: #1578
#### res_audiosocket: fix temporarily unavailable #### res_audiosocket: fix temporarily unavailable
Author: Roman Pertsev Author: Roman Pertsev
Date: 2025-10-07 Date: 2025-10-07
@@ -345,13 +364,7 @@
which will allow suppressing attestation on a call-by-call basis which will allow suppressing attestation on a call-by-call basis
regardless of the profile attached to the outgoing endpoint. regardless of the profile attached to the outgoing endpoint.
#### Revert "pjsip: Move from threadpool to taskpool" #### iostream.c: Handle TLS handshake attacks in order to resolve the issue of exceeding the maximum number of HTTPS sessions.
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 Author: Tinet-mucw
Date: 2025-10-26 Date: 2025-10-26
@@ -371,32 +384,12 @@
Resolves: #1535 Resolves: #1535
#### chan_websocket: Add channel_id to MEDIA_START, DRIVER_STATUS and DTMF_END even.. #### chan_websocket: Add channel_id to MEDIA_START, DRIVER_STATUS and DTMF_END events.
Author: gauravs456 Author: gauravs456
Date: 2025-10-21 Date: 2025-10-21
Resolves: #1544 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. #### safe_asterisk: Fix logging and sorting issue.
Author: George Joseph Author: George Joseph
Date: 2025-10-17 Date: 2025-10-17
@@ -872,7 +865,7 @@
Date: 2025-09-21 Date: 2025-09-21
#### stasis_channels.c: Add null check for referred_by in ast_ari_transfer_message_.. #### stasis_channels.c: Add null check for referred_by in ast_ari_transfer_message_create
Author: Sven Kube Author: Sven Kube
Date: 2025-09-18 Date: 2025-09-18

4
README.html
View File

@@ -1,4 +1,4 @@
<html><head><title>Readme for asterisk-23.1.0-rc2</title></head><body> <html><head><title>Readme for asterisk-23.1.0</title></head><body>
<h1>The Asterisk(R) Open Source PBX</h1> <h1>The Asterisk(R) Open Source PBX</h1>
<pre><code>By Mark Spencer &lt;markster@digium.com&gt; and the Asterisk.org developer community. <pre><code>By Mark Spencer &lt;markster@digium.com&gt; and the Asterisk.org developer community.
Copyright (C) 2001-2025 Sangoma Technologies Corporation and other copyright holders. Copyright (C) 2001-2025 Sangoma Technologies Corporation and other copyright holders.
@@ -37,7 +37,7 @@ hardware.</p>
<p>If you are updating from a previous version of Asterisk, make sure you <p>If you are updating from a previous version of Asterisk, make sure you
read the Change Logs.</p> read the Change Logs.</p>
<!-- CHANGELOGS (the URL will change based on the location of this README) --> <!-- CHANGELOGS (the URL will change based on the location of this README) -->
<p><a href="ChangeLogs/ChangeLog-23.1.0-rc2.html">Change Logs</a></p> <p><a href="ChangeLogs/ChangeLog-23.1.0.html">Change Logs</a></p>
<!-- END-CHANGELOGS --> <!-- END-CHANGELOGS -->
<h3>NEW INSTALLATIONS</h3> <h3>NEW INSTALLATIONS</h3>

2
README.md
View File

@@ -55,7 +55,7 @@ If you are updating from a previous version of Asterisk, make sure you
read the Change Logs. read the Change Logs.
<!-- CHANGELOGS (the URL will change based on the location of this README) --> <!-- CHANGELOGS (the URL will change based on the location of this README) -->
[Change Logs](ChangeLogs/ChangeLog-23.1.0-rc2.html) [Change Logs](ChangeLogs/ChangeLog-23.1.0.html)
<!-- END-CHANGELOGS --> <!-- END-CHANGELOGS -->
### NEW INSTALLATIONS ### NEW INSTALLATIONS