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Update for 20.17.0

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Asterisk Development Team
2025-11-20 18:11:47 +00:00
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20.17.0-rc2
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<html><head><title>ChangeLog for asterisk-20.17.0-rc2</title></head><body>
<h2>Change Log for Release asterisk-20.17.0-rc2</h2>
<h3>Links:</h3>
<ul>
<li><a href="https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.17.0-rc2.html">Full ChangeLog</a> </li>
<li><a href="https://github.com/asterisk/asterisk/compare/20.17.0-rc1...20.17.0-rc2">GitHub Diff</a> </li>
<li><a href="https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.17.0-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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ChangeLogs/ChangeLog-20.17.0-rc2.md
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## Change Log for Release asterisk-20.17.0-rc2
### Links:
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.17.0-rc2.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.17.0-rc1...20.17.0-rc2)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.17.0-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-20.17.0-rc1</title></head><body>
<h2>Change Log for Release asterisk-20.17.0-rc1</h2>
<html><head><title>ChangeLog for asterisk-20.17.0</title></head><body>
<h2>Change Log for Release asterisk-20.17.0</h2>
<h3>Links:</h3>
<ul>
<li><a href="https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.17.0-rc1.html">Full ChangeLog</a> </li>
<li><a href="https://github.com/asterisk/asterisk/compare/20.16.0...20.17.0-rc1">GitHub Diff</a> </li>
<li><a href="https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.17.0-rc1.tar.gz">Tarball</a> </li>
<li><a href="https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.17.0.html">Full ChangeLog</a> </li>
<li><a href="https://github.com/asterisk/asterisk/compare/20.16.0...20.17.0">GitHub Diff</a> </li>
<li><a href="https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.17.0.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: 51</li>
<li>Commits: 50</li>
<li>Commit Authors: 16</li>
<li>Issues Resolved: 33</li>
<li>Issues Resolved: 34</li>
<li>Security Advisories Resolved: 0</li>
</ul>
<h3>User Notes:</h3>
@@ -57,15 +57,6 @@
<h3>Upgrade Notes:</h3>
<ul>
<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>res_audiosocket: add message types for all slin sample rates</h4>
<p>New audiosocket message types 0x11 - 0x18 has been added
for slin12, slin16, slin24, slin32, slin44, slin48, slin96, and
@@ -110,18 +101,18 @@
<li>Bastian Triller: (1)</li>
<li>Ben Ford: (1)</li>
<li>Christoph Moench-Tegeder: (1)</li>
<li>Gauravs456: (1)</li>
<li>George Joseph: (8)</li>
<li>George Joseph: (9)</li>
<li>Igor Goncharovsky: (1)</li>
<li>Joshua C. Colp: (8)</li>
<li>Joshua C. Colp: (6)</li>
<li>Max Grobecker: (1)</li>
<li>Nathan Monfils: (1)</li>
<li>Naveen Albert: (17)</li>
<li>Phoneben: (2)</li>
<li>Roman Pertsev: (1)</li>
<li>Sean Bright: (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>
<h2>Issue and Commit Detail:</h2>
<h3>Closed Issues:</h3>
@@ -159,6 +150,7 @@
<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>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>
<h3>Commits By Author:</h3>
<ul>
@@ -166,140 +158,60 @@
<h4>Anthony Minessale (1):</h4>
</li>
<li>
<p>Update contact information for anthm</p>
</li>
<li>
<h4>Bastian Triller (1):</h4>
</li>
<li>
<p>Fix some doxygen, typos and whitespace</p>
</li>
<li>
<h4>Ben Ford (1):</h4>
</li>
<li>
<p>rtp_engine.c: Add exception for comfort noise payload.</p>
</li>
<li>
<h4>Christoph Moench-Tegeder (1):</h4>
</li>
<li>
<p>Fix Endianness detection in utils.h for non-Linux</p>
</li>
<li>
<h4>George Joseph (8):</h4>
</li>
<li>ARI: The bridges play and record APIs now handle sample rates &gt; 8K correctly.</li>
<li>channelstorage_cpp_map_name_id: Add read locking around retrievals.</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>
<p>res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function.</p>
<h4>George Joseph (9):</h4>
</li>
<li>
<h4>Igor Goncharovsky (1):</h4>
</li>
<li>
<p>func_hangupcause.c: Add access to Reason headers via HANGUPCAUSE()</p>
</li>
<li>
<h4>Joshua C. Colp (8):</h4>
</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>pjsip: Move from threadpool to taskpool</li>
<li>
<p>Revert "pjsip: Move from threadpool to taskpool"</p>
<h4>Joshua C. Colp (6):</h4>
</li>
<li>
<h4>Max Grobecker (1):</h4>
</li>
<li>
<p>res_pjsip_geolocation: Add support for Geolocation loc-src parameter</p>
</li>
<li>
<h4>Nathan Monfils (1):</h4>
</li>
<li>
<p>manager.c: Fix presencestate object leak</p>
</li>
<li>
<h4>Naveen Albert (17):</h4>
</li>
<li>pbx_variables.c: Create real channel for "dialplan eval function".</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>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>
<p>func_callerid: Document limitation of DNID fields.</p>
</li>
<li>
<h4>Roman Pertsev (1):</h4>
</li>
<li>
<p>res_audiosocket: fix temporarily unavailable</p>
</li>
<li>
<h4>Sean Bright (3):</h4>
</li>
<li>audiohook.c: Ensure correct AO2 reference is dereffed.</li>
<li>app_externalivr: Prevent out-of-bounds read during argument processing.</li>
<li>
<p>safe_asterisk: Resolve a POSIX sh problem and restore globbing behavior.</p>
</li>
<li>
<h4>Sven Kube (3):</h4>
</li>
<li>stasis_channels.c: Add null check for referred_by in ast_ari_transfer_message_..</li>
<li>stasis_channels.c: Make protocol_id optional to enable blind transfer via ari</li>
<li>
<p>res_audiosocket: add message types for all slin sample rates</p>
</li>
<li>
<h4>Tinet-mucw (1):</h4>
</li>
<li>
<p>iostream.c: Handle TLS handshake attacks in order to resolve the issue of exce..</p>
</li>
<li>
<h4>gauravs456 (1):</h4>
</li>
<li>
<p>chan_websocket: Add channel_id to MEDIA_START, DRIVER_STATUS and DTMF_END even..</p>
</li>
<li>
<h4>phoneben (2):</h4>
</li>
<li>app_queue: Add NULL pointer checks in app_queue</li>
<li>res_fax.c: lower FAXOPT read warning to debug level</li>
</ul>
<h3>Commit List:</h3>
<ul>
<li>channelstorage: Allow storage driver read locking to be skipped.</li>
<li>res_audiosocket: fix temporarily unavailable</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>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>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>Fix Endianness detection in utils.h for non-Linux</li>
<li>devicestate: Don't publish redundant device state messages.</li>
@@ -331,6 +243,7 @@
<li>sorcery: Move from threadpool to taskpool.</li>
<li>stasis_channels.c: Make protocol_id optional to enable blind transfer via ari</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_externalivr: Prevent out-of-bounds read during argument processing.</li>
<li>chan_dahdi: Add DAHDI_CHANNEL function.</li>
@@ -344,6 +257,27 @@
<li>pbx_variables.c: Create real channel for "dialplan eval function".</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>
<h4>res_audiosocket: fix temporarily unavailable</h4>
<p>Author: Roman Pertsev
Date: 2025-10-07</p>
@@ -377,11 +311,7 @@
<p>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.</p>
<h4>Revert "pjsip: Move from threadpool to taskpool"</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>
<h4>iostream.c: Handle TLS handshake attacks in order to resolve the issue of exceeding the maximum number of HTTPS sessions.</h4>
<p>Author: Tinet-mucw
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().
@@ -395,27 +325,10 @@
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>
<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
Date: 2025-10-21</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>
<p>Author: George Joseph
Date: 2025-10-17</p>
@@ -751,7 +664,7 @@
<h4>Fix some doxygen, typos and whitespace</h4>
<p>Author: Bastian Triller
Date: 2025-09-22</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
Date: 2025-09-18</p>
<p>When handling SIP transfers via ARI, the <code>referred_by</code> field in

175
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View File

@@ -1,71 +1,63 @@
## Change Log for Release asterisk-20.17.0-rc1
## Change Log for Release asterisk-20.17.0
### Links:
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.17.0-rc1.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.16.0...20.17.0-rc1)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.17.0-rc1.tar.gz)
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.17.0.html)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.16.0...20.17.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.17.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
### Summary:
- Commits: 51
- Commits: 50
- Commit Authors: 16
- Issues Resolved: 33
- Issues Resolved: 34
- Security Advisories Resolved: 0
### User Notes:
- #### res_stir_shaken: Add STIR_SHAKEN_ATTESTATION dialplan function.
- #### 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.
- #### 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()
- #### 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.
- #### 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.
- #### 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.
- #### 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.
- #### 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.
- #### res_audiosocket: add message types for all slin sample rates
- #### 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.
- #### 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.
@@ -73,7 +65,7 @@
### Developer Notes:
- #### chan_pjsip: Add technology-specific off-nominal hangup cause to events.
- #### 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
@@ -81,12 +73,12 @@
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.
- #### 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.
- #### 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.
@@ -98,18 +90,18 @@
- Bastian Triller: (1)
- Ben Ford: (1)
- Christoph Moench-Tegeder: (1)
- Gauravs456: (1)
- George Joseph: (8)
- George Joseph: (9)
- Igor Goncharovsky: (1)
- Joshua C. Colp: (8)
- Joshua C. Colp: (6)
- Max Grobecker: (1)
- Nathan Monfils: (1)
- Naveen Albert: (17)
- Phoneben: (2)
- Roman Pertsev: (1)
- Sean Bright: (3)
- Sven Kube: (3)
- Tinet-Mucw: (1)
- Tinet-mucw: (1)
- gauravs456: (1)
- phoneben: (2)
## Issue and Commit Detail:
@@ -148,101 +140,51 @@
- 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.
- 1578: [bug]: Deadlock with externalMedia custom channel id and cpp map channel backend
### Commits By Author:
- #### Anthony Minessale (1):
- Update contact information for anthm
- #### Bastian Triller (1):
- Fix some doxygen, typos and whitespace
- #### Ben Ford (1):
- rtp_engine.c: Add exception for comfort noise payload.
- #### 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.
- #### George Joseph (9):
- #### 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"
- #### Joshua C. Colp (6):
- #### Max Grobecker (1):
- res_pjsip_geolocation: Add support for Geolocation loc-src parameter
- #### Nathan Monfils (1):
- manager.c: Fix presencestate object leak
- #### Naveen Albert (17):
- 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.
- 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:
- channelstorage: Allow storage driver read locking to be skipped.
- 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"
- 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.
- 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.
- Fix Endianness detection in utils.h for non-Linux
- devicestate: Don't publish redundant device state messages.
@@ -274,6 +216,7 @@
- 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
- 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_externalivr: Prevent out-of-bounds read during argument processing.
- chan_dahdi: Add DAHDI_CHANNEL function.
@@ -288,6 +231,32 @@
### 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
Author: Roman Pertsev
Date: 2025-10-07
@@ -325,13 +294,7 @@
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..
#### iostream.c: Handle TLS handshake attacks in order to resolve the issue of exceeding the maximum number of HTTPS sessions.
Author: Tinet-mucw
Date: 2025-10-26
@@ -351,32 +314,12 @@
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
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
@@ -799,7 +742,7 @@
Date: 2025-09-22
#### 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
Date: 2025-09-18

4
README.html
View File

@@ -1,4 +1,4 @@
<html><head><title>Readme for asterisk-20.17.0-rc2</title></head><body>
<html><head><title>Readme for asterisk-20.17.0</title></head><body>
<h1>The Asterisk(R) Open Source PBX</h1>
<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.
@@ -37,7 +37,7 @@ hardware.</p>
<p>If you are updating from a previous version of Asterisk, make sure you
read the Change Logs.</p>
<!-- CHANGELOGS (the URL will change based on the location of this README) -->
<p><a href="ChangeLogs/ChangeLog-20.17.0-rc2.html">Change Logs</a></p>
<p><a href="ChangeLogs/ChangeLog-20.17.0.html">Change Logs</a></p>
<!-- END-CHANGELOGS -->
<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.
<!-- CHANGELOGS (the URL will change based on the location of this README) -->
[Change Logs](ChangeLogs/ChangeLog-20.17.0-rc2.html)
[Change Logs](ChangeLogs/ChangeLog-20.17.0.html)
<!-- END-CHANGELOGS -->
### NEW INSTALLATIONS