## Change Log for Release asterisk-20.11.0-rc1

### Links:

 - [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-20.11.0-rc1.md)  
 - [GitHub Diff](https://github.com/asterisk/asterisk/compare/20.10.0...20.11.0-rc1)  
 - [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-20.11.0-rc1.tar.gz)  
 - [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)  

### Summary:

- Commits: 39
- Commit Authors: 9
- Issues Resolved: 21
- Security Advisories Resolved: 0

### User Notes:

- #### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"                   
  The new "suppress_moh_on_sendonly" endpoint option
  can be used to prevent playing MOH back to a caller if the remote
  end sends "sendonly" or "inactive" (hold) to Asterisk in an SDP.

- #### app_mixmonitor: Add 'D' option for dual-channel audio.                          
  The MixMonitor application now has a new 'D' option which
  interleaves the recorded audio in the output frames. This allows for
  stereo recording output with one channel being the transmitted audio and
  the other being the received audio. The 't' and 't' options are
  compatible with this.

- #### manager.c: Restrict ModuleLoad to the configured modules directory.             
  The ModuleLoad AMI action now restricts modules to the
  configured modules directory.

- #### manager: Enhance event filtering for performance                                
  You can now perform more granular filtering on events
  in manager.conf using expressions like
  `eventfilter(name(Newchannel),header(Channel),method(starts_with)) = PJSIP/`
  This is much more efficient than
  `eventfilter = Event: Newchannel.*Channel: PJSIP/`
  Full syntax guide is in configs/samples/manager.conf.sample.

- #### db.c: Remove limit on family/key length                                         
  The `ast_db_*()` APIs have had the 253 byte limit on
  "/family/key" removed and will now accept families and keys with a
  total length of up to SQLITE_MAX_LENGTH (currently 1e9!).  This
  affects the `DB*` dialplan applications, dialplan functions,
  manager actions and `databse` CLI commands.  Since the
  media_cache also uses the `ast_db_*()` APIs, you can now store
  resources with URIs longer than 253 bytes.


### Upgrade Notes:


### Commit Authors:

- Allan Nathanson: (1)
- Ben Ford: (3)
- Chrsmj: (1)
- George Joseph: (14)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (8)
- Thomas Guebels: (2)

## Issue and Commit Detail:

### Closed Issues:

  - 487: [bug]: Segfault possibly in ast_rtp_stop
  - 821: [bug]: app_dial:  The progress timeout doesn't cause Dial to exit
  - 881: [bug]: Long URLs are being rejected by the media cache because of an astdb key length limit
  - 882: [bug]: Value CHANNEL(userfield) is lost by BRIDGE_ENTER
  - 897: [improvement]: Restrict ModuleLoad AMI action to the modules directory
  - 900: [bug]: astfd.c: NULL pointer passed to fclose with nonnull attribute causes compilation failure
  - 902: [bug]: app_voicemail: Pager emails are ill-formatted when custom subject is used
  - 916: [bug]: Compilation errors on FreeBSD
  - 923: [bug]: Transport monitor shutdown callback only works on the first disconnection
  - 924: [bug]: dnsmgr.c: dnsmgr_refresh() should not flag change if IP address order changes
  - 928: [bug]: chan_dahdi: MWI while off-hook when hung up on after recall ring
  - 932: [bug]: When connected to multiple IP addresses the transport monitor is only set on the first one
  - 937: [bug]: Wrong format for sample config file 'geolocation.conf.sample'
  - 938: [bug]: memory leak - CBAnn leaks a small amount format_cap related memory for every confbridge
  - 945: [improvement]: Add stereo recording support for app_mixmonitor
  - 951: [new-feature]: func_evalexten: Add `EVAL_SUB` function
  - 974: [improvement]: change and/or remove some wiki mentions to docs mentions in the sample configs
  - 979: [improvement]: Add ability to suppress MOH when a remote endpoint sends "sendonly" or "inactive"
  - 982: [bug]: The addition of tenantid to the ast_sip_endpoint structure broke ABI compatibility
  - 990: [improvement]: The help for PJSIP_AOR should indicate that you need to call PJSIP_CONTACT to get contact details

### Commits By Author:

- #### Allan Nathanson (1):
  - dnsmgr.c: dnsmgr_refresh() incorrectly flags change with DNS round-robin

- #### Ben Ford (3):
  - manager.c: Restrict ModuleLoad to the configured modules directory.
  - app_mixmonitor: Add 'D' option for dual-channel audio.
  - Add res_pjsip_config_sangoma external module.

- #### George Joseph (14):
  - db.c: Remove limit on family/key length
  - manager.c: Split XML documentation to manager_doc.xml
  - manager: Enhance event filtering for performance
  - manager.conf.sample: Fix mathcing typo
  - Fix application references to Background
  - res_rtp_asterisk: Fix dtls timer issues causing FRACKs and SEGVs
  - manager.c: Add unit test for Originate app and appdata permissions
  - geolocation.sample.conf: Fix comment marker at end of file
  - core_unreal.c: Fix memory leak in ast_unreal_new_channels()
  - pjproject_bundled:  Tweaks to support out-of-tree development
  - res_srtp: Change Unsupported crypto suite msg from verbose to debug
  - res_pjsip: Move tenantid to end of ast_sip_endpoint
  - func_pjsip_aor/contact: Fix documentation for contact ID
  - res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"

- #### Naveen Albert (7):
  - app_voicemail: Fix ill-formatted pager emails with custom subject.
  - astfd.c: Avoid calling fclose with NULL argument.
  - main, res, tests: Fix compilation errors on FreeBSD.
  - chan_dahdi: Never send MWI while off-hook.
  - app_dial: Fix progress timeout.
  - app_dial: Fix progress timeout calculation with no answer timeout.
  - func_evalexten: Add EVAL_SUB function.

- #### Peter Jannesen (2):
  - cel_custom: Allow absolute filenames.
  - channel: Preserve CHANNEL(userfield) on masquerade.

- #### Sean Bright (8):
  - res_pjsip_pubsub: Persist subscription 'generator_data' in sorcery
  - cdr_custom: Allow absolute filenames.
  - res_agi.c: Ensure SIGCHLD handler functions are properly balanced.
  - alembic: Drop redundant voicemail_messages index.
  - func_base64.c: Ensure we set aside enough room for base64 encoded data.
  - Revert "res_rtp_asterisk: Count a roll-over of the sequence number even on los..
  - chan_sip.c: Fix __sip_reliable_xmit build error
  - res_pjsip.c: Fix Contact header rendering for IPv6 addresses.

- #### Thomas Guebels (2):
  - pjsip_transport_events: Avoid monitor destruction
  - pjsip_transport_events: handle multiple addresses for a domain

- #### chrsmj (1):
  - samples: remove and/or change some wiki mentions

- #### jiangxc (1):
  - res_agi.c: Prevent possible double free during `SPEECH RECOGNIZE`


### Commit List:

-  res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
-  res_pjsip.c: Fix Contact header rendering for IPv6 addresses.
-  samples: remove and/or change some wiki mentions
-  func_pjsip_aor/contact: Fix documentation for contact ID
-  res_pjsip: Move tenantid to end of ast_sip_endpoint
-  pjsip_transport_events: handle multiple addresses for a domain
-  func_evalexten: Add EVAL_SUB function.
-  res_srtp: Change Unsupported crypto suite msg from verbose to debug
-  Add res_pjsip_config_sangoma external module.
-  app_mixmonitor: Add 'D' option for dual-channel audio.
-  pjsip_transport_events: Avoid monitor destruction
-  app_dial: Fix progress timeout calculation with no answer timeout.
-  pjproject_bundled:  Tweaks to support out-of-tree development
-  chan_sip.c: Fix __sip_reliable_xmit build error
-  core_unreal.c: Fix memory leak in ast_unreal_new_channels()
-  dnsmgr.c: dnsmgr_refresh() incorrectly flags change with DNS round-robin
-  geolocation.sample.conf: Fix comment marker at end of file
-  func_base64.c: Ensure we set aside enough room for base64 encoded data.
-  app_dial: Fix progress timeout.
-  chan_dahdi: Never send MWI while off-hook.
-  manager.c: Add unit test for Originate app and appdata permissions
-  alembic: Drop redundant voicemail_messages index.
-  res_agi.c: Ensure SIGCHLD handler functions are properly balanced.
-  main, res, tests: Fix compilation errors on FreeBSD.
-  res_rtp_asterisk: Fix dtls timer issues causing FRACKs and SEGVs
-  manager.c: Restrict ModuleLoad to the configured modules directory.
-  res_agi.c: Prevent possible double free during `SPEECH RECOGNIZE`
-  cdr_custom: Allow absolute filenames.
-  astfd.c: Avoid calling fclose with NULL argument.
-  channel: Preserve CHANNEL(userfield) on masquerade.
-  cel_custom: Allow absolute filenames.
-  app_voicemail: Fix ill-formatted pager emails with custom subject.
-  res_pjsip_pubsub: Persist subscription 'generator_data' in sorcery
-  Fix application references to Background
-  manager.conf.sample: Fix mathcing typo
-  manager: Enhance event filtering for performance
-  manager.c: Split XML documentation to manager_doc.xml
-  db.c: Remove limit on family/key length

### Commit Details:

#### res_pjsip: Add new endpoint option "suppress_moh_on_sendonly"
  Author: George Joseph
  Date:   2024-11-05

  Normally, when one party in a call sends Asterisk an SDP with
  a "sendonly" or "inactive" attribute it means "hold" and causes
  Asterisk to start playing MOH back to the other party. This can be
  problematic if it happens at certain times, such as in a 183
  Progress message, because the MOH will replace any early media you
  may be playing to the calling party. If you set this option
  to "yes" on an endpoint and the endpoint receives an SDP
  with "sendonly" or "inactive", Asterisk will NOT play MOH back to
  the other party.

  Resolves: #979

  UserNote: The new "suppress_moh_on_sendonly" endpoint option
  can be used to prevent playing MOH back to a caller if the remote
  end sends "sendonly" or "inactive" (hold) to Asterisk in an SDP.


#### res_pjsip.c: Fix Contact header rendering for IPv6 addresses.
  Author: Sean Bright
  Date:   2024-11-08

  Fix suggested by @nvsystems.

  Fixes #985


#### samples: remove and/or change some wiki mentions
  Author: chrsmj
  Date:   2024-11-01

  Cleaned some dead links. Replaced word wiki with
  either docs or link to https://docs.asterisk.org/

  Resolves: #974

#### func_pjsip_aor/contact: Fix documentation for contact ID
  Author: George Joseph
  Date:   2024-11-09

  Clarified the use of the contact ID returned from PJSIP_AOR.

  Resolves: #990

#### res_pjsip: Move tenantid to end of ast_sip_endpoint
  Author: George Joseph
  Date:   2024-11-06

  The tenantid field was originally added to the ast_sip_endpoint
  structure at the end of the AST_DECLARE_STRING_FIELDS block.  This
  caused everything after it in the structure to move down in memory
  and break ABI compatibility.  It's now at the end of the structure
  as an AST_STRING_FIELD_EXTENDED.  Given the number of string fields
  in the structure now, the initial string field allocation was
  also increased from 64 to 128 bytes.

  Resolves: #982

#### pjsip_transport_events: handle multiple addresses for a domain
  Author: Thomas Guebels
  Date:   2024-10-29

  The key used for transport monitors was the remote host name for the
  transport and not the remote address resolved for this domain.

  This was problematic for domains returning multiple addresses as several
  transport monitors were created with the same key.

  Whenever a subsystem wanted to register a callback it would always end
  up attached to the first transport monitor with a matching key.

  The key used for transport monitors is now the remote address and port
  the transport actually connected to.

  Fixes: #932

#### func_evalexten: Add EVAL_SUB function.
  Author: Naveen Albert
  Date:   2024-10-17

  This adds an EVAL_SUB function, which is similar to the existing
  EVAL_EXTEN function but significantly more powerful, as it allows
  executing arbitrary dialplan and capturing its return value as
  the function's output. While EVAL_EXTEN should be preferred if it
  is possible to use it, EVAL_SUB can be used in a wider variety
  of cases and allows arbitrary computation to be performed in
  a dialplan function call, leveraging the dialplan.

  Resolves: #951

#### res_srtp: Change Unsupported crypto suite msg from verbose to debug
  Author: George Joseph
  Date:   2024-11-01

  There's really no point in spamming logs with a verbose message
  for every unsupported crypto suite an older client may send
  in an SDP.  If none are supported, there will be an error or
  warning.


#### Add res_pjsip_config_sangoma external module.
  Author: Ben Ford
  Date:   2024-11-01

  Adds res_pjsip_config_sangoma as an external module that can be
  downloaded via menuselect. It lives under the Resource Modules section.


#### app_mixmonitor: Add 'D' option for dual-channel audio.
  Author: Ben Ford
  Date:   2024-10-28

  Adds the 'D' option to app_mixmonitor that interleaves the input and
  output frames of the channel being recorded in the monitor output frame.
  This allows for two streams in the recording: the transmitted audio and
  the received audio. The 't' and 'r' options are compatible with this.

  Fixes: #945

  UserNote: The MixMonitor application now has a new 'D' option which
  interleaves the recorded audio in the output frames. This allows for
  stereo recording output with one channel being the transmitted audio and
  the other being the received audio. The 't' and 't' options are
  compatible with this.


#### pjsip_transport_events: Avoid monitor destruction
  Author: Thomas Guebels
  Date:   2024-10-28

  When a transport is disconnected, several events can arrive following
  each other. The first event will be PJSIP_TP_STATE_DISCONNECT and it
  will trigger the destruction of the transport monitor object. The lookup
  for the transport monitor to destroy is done using the transport key,
  that contains the transport destination host:port.

  A reconnect attempt by pjsip will be triggered as soon something needs to
  send a packet using that transport. This can happen directly after a
  disconnect since ca

  Subsequent events can arrive later like PJSIP_TP_STATE_DESTROY and will
  also try to trigger the destruction of the transport monitor if not
  already done. Since the lookup for the transport monitor to destroy is
  done using the transport key, it can match newly created transports
  towards the same destination and destroy their monitor object.

  Because of this, it was sometimes not possible to monitor a transport
  after one or more disconnections.

  This fix adds an additional check on the transport pointer to ensure
  only a monitor for that specific transport is removed.

  Fixes: #923

#### app_dial: Fix progress timeout calculation with no answer timeout.
  Author: Naveen Albert
  Date:   2024-10-16

  If to_answer is -1, simply comparing to see if the progress timeout
  is smaller than the answer timeout to prefer it will fail. Add
  an additional check that chooses the progress timeout if there is
  no answer timeout (or as before, if the progress timeout is smaller).

  Resolves: #821

#### pjproject_bundled:  Tweaks to support out-of-tree development
  Author: George Joseph
  Date:   2024-10-17

  * pjproject is now configured with --disable-libsrtp so it will
    build correctly when doing "out-of-tree" development.  Asterisk
    doesn't use pjproject for handling media so pjproject doesn't
    need libsrtp itself.

  * The pjsua app (which we used to use for the testsuite) no longer
    builds in pjproject's master branch so we just skip it.  The
    testsuite no longer needs it anyway.

  See third-party/pjproject/README-hacking.md for more info on building
  pjproject "out-of-tree".


#### chan_sip.c: Fix __sip_reliable_xmit build error
  Author: Sean Bright
  Date:   2024-10-17

  Fixes #954


#### Revert "res_rtp_asterisk: Count a roll-over of the sequence number even on los..
  Author: Sean Bright
  Date:   2024-10-07

  This reverts commit cb5e3445be6c55517c8d05aca601b648341f8ae9.

  The original change from 16 to 15 bit sequence numbers was predicated
  on the following from the now-defunct libSRTP FAQ on sourceforge.net:

  > *Q6. The use of implicit synchronization via ROC seems
  > dangerous. Can senders and receivers lose ROC synchronization?*
  >
  > **A.** It is possible to lose ROC synchronization between sender and
  > receiver(s), though it is not likely in practice, and practical
  > steps can be taken to avoid it. A burst loss of 2^16 packets or more
  > will always break synchronization. For example, a conversational
  > voice codec that sends 50 packets per second will have its ROC
  > increment about every 22 minutes. A network with a burst of packet
  > loss that long has problems other than ROC synchronization.
  >
  > There is a higher sensitivity to loss at the very outset of an SRTP
  > stream. If the sender's initial sequence number is close to the
  > maximum value of 2^16-1, and all packets are lost from the initial
  > packet until the sequence number cycles back to zero, the sender
  > will increment its ROC, but the receiver will not. The receiver
  > cannot determine that the initial packets were lost and that
  > sequence-number rollover has occurred. In this case, the receiver's
  > ROC would be zero whereas the sender's ROC would be one, while their
  > sequence numbers would be so close that the ROC-guessing algorithm
  > could not detect this fact.
  >
  > There is a simple solution to this problem: the SRTP sender should
  > randomly select an initial sequence number that is always less than
  > 2^15. This ensures correct SRTP operation so long as fewer than 2^15
  > initial packets are lost in succession, which is within the maximum
  > tolerance of SRTP packet-index determination (see Appendix A and
  > page 14, first paragraph of RFC 3711). An SRTP receiver should
  > carefully implement the index-guessing algorithm. A naive
  > implementation can unintentionally guess the value of
  > 0xffffffffffffLL whenever the SEQ in the packet is greater than 2^15
  > and the locally stored SEQ and ROC are zero. (This can happen when
  > the implementation fails to treat those zero values as a special
  > case.)
  >
  > When ROC synchronization is lost, the receiver will not be able to
  > properly process the packets. If anti-replay protection is turned
  > on, then the desynchronization will appear as a burst of replay
  > check failures. Otherwise, if authentication is being checked, then
  > it will appear as a burst of authentication failures. Otherwise, if
  > encryption is being used, the desynchronization may not be detected
  > by the SRTP layer, and the packets may be improperly decrypted.

  However, modern libSRTP (as of 1.0.1[1]) now mentions the following in
  their README.md[2]:

  > The sequence number in the rtp packet is used as the low 16 bits of
  > the sender's local packet index. Note that RTP will start its
  > sequence number in a random place, and the SRTP layer just jumps
  > forward to that number at its first invocation. An earlier version
  > of this library used initial sequence numbers that are less than
  > 32,768; this trick is no longer required as the
  > rdbx_estimate_index(...) function has been made smarter.

  So truncating our initial sequence number to 15 bit is no longer
  necessary.

  1. https://github.com/cisco/libsrtp/blob/0eb007f0dc611f27cbfe0bf9855ed85182496cec/CHANGES#L271-L289
  2. https://github.com/cisco/libsrtp/blob/2de20dd9e9c8afbaf02fcf5d4048ce1ec9ddc0ae/README.md#implementation-notes


#### core_unreal.c: Fix memory leak in ast_unreal_new_channels()
  Author: George Joseph
  Date:   2024-10-15

  When the channel tech is multistream capable, the reference to
  chan_topology was passed to the new channel.  When the channel tech
  isn't multistream capable, the reference to chan_topology was never
  released.  "Local" channels are multistream capable so it didn't
  affect them but the confbridge "CBAnn" and the bridge_media
  "Recorder" channels are not so they caused a leak every time one
  of them was created.

  Also added tracing to ast_stream_topology_alloc() and
  stream_topology_destroy() to assist with debugging.

  Resolves: #938

#### dnsmgr.c: dnsmgr_refresh() incorrectly flags change with DNS round-robin
  Author: Allan Nathanson
  Date:   2024-09-29

  The dnsmgr_refresh() function checks to see if the IP address associated
  with a name/service has changed. The gotcha is that the ast_get_ip_or_srv()
  function only returns the first IP address returned by the DNS query. If
  there are multiple IPs associated with the name and the returned order is
  not consistent (e.g. with DNS round-robin) then the other IP addresses are
  not included in the comparison and the entry is flagged as changed even
  though the IP is still valid.

  Updated the code to check all IP addresses and flag a change only if the
  original IP is no longer valid.

  Resolves: #924

#### geolocation.sample.conf: Fix comment marker at end of file
  Author: George Joseph
  Date:   2024-10-08

  Resolves: #937

#### func_base64.c: Ensure we set aside enough room for base64 encoded data.
  Author: Sean Bright
  Date:   2024-10-08

  Reported by SingularTricycle on IRC.

  Fixes #940


#### app_dial: Fix progress timeout.
  Author: Naveen Albert
  Date:   2024-10-03

  Under some circumstances, the progress timeout feature added in commit
  320c98eec87c473bfa814f76188a37603ea65ddd does not work as expected,
  such as if there is no media flowing. Adjust the waitfor call to
  explicitly use the progress timeout if it would be reached sooner than
  the answer timeout to ensure we handle the timers properly.

  Resolves: #821

#### chan_dahdi: Never send MWI while off-hook.
  Author: Naveen Albert
  Date:   2024-10-01

  In some circumstances, it is possible for the do_monitor thread to
  erroneously think that a line is on-hook and send an MWI FSK spill
  to it when the line is really off-hook and no MWI should be sent.
  Commit 0a8b3d34673277b70be6b0e8ac50191b1f3c72c6 previously fixed this
  issue in a more readily encountered scenario, but it has still been
  possible for MWI to be sent when it shouldn't be. To robustly fix
  this issue, query DAHDI for the hook status to ensure we don't send
  MWI on a line that is actually still off hook.

  Resolves: #928

#### manager.c: Add unit test for Originate app and appdata permissions
  Author: George Joseph
  Date:   2024-10-03

  This unit test checks that dialplan apps and app data specified
  as parameters for the Originate action are allowed with the
  permissions the user has.


#### alembic: Drop redundant voicemail_messages index.
  Author: Sean Bright
  Date:   2024-09-26

  The `voicemail_messages_dir` index is a left prefix of the table's
  primary key and therefore unnecessary.


#### res_agi.c: Ensure SIGCHLD handler functions are properly balanced.
  Author: Sean Bright
  Date:   2024-09-30

  Calls to `ast_replace_sigchld()` and `ast_unreplace_sigchld()` must be
  balanced to ensure that we can capture the exit status of child
  processes when we need to. This extends to functions that call
  `ast_replace_sigchld()` and `ast_unreplace_sigchld()` such as
  `ast_safe_fork()` and `ast_safe_fork_cleanup()`.

  The primary change here is ensuring that we do not call
  `ast_safe_fork_cleanup()` in `res_agi.c` if we have not previously
  called `ast_safe_fork()`.

  Additionally we reinforce some of the documentation and add an
  assertion to, ideally, catch this sooner were this to happen again.

  Fixes #922


#### main, res, tests: Fix compilation errors on FreeBSD.
  Author: Naveen Albert
  Date:   2024-09-29

  asterisk.c, manager.c: Increase buffer sizes to avoid truncation warnings.
  config.c: Include header file for WIFEXITED/WEXITSTATUS macros.
  res_timing_kqueue: Use more portable format specifier.
  test_crypto: Use non-linux limits.h header file.

  Resolves: #916

#### res_rtp_asterisk: Fix dtls timer issues causing FRACKs and SEGVs
  Author: George Joseph
  Date:   2024-09-16

  In dtls_srtp_handle_timeout(), when DTLSv1_get_timeout() returned
  success but with a timeout of 0, we were stopping the timer and
  decrementing the refcount on instance but not resetting the
  timeout_timer to -1.  When dtls_srtp_stop_timeout_timer()
  was later called, it was atempting to stop a stale timer and could
  decrement the refcount on instance again which would then cause
  the instance destructor to run early.  This would result in either
  a FRACK or a SEGV when ast_rtp_stop(0 was called.

  According to the OpenSSL docs, we shouldn't have been stopping the
  timer when DTLSv1_get_timeout() returned success and the new timeout
  was 0 anyway.  We should have been calling DTLSv1_handle_timeout()
  again immediately so we now reschedule the timer callback for
  1ms (almost immediately).

  Additionally, instead of scheduling the timer callback at a fixed
  interval returned by the initial call to DTLSv1_get_timeout()
  (usually 999 ms), we now reschedule the next callback based on
  the last call to DTLSv1_get_timeout().

  Resolves: #487

#### manager.c: Restrict ModuleLoad to the configured modules directory.
  Author: Ben Ford
  Date:   2024-09-25

  When using the ModuleLoad AMI action, it was possible to traverse
  upwards through the directories to files outside of the configured
  modules directory. We decided it would be best to restrict access to
  modules exclusively in the configured directory. You will now get an
  error when the specified module is outside of this limitation.

  Fixes: #897

  UserNote: The ModuleLoad AMI action now restricts modules to the
  configured modules directory.


#### res_agi.c: Prevent possible double free during `SPEECH RECOGNIZE`
  Author: jiangxc
  Date:   2024-07-17

  When using the speech recognition module, crashes can occur
  sporadically due to a "double free or corruption (out)" error. Now, in
  the section where the audio stream is being captured in a loop, each
  time after releasing fr, it is set to NULL to prevent repeated
  deallocation.

  Fixes #772


#### cdr_custom: Allow absolute filenames.
  Author: Sean Bright
  Date:   2024-09-26

  A follow up to #893 that brings the same functionality to
  cdr_custom. Also update the sample configuration files to note support
  for absolute paths.


#### astfd.c: Avoid calling fclose with NULL argument.
  Author: Naveen Albert
  Date:   2024-09-24

  Don't pass through a NULL argument to fclose, which is undefined
  behavior, and instead return -1 and set errno appropriately. This
  also avoids a compiler warning with glibc 2.38 and newer, as glibc
  commit 71d9e0fe766a3c22a730995b9d024960970670af
  added the nonnull attribute to this argument.

  Resolves: #900

#### channel: Preserve CHANNEL(userfield) on masquerade.
  Author: Peter Jannesen
  Date:   2024-09-20

  In certain circumstances a channel may undergo an operation
  referred to as a masquerade. If this occurs the CHANNEL(userfield)
  value was not preserved causing it to get lost. This change makes
  it so that this field is now preserved.

  Fixes: #882

#### cel_custom: Allow absolute filenames.
  Author: Peter Jannesen
  Date:   2024-09-20

  If a filename starts with a '/' in cel_custom [mappings] assume it is
  a absolute file path and not relative filename/path to
  AST_LOG_DIR/cel_custom/


#### app_voicemail: Fix ill-formatted pager emails with custom subject.
  Author: Naveen Albert
  Date:   2024-09-24

  Add missing end-of-headers newline to pager emails with custom
  subjects, since this was missing from this code path.

  Resolves: #902

#### res_pjsip_pubsub: Persist subscription 'generator_data' in sorcery
  Author: Sean Bright
  Date:   2024-09-23

  Fixes #895


#### Fix application references to Background
  Author: George Joseph
  Date:   2024-09-20

  The app is actually named "BackGround" but several references
  in XML documentation were spelled "Background" with the lower
  case "g".  This was causing documentation links to return
  "not found" messages.


#### manager.conf.sample: Fix mathcing typo
  Author: George Joseph
  Date:   2024-09-24


#### manager: Enhance event filtering for performance
  Author: George Joseph
  Date:   2024-07-31

  UserNote: You can now perform more granular filtering on events
  in manager.conf using expressions like
  `eventfilter(name(Newchannel),header(Channel),method(starts_with)) = PJSIP/`
  This is much more efficient than
  `eventfilter = Event: Newchannel.*Channel: PJSIP/`
  Full syntax guide is in configs/samples/manager.conf.sample.


#### manager.c: Split XML documentation to manager_doc.xml
  Author: George Joseph
  Date:   2024-08-01


#### db.c: Remove limit on family/key length
  Author: George Joseph
  Date:   2024-09-11

  Consumers like media_cache have been running into issues with
  the previous astdb "/family/key" limit of 253 bytes when needing
  to store things like long URIs.  An Amazon S3 URI is a good example
  of this.  Now, instead of using a static 256 byte buffer for
  "/family/key", we use ast_asprintf() to dynamically create it.

  Both test_db.c and test_media_cache.c were also updated to use
  keys/URIs over the old 253 character limit.

  Resolves: #881

  UserNote: The `ast_db_*()` APIs have had the 253 byte limit on
  "/family/key" removed and will now accept families and keys with a
  total length of up to SQLITE_MAX_LENGTH (currently 1e9!).  This
  affects the `DB*` dialplan applications, dialplan functions,
  manager actions and `databse` CLI commands.  Since the
  media_cache also uses the `ast_db_*()` APIs, you can now store
  resources with URIs longer than 253 bytes.