Asterisk
/
asterisk
mirror of https://github.com/asterisk/asterisk
1
0
Fork 0
Code Issues Releases Wiki Activity
You can not select more than 25 topics Topics must start with a letter or number, can include dashes ('-') and can be up to 35 characters long.
Branches Tags
${ item.name }
Create tag ${ searchTerm }
Create branch ${ searchTerm }
from 'ff38e11ded'
${ noResults }
asterisk/ChangeLogs/ChangeLog-21.6.0.md

758 lines
30 KiB
Raw Blame History

## Change Log for Release asterisk-21.6.0
### Links:
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-21.6.0.md)
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/21.5.0...21.6.0)
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-21.6.0.tar.gz)
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
### Summary:
- Commits: 39
- Commit Authors: 9
- Issues Resolved: 22
- 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: (15)
- Jiangxc: (1)
- Naveen Albert: (7)
- Peter Jannesen: (2)
- Sean Bright: (7)
- 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
- 995: [bug]: suppress_moh_on_sendonly should use AST_BOOL_VALUES instead of YESNO_VALUES in alembic script
### 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 (15):
- 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"
- res_pjsip: Change suppress_moh_on_sendonly to OPT_BOOL_T
- #### 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 (7):
- 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..
- 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: Change suppress_moh_on_sendonly to OPT_BOOL_T
- 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
- 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: Change suppress_moh_on_sendonly to OPT_BOOL_T
Author: George Joseph
Date: 2024-11-15
The suppress_moh_on_sendonly endpoint option should have been
defined as OPT_BOOL_T in pjsip_configuration.c and AST_BOOL_VALUES
in the alembic script instead of OPT_YESNO_T and YESNO_VALUES.
Also updated contrib/ast-db-manage/README.md to indicate that
AST_BOOL_VALUES should always be used and provided an example.
Resolves: #995
#### 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".
#### 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.
Reference in new issue View Git Blame Copy Permalink