mirror of https://github.com/asterisk/asterisk
Asterisk Development Team
11 months ago
parent
ca5b810fe3
commit
be6dd21420
@ -1 +1 @@
|
||||
ChangeLogs/ChangeLog-18.25.0.md
|
||||
ChangeLogs/ChangeLog-18.26.0-rc1.md
|
||||
@ -0,0 +1,545 @@
|
||||
|
||||
## Change Log for Release asterisk-18.26.0-rc1
|
||||
|
||||
### Links:
|
||||
|
||||
- [Full ChangeLog](https://downloads.asterisk.org/pub/telephony/asterisk/releases/ChangeLog-18.26.0-rc1.md)
|
||||
- [GitHub Diff](https://github.com/asterisk/asterisk/compare/18.25.0...18.26.0-rc1)
|
||||
- [Tarball](https://downloads.asterisk.org/pub/telephony/asterisk/asterisk-18.26.0-rc1.tar.gz)
|
||||
- [Downloads](https://downloads.asterisk.org/pub/telephony/asterisk)
|
||||
|
||||
### Summary:
|
||||
|
||||
- Commits: 28
|
||||
- Commit Authors: 7
|
||||
- Issues Resolved: 13
|
||||
- Security Advisories Resolved: 0
|
||||
|
||||
### User Notes:
|
||||
|
||||
- #### 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: (1)
|
||||
- George Joseph: (10)
|
||||
- Jiangxc: (1)
|
||||
- Naveen Albert: (6)
|
||||
- Peter Jannesen: (2)
|
||||
- Sean Bright: (7)
|
||||
|
||||
## 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
|
||||
- 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
|
||||
- 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
|
||||
|
||||
### Commits By Author:
|
||||
|
||||
- #### Allan Nathanson (1):
|
||||
- dnsmgr.c: dnsmgr_refresh() incorrectly flags change with DNS round-robin
|
||||
|
||||
- #### Ben Ford (1):
|
||||
- manager.c: Restrict ModuleLoad to the configured modules directory.
|
||||
|
||||
- #### George Joseph (10):
|
||||
- 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
|
||||
|
||||
- #### Naveen Albert (6):
|
||||
- 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.
|
||||
|
||||
- #### 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..
|
||||
- chan_sip.c: Fix __sip_reliable_xmit build error
|
||||
|
||||
- #### jiangxc (1):
|
||||
- res_agi.c: Prevent possible double free during `SPEECH RECOGNIZE`
|
||||
|
||||
|
||||
### Commit List:
|
||||
|
||||
- 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:
|
||||
|
||||
#### 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.
|
||||
|
||||
|
||||
Loading…
Reference in new issue