Based on the firing order of the PJSIP call-backs on a redirect, it was possible for
the Diversion header to not be included in the outgoing 181 response to the UAC and
the INVITE to the UAS.
This change moves the Diversion header processing to an earlier PJSIP callback while also
preventing the corresponding update that can cause a duplicate 181 response when processing
the header at that time.
Resolves: #1349
UpgradeNote: The safe_asterisk script now checks that, if it was run by the
root user, the /etc/asterisk/startup.d directory and all the files it contains
are owned by root. If the checks fail, safe_asterisk will exit with an error
and Asterisk will not be started. Additionally, the default logging
destination is now stderr instead of tty "9" which probably won't exist
in modern systems.
Resolves: #GHSA-v9q8-9j8m-5xwp
The CDR tenantid was being set in cdr_object_alloc from the channel->base
snapshot. Since this happens at channel creation before the dialplan is even
reached, calls to `CHANNEL(tenantid)=<something>` in the dialplan were being
ignored. Instead we now take tenantid from party_a when
cdr_object_create_public_records() is called which is after the call has
ended and all channel snapshots rebuilt. This is exactly how accountcode
and amaflags, which can also be set in tha dialplpan, are handled.
Resolves: #1259
UserNote: A new asterisk.conf option 'disable_remote_console_shell' has
been added that, when set, will prevent remote consoles from executing
shell commands using the '!' prefix.
Resolves: #GHSA-c7p6-7mvq-8jq2
Incoming SIP MESSAGEs will now have their From header's display name
sanitized by replacing any characters < 32 (space) with a space.
Resolves: #GHSA-2grh-7mhv-fcfw
The following utilities have been backported:
ast_sip_is_uri_sip_sips
ast_sip_is_allowed_uri
ast_sip_pjsip_uri_get_username
ast_sip_pjsip_uri_get_hostname
ast_sip_pjsip_uri_get_other_param
They were originally included in the commit for supporting TEL uris.
Support for TEL uris is NOT included here however.
There are several build fixes that never made it into certified/18.9.
Unfortunately the commits that contained the fixes also contained other
stuff that won't cherry-pick into cert so the build files had to be
just copied from 18.
With gcc (Ubuntu 11.2.0-19ubuntu1) 11.2.0:
> chan_dahdi.c:4129:18: error: ‘%s’ directive output may be truncated
> writing up to 255 bytes into a region of size between 242 and 252
> [-Werror=format-truncation=]
This removes the error-prone sizeof(...) calculations in favor of just
doubling the size of the base buffer.
Change-Id: I2d276785286730d3d5d0a921bcea2e065dbf27c5
Two functions are deprecated as of libxml2 2.12:
* xmlSubstituteEntitiesDefault
* xmlParseMemory
So we update those with supported API.
Additionally, `res_calendar_caldav` has been updated to use libxml2's
xmlreader API instead of the SAX2 API which has always felt a little
hacky (see deleted comment block in `res_calendar_caldav.c`).
The xmlreader API has been around since libxml2 2.5.0 which was
released in 2003.
Fixes#725
Gosub and Goto were not displaying their syntax correctly on the docs
site. This change adds a new way to specify an optional context, an
optional extension, and a required priority that the xml stylesheet can
parse without having to know which optional parameters come in which
order. In Asterisk, it looks like this:
parameter name="context" documentationtype="dialplan_context"
parameter name="extension" documentationtype="dialplan_extension"
parameter name="priority" documentationtype="dialplan_priority" required="true"
The stylesheet will ignore the context and extension parameters, but for
priority, it will automatically inject the following:
[[context,]extension,]priority
This is the correct oder for applications such as Gosub and Goto.
* Outdated information has been removed.
* New links added.
* Placeholder added for link to change logs.
Going forward, the release process will create HTML versions of the README
and change log and will update the link in the README to the current
change log for the branch...
* In the development branches, the link will always point to the current
release on GitHub.
* In the "releases/*" branches and the tarballs, the link will point to the
ChangeLogs/ChangeLog-<version>.html file in the source directory.
* On the downloads website, the link will point to the
ChangeLog-<version>.html file in the same directory.
Resolves: #1131
This fixes a number of broken links throughout the
tree, mostly caused by wiki.asterisk.org being replaced
with docs.asterisk.org, which should eliminate the
need for sporadic fixes as in f28047db36.
Resolves: #430
Adding information in the readme about running the install_preqreq script to install components that the ./configure script might indicate as missing.
ASTERISK-29976 #close
Change-Id: Ic287b46300168729838bddd8f9265e98fc22bce6
GitHub strikes again. Apparently the github.ref context variable only
contains the PR number if the workflow is triggered by "pull_request" so
since we just changed the trigger to "pull_request_target" the variable
no longer contains the PR number and is therefore not unique and can't be
used as a concurrency group id. We now use
`github.triggering_actor-github.head_ref`.
After careful review, we believe we can now use the "pull_request_target"
workflow trigger instead of "pull_request" which required a separate
privliged workflow to add labels and comments to PRs when they are submitted
or updated. This allows us to greatly streamline our workflows and remove
unneeded ones.
* The OnPRChanged workflow was...
* Renamed to OnPRCheck
* Changed to trigger on pull_request_target and the "recheckpr" label.
* Changed to simply call reusable workflows in asterisk-ci-actions.
* Changed to use better concurrency groups.
* The OnPRCPCheck and OnPRMergeApproved workflows were also...
* Changed to simply call reusable workflows in asterisk-ci-actions.
* Changed to use better concurrency groups.
* The NightlyTest and CreateDocs were also tweaked
When using the ListCategories AMI action, it was possible to traverse
upwards through the directories to files outside of the configured
configuration directory. This action is now restricted to the configured
directory and an error will now be returned if the specified file is
outside of this limitation.
Resolves: #GHSA-33x6-fj46-6rfh
UserNote: The ListCategories AMI action now restricts files to the
configured configuration directory.
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
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.
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
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.
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
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.
