res_pjsip_header_funcs: Duplicate new header value, don't copy.
When updating an existing header the 'update' code incorrectly
just copied the new value into the existing buffer. If the
new value exceeded the available buffer size memory outside
of the buffer would be written into, potentially causing
a crash.
This change makes it so that the 'update' now duplicates
the new header value instead of copying it into the existing
buffer.
Mike Bradeen [Tue, 25 Jul 2023 16:23:12 +0000 (10:23 -0600)]
res_pjsip: disable raw bad packet logging
Add patch to split the log level for invalid packets received on the
signaling port. The warning regarding the packet will move to level 2
so that it can still be displayed, while the raw packet will be at level
4.
George Joseph [Thu, 9 Nov 2023 16:14:03 +0000 (09:14 -0700)]
res_rtp_asterisk.c: Check DTLS packets against ICE candidate list
When ICE is in use, we can prevent a possible DOS attack by allowing
DTLS protocol messages (client hello, etc) only from sources that
are in the active remote candidates list.
Ben Ford [Mon, 13 Nov 2023 17:08:50 +0000 (11:08 -0600)]
manager.c: Prevent path traversal with GetConfig.
When using AMI GetConfig, it was possible to access files outside of the
Asterisk configuration directory by using filenames with ".." and "./"
even while live_dangerously was not enabled. This change resolves the
full path and ensures we are still in the configuration directory before
attempting to access the file.
George Joseph [Tue, 5 Sep 2023 15:51:19 +0000 (09:51 -0600)]
ari-stubs: Fix broken documentation anchors
All of the links that reference page anchors with capital letters in
the ids (#Something) have been changed to lower case to match the
anchors that are generated by mkdocs.
Bastian Triller [Mon, 28 Aug 2023 07:16:43 +0000 (09:16 +0200)]
res_pjsip_session: Send Session Interval too small response
Handle session interval lower than endpoint's configured minimum timer
when sending first answer. Timer setting is checked during this step and
needs to handled appropriately.
Before this change, no response was sent at all. After this change a
response with 422 Session Interval too small is sent to UAC.
Naveen Albert [Mon, 28 Aug 2023 17:51:23 +0000 (13:51 -0400)]
app_dial: Fix infinite loop when sending digits.
If the called party hangs up while digits are being
sent, -1 is returned to indicate so, but app_dial
was not checking the return value, resulting in
the hangup being lost and looping forever until
the caller manually hangs up the channel. We now
abort if digit sending fails.
George Joseph [Mon, 28 Aug 2023 13:11:27 +0000 (07:11 -0600)]
alembic: Fix quoting of the 100rel column
Add quoting around the ps_endpoints 100rel column in the ALTER
statements. Although alembic doesn't complain when generating
sql statements, postgresql does (rightly so).
George Joseph [Fri, 18 Aug 2023 16:20:47 +0000 (10:20 -0600)]
download_externals: Fix a few version related issues
* Fixed issue with the script not parsing the new tag format for
certified releases. The format changed from certified/18.9-cert5
to certified-18.9-cert5.
* Fixed issue where the asterisk version wasn't being considered
when looking for cached versions.
Naveen Albert [Wed, 9 Aug 2023 22:04:23 +0000 (22:04 +0000)]
sig_analog: Add Called Subscriber Held capability.
This adds support for Called Subscriber Held for FXS
lines, which allows users to go on hook when receiving
a call and resume the call later from another phone on
the same line, without disconnecting the call. This is
a convenience mechanism that most real PSTN telephone
switches support.
ASTERISK-30372 #close
Resolves: #240
UserNote: Called Subscriber Held is now supported for analog
FXS channels, using the calledsubscriberheld option. This allows
a station user to go on hook when receiving an incoming call
and resume from another phone on the same line by going on hook,
without disconnecting the call.
app_macro sometimes would crash due to datastore list corruption on the
channel because of lack of locking around find and create process for
the macro datastore. This patch locks the channel lock prior to protect
against this problem.
This seems to break the case when a predial macro calls a gosub.
When the gosub calls return, the Return function outputs:
app_stack.c:423 return_exec: Return without Gosub: stack is empty
This returns -1 to the calling macro, which returns to app_dial
and causes the call to hangup instead of proceeding with the macro
that invoked the gosub.
install_prereq: Fix dependency install on aarch64.
Fixes dependency solutions in install_prereq for Debian aarch64
platforms. install_prereq was attempting to forcibly install 32-bit
armhf packages due to the aptitude search for dependencies.
MikeNaso [Tue, 8 Aug 2023 14:23:37 +0000 (10:23 -0400)]
res_pjsip.c: Set contact_user on incoming call local Contact header
If the contact_user is configured on the endpoint it will now be set on the local Contact header URI for incoming calls. The contact_user has already been set on the local Contact header URI for outgoing calls.
Sean Bright [Wed, 12 Jul 2023 16:03:59 +0000 (12:03 -0400)]
extconfig: Allow explicit DB result set ordering to be disabled.
Added a new boolean configuration flag -
`order_multi_row_results_by_initial_column` - to both res_pgsql.conf
and res_config_odbc.conf that allows the administrator to disable the
explicit `ORDER BY` that was previously being added to all generated
SQL statements that returned multiple rows.
George Joseph [Wed, 9 Aug 2023 19:10:22 +0000 (13:10 -0600)]
rest-api: Run make ari-stubs
An earlier cherry-pick that involved rest-api somehow didn't include
a comment change in res/ari/resource_endpoints.h. This commit
corrects that. No changes other than the comment.
Naveen Albert [Wed, 9 Aug 2023 23:17:26 +0000 (23:17 +0000)]
res_pjsip_header_funcs: Make prefix argument optional.
The documentation for PJSIP_HEADERS claims that
prefix is optional, but in the code it is actually not.
However, there is no inherent reason for this, as users
may want to retrieve all header names, not just those
beginning with a certain prefix.
This makes the prefix optional for this function,
simply fetching all header names if not specified.
As a result, the documentation is now correct.
Resolves: #230
UserNote: The prefix argument to PJSIP_HEADERS is now
optional. If not specified, all header names will be
returned.
George Joseph [Fri, 11 Aug 2023 13:10:18 +0000 (07:10 -0600)]
pjproject_bundled: Increase PJSIP_MAX_MODULE to 38
The default is 32 with 8 being used by pjproject itself. Recent
commits have put us over the limit resulting in assertions in
pjproject. Since this value is used in invites, dialogs,
transports and subscriptions as well as the global pjproject
endpoint, we don't want to increase it too much.
Joshua C. Colp [Wed, 9 Aug 2023 10:57:33 +0000 (07:57 -0300)]
manager: Tolerate stasis messages with no channel snapshot.
In some cases I have yet to determine some stasis messages may
be created without a channel snapshot. This change adds some
tolerance to this scenario, preventing a crash from occurring.
This change adds support for refers that are not session based. It
includes a refer implementation for the PJSIP technology which results
in out-of-dialog REFERs being sent to a PJSIP endpoint. These can be
triggered using the new ARI endpoint `/endpoints/refer`.
Resolves: #71
UserNote: There is a new ARI endpoint `/endpoints/refer` for referring
an endpoint to some URI or endpoint.
Naveen Albert [Fri, 4 Aug 2023 17:26:05 +0000 (17:26 +0000)]
chan_dahdi: Allow autoreoriginating after hangup.
Currently, if an FXS channel is still off hook when
all calls on the line have hung up, the user is provided
reorder tone until going back on hook again.
In addition to not reflecting what most commercial switches
actually do, it's very common for switches to automatically
reoriginate for the user so that dial tone is provided without
the user having to depress and release the hookswitch manually.
This can increase convenience for users.
This behavior is now supported for kewlstart FXS channels.
It's supported only for kewlstart (FXOKS) mainly because the
behavior doesn't make any sense for ground start channels,
and loop start signalling doesn't provide the necessary DAHDI
event that makes this easy to implement. Likely almost everyone
is using FXOKS over FXOLS anyways since FXOLS is pretty useless
these days.
ASTERISK-30357 #close
Resolves: #224
UserNote: The autoreoriginate setting now allows for kewlstart FXS
channels to automatically reoriginate and provide dial tone to the
user again after all calls on the line have cleared. This saves users
from having to manually hang up and pick up the receiver again before
making another call.
Naveen Albert [Mon, 10 Jul 2023 18:53:43 +0000 (18:53 +0000)]
sig_analog: Allow three-way flash to time out to silence.
sig_analog allows users to flash and use the three-way dial
tone as a primitive hold function, simply by never timing
it out.
Some systems allow this dial tone to time out to silence,
so the user is not annoyed by a persistent dial tone.
This option allows the dial tone to time out normally to
silence.
ASTERISK-30004 #close
Resolves: #205
UserNote: The threewaysilenthold option now allows the three-way
dial tone to time out to silence, rather than continuing forever.
app_queue: Add support for applying caller priority change immediately.
The app_queue module provides both an AMI action and a CLI command
to change the priority of a caller in a queue. Up to now this change
of priority has only been reflected to new callers into the queue.
This change adds an "immediate" option to both the AMI action and
CLI command which immediately applies the priority change respective
to the other callers already in the queue. This can allow, for example,
a caller to be placed at the head of the queue immediately if their
priority is sufficient.
Resolves: #202
UserNote: The 'queue priority caller' CLI command and
'QueueChangePriorityCaller' AMI action now have an 'immediate'
argument which allows the caller priority change to be reflected
immediately, causing the position of a caller to move within the
queue depending on the priorities of the other callers.
Sean Bright [Fri, 7 Jul 2023 19:57:39 +0000 (15:57 -0400)]
chan_iax2.c: Avoid crash with IAX2 switch support.
A change made in 82cebaa0 did not properly handle the case when a
channel was not provided, triggering a crash. ast_check_hangup(...)
does not protect against NULL pointers.
Mike Bradeen [Thu, 29 Jun 2023 18:20:53 +0000 (12:20 -0600)]
Adds manager actions to allow move/remove/forward individual messages
in a particular mailbox folder. The forward command can be used
to copy a message within a mailbox or to another mailbox. Also adds
a VoicemailBoxSummarry, required to retrieve message ID's.
Resolves: #181
UserNote: The following manager actions have been added
VoicemailBoxSummary - Generate message list for a given mailbox
VoicemailRemove - Remove a message from a mailbox folder
VoicemailMove - Move a message from one folder to another within a mailbox
VoicemailForward - Copy a message from one folder in one mailbox
to another folder in another or the same mailbox.
Mike Bradeen [Tue, 20 Jun 2023 16:32:14 +0000 (10:32 -0600)]
app_voicemail: add CLI commands for message manipulation
Adds CLI commands to allow move/remove/forward individual messages
from a particular mailbox folder. The forward command can be used
to copy a message within a mailbox or to another mailbox. Also adds
a show mailbox, required to retrieve message ID's.
Resolves: #170
UserNote: The following CLI commands have been added to app_voicemail
voicemail show mailbox <mailbox> <context>
Show contents of mailbox <mailbox>@<context>
voicemail remove <mailbox> <context> <from_folder> <messageid>
Remove message <messageid> from <from_folder> in mailbox <mailbox>@<context>
voicemail move <mailbox> <context> <from_folder> <messageid> <to_folder>
Move message <messageid> in mailbox <mailbox>&<context> from <from_folder> to <to_folder>
voicemail forward <from_mailbox> <from_context> <from_folder> <messageid> <to_mailbox> <to_context> <to_folder>
Forward message <messageid> in mailbox <mailbox>@<context> <from_folder> to
mailbox <mailbox>@<context> <to_folder>
zhengsh [Fri, 30 Jun 2023 10:39:20 +0000 (18:39 +0800)]
res_rtp_asterisk: Move ast_rtp_rtcp_report_alloc using `rtp->themssrc_valid` into the scope of the rtp_instance lock.
From the gdb information, it was found that when calling __ast_free, the size of the
allocated space pointed to by the pointer matches the size created when rtp->themssrc_valid
is equal to 0. However, in reality, when reading the value of rtp->themssrc_valid in gdb,
it is found to be 1.
Within ast_rtcp_write(), the call to ast_rtp_rtcp_report_alloc() uses rtp->themssrc_valid,
which is outside the protection of the rtp_instance lock. However,
ast_rtcp_generate_report(), which is called by ast_rtcp_generate_compound_prefix(), uses
rtp->themssrc_valid within the protection of the rtp_instance lock.
This can lead to the possibility that the value of rtp->themssrc_valid used in the call to
ast_rtp_rtcp_report_alloc() may be different from the value of rtp->themssrc_valid used
within ast_rtcp_generate_report().
George Joseph [Tue, 11 Jul 2023 12:31:04 +0000 (06:31 -0600)]
.github: Suppress cherry-pick reminder for some situations
In PROpenedOrUpdated, the cherry-pick reminder will now be
suppressed if there are already valid 'cherry-pick-to' comments
in the PR or the PR contained a 'cherry-pick-to: none' comment.
Naveen Albert [Thu, 8 Jun 2023 22:04:31 +0000 (22:04 +0000)]
sig_analog: Allow immediate fake ring to be suppressed.
When immediate=yes on an FXS channel, sig_analog will
start fake audible ringback that continues until the
channel is answered. Even if it answers immediately,
the ringback is still audible for a brief moment.
This can be disruptive and unwanted behavior.
This adds an option to disable this behavior, though
the default behavior remains unchanged.
ASTERISK-30003 #close
Resolves: #118
UserNote: The immediatering option can now be set to no to suppress
the fake audible ringback provided when immediate=yes on FXS channels.
George Joseph [Mon, 10 Jul 2023 13:52:32 +0000 (07:52 -0600)]
app.h: Move declaration of ast_getdata_result before its first use
The ast_app_getdata() and ast_app_getdata_terminator() declarations
in app.h were changed recently to return enum ast_getdata_result
(which is how they were defined in app.c). The existing
declaration of ast_getdata_result in app.h was about 1000 lines
after those functions however so under certain circumstances,
a "use before declaration" error was thrown by the compiler.
The declaration of the enum was therefore moved to before those
functions.
Mike Bradeen [Mon, 26 Jun 2023 15:40:03 +0000 (09:40 -0600)]
app_voicemail: fix imap compilation errors
Fixes two compilation errors in app_voicemail_imap, one due to an unsed
variable and one due to a new variable added in the incorrect location
in _163.
Mike Bradeen [Tue, 23 May 2023 18:33:55 +0000 (12:33 -0600)]
utils: add lock timestamps for DEBUG_THREADS
Adds last locked and unlocked timestamps as well as a
counter for the number of times the lock has been
attempted (vs locked/unlocked) to debug output printed
using the DEBUG_THREADS option.
George Joseph [Mon, 26 Jun 2023 12:55:49 +0000 (06:55 -0600)]
rest-api: Updates for new documentation site
The new documentation site uses traditional markdown instead
of the Confluence flavored version. This required changes in
the mustache templates and the python that generates the files.
Olaf Titz [Thu, 15 Jun 2023 15:36:26 +0000 (17:36 +0200)]
app_voicemail_imap: Fix message count when IMAP server is unavailable
Some callers of __messagecount did not correctly handle error return,
instead returning a -1 message count.
This caused a notification with "Messages-Waiting: yes" and
"Voice-Message: -1/0 (0/0)" if the IMAP server was unavailable.
Ben Ford [Mon, 5 Jun 2023 19:13:16 +0000 (14:13 -0500)]
res_pjsip_session: Added new function calls to avoid ABI issues.
Added two new functions (ast_sip_session_get_dialog and
ast_sip_session_get_pjsip_inv_state) that retrieve the dialog and the
pjsip_inv_state respectively from the pjsip_inv_session on the
ast_sip_session struct. This is due to pjproject adding a new field to
the pjsip_inv_session struct that caused crashes when trying to access
fields that were no longer where they were expected to be if a module
was compiled against a different version of pjproject.
This adds an option 'force_longest_waiting_caller' which changes the
global behavior of the queue engine to prevent queue callers from
'jumping ahead' when an agent is in multiple queues.
Resolves: #108
Also closes old asterisk issues:
- ASTERISK-17732
- ASTERISK-17570
Sean Bright [Thu, 1 Jun 2023 18:53:43 +0000 (18:53 +0000)]
configure: Remove obsolete and deprecated constructs.
These were uncovered when trying to run `bootstrap.sh` with Autoconf
2.71:
* AC_CONFIG_HEADER() is deprecated in favor of AC_CONFIG_HEADERS().
* AC_HEADER_TIME is obsolete.
* $as_echo is deprecated in favor of AS_ECHO() which requires an update
to ax_pthread.m4.
Note that the generated artifacts in this commit are from Autoconf 2.69.
Sean Bright [Tue, 6 Jun 2023 13:42:58 +0000 (09:42 -0400)]
ast-db-manage: Synchronize revisions between comments and code.
In a handful of migrations, the comment header that indicates the
current and previous revisions has drifted from the identifiers
revision and down_revision variables. This updates the comment headers
to match the code.
Jaco Kroon [Fri, 2 Jun 2023 07:11:22 +0000 (09:11 +0200)]
tcptls: when disabling a server port, we should set the accept_fd to -1.
If we don't set this to -1 if the structure can be potentially re-used
later then it's possible that we'll issue a close() on an unrelated file
descriptor, breaking asterisk in other interesting ways.
I believe this to be an unlikely scenario, but it costs nothing to be
safe.
Jiajian Zhou [Fri, 19 May 2023 02:48:11 +0000 (10:48 +0800)]
AMI: Add parking position parameter to Park action
Add a parking space extension parameter (ParkingSpace) to the Park action.
Park action will attempt to park the call to that extension.
If the extension is already in use, then execution will continue at the next priority.
George Joseph [Fri, 9 Jun 2023 16:41:13 +0000 (10:41 -0600)]
test_stasis_endpoints.c: Make channel_messages more stable
The channel_messages test was assuming that stasis would return
messages in a specific order. This is an incorrect assumption as
message ordering was never guaranteed. This was causing the test
to fail occasionally. We now test all the messages for the
required message types instead of testing one by one.
George Joseph [Fri, 9 Jun 2023 14:41:32 +0000 (08:41 -0600)]
build: Fix a few gcc 13 issues
* gcc 13 is now catching when a function is declared as returning
an enum but defined as returning an int or vice versa. Fixed
a few in app.h, loader.c, stasis_message.c.
* gcc 13 is also now (incorrectly) complaining of dangling pointers
when assigning a pointer to a local char array to a char *. Had
to change that to an ast_alloca.
alex2grad [Mon, 5 Jun 2023 18:23:03 +0000 (14:23 -0400)]
app_followme: fix issue with enable_callee_prompt=no (#88)
* app_followme: fix issue with enable_callee_prompt=no
If the FollowMe option 'enable_callee_prompt' is set to 'no' then Asterisk
incorrectly sets a winner channel to the channel from which any control frame was read.
This fix sets the winner channel only to the answered channel.
Naveen Albert [Thu, 25 May 2023 23:58:41 +0000 (23:58 +0000)]
res_musiconhold: Add option to loop last file.
Adds the loop_last option to res_musiconhold,
which allows the last audio file in the directory
to be looped perpetually once reached, rather than
circling back to the beginning again.
Resolves: #122
ASTERISK-30462
UserNote: The loop_last option in musiconhold.conf now
allows the last file in the directory to be looped once reached.
Naveen Albert [Thu, 25 May 2023 23:49:35 +0000 (23:49 +0000)]
chan_dahdi: Fix Caller ID presentation for FXO ports.
Currently, the presentation for incoming channels is
always available, because it is never actually set,
meaning the channel presentation can be nonsensical.
If the presentation from the incoming Caller ID spill
is private or unavailable, we now update the channel
presentation to reflect this.
Ben Ford [Thu, 18 May 2023 21:00:16 +0000 (16:00 -0500)]
AMI: Add CoreShowChannelMap action.
Adds a new AMI action (CoreShowChannelMap) that takes in a channel name
and provides a list of all channels that are connected to that channel,
following local channel connections as well.
Naveen Albert [Thu, 18 May 2023 16:29:35 +0000 (16:29 +0000)]
sig_analog: Add fuller Caller ID support.
A previous change, ASTERISK_29991, made it possible
to send additional Caller ID parameters that were
not previously supported.
This change adds support for analog DAHDI channels
to now be able to receive these parameters for
on-hook Caller ID, in order to enhance the usability
of CPE that support these parameters.
Resolves: #94
ASTERISK-30331
UserNote: Additional Caller ID properties are now supported on
incoming calls to FXS stations, namely the
redirecting reason and call qualifier.
Joe Searle [Thu, 25 May 2023 09:58:45 +0000 (10:58 +0100)]
res_stasis.c: Add new type 'sdp_label' for bridge creation.
Add new type 'sdp_label' when creating a bridge using the ARI. This will
add labels to the SDP for each stream, the label is set to the
corresponding channel id.
Resolves: #91
UserNote: When creating a bridge using the ARI the 'type' argument now
accepts a new value 'sdp_label' which will configure the bridge to add
labels for each stream in the SDP with the corresponding channel id.
Naveen Albert [Thu, 18 May 2023 17:31:14 +0000 (17:31 +0000)]
callerid: Allow specifying timezone for date/time.
The Caller ID generation routine currently is hardcoded
to always use the system time zone. This makes it possible
to optionally specify any TZ-format time zone.
Naveen Albert [Thu, 18 May 2023 16:55:45 +0000 (16:55 +0000)]
logrotate: Fix duplicate log entries.
The Asterisk logrotate script contains explicit
references to files with the .log extension,
which are also included when *log is expanded.
This causes issues with newer versions of logrotate.
This fixes this by ensuring that a log file cannot
be referenced multiple times after expansion occurs.
Resolves: #96
ASTERISK-30442
Reported by: EN Barnett
Tested by: EN Barnett
chan_pjsip: Allow topology/session refreshes in early media state
With this change, session modifications in the early media state are
possible if the SDP was sent reliably and confirmed by a PRACK. For
details, see RFC 6337, escpecially section 3.2.
projects / thirdparty / asterisk.git / log
