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.
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.
Naveen Albert [Thu, 18 May 2023 17:37:19 +0000 (17:37 +0000)]
chan_dahdi: Fix broken hidecallerid setting.
The hidecallerid setting in chan_dahdi.conf currently
is broken for a couple reasons.
First, the actual code in sig_analog to "allow" or "block"
Caller ID depending on this setting improperly used
ast_set_callerid instead of updating the presentation.
This issue was mostly fixed in ASTERISK_29991, and that
fix is carried forward to this code as well.
Secondly, the hidecallerid setting is set on the DAHDI
pvt but not carried forward to the analog pvt properly.
This is because the chan_dahdi config loading code improperly
set permhidecallerid to permhidecallerid from the config file,
even though hidecallerid is what is actually set from the config
file. (This is done correctly for call waiting, a few lines above.)
This is fixed to read the proper value.
Thirdly, in sig_analog, hidecallerid is set to permhidecallerid
only on hangup. This can lead to potential security vulnerabilities
as an allowed Caller ID from an initial call can "leak" into subsequent
calls if no hangup occurs between them. This is fixed by setting
hidecallerid to permcallerid when calls begin, rather than when they end.
This also means we don't need to also set hidecallerid in chan_dahdi.c
when copying from the config, as we would have to otherwise.
Fourthly, sig_analog currently only allows dialing *67 or *82 if
that would actually toggle the presentation. A comment is added
clarifying that this behavior is okay.
Finally, a couple log messages are updated to be more accurate.
Naveen Albert [Thu, 18 May 2023 18:57:34 +0000 (18:57 +0000)]
asterisk.c: Fix option warning for remote console.
Commit 09e989f972e2583df4e9bf585c246c37322d8d2f
categorized the T option as not being compatible
with remote consoles, but they do affect verbose
messages with remote console. This fixes this.
Sean Bright [Tue, 9 May 2023 13:27:33 +0000 (09:27 -0400)]
xml.c: Process XML Inclusions recursively.
If processing an XInclude results in new <xi:include> elements, we
need to run XInclude processing again. This continues until no
replacement occurs or an error is encountered.
There is a separate issue with dynamic strings (ast_str) that will be
addressed separately.
George Joseph [Thu, 6 Jul 2023 12:37:50 +0000 (06:37 -0600)]
apply_patches: Sort patch list before applying
The apply_patches script wasn't sorting the list of patches in
the "patches" directory before applying them. This left the list
in an indeterminate order. In most cases, the list is actually
sorted but rarely, they can be out of order and cause dependent
patches to fail to apply.
We now sort the list but the "sort" program wasn't in the
configure scripts so we needed to add that and regenerate
the scripts as well.
chan_pjsip: also return all codecs on empty re-INVITE for late offers
We should also return all codecs on an re-INVITE without SDP for a
call that used late offer (e.g. no SDP in the initial INVITE, SDP
in the ACK). Bugfix for feature introduced in ASTERISK-30193
(https://issues.asterisk.org/jira/browse/ASTERISK-30193)
Migration from previous gerrit change that was not merged.
Mike Bradeen [Tue, 2 May 2023 15:20:22 +0000 (09:20 -0600)]
cel: add local optimization begin event
The current AST_CEL_LOCAL_OPTIMIZE event is and has been
triggered on a local optimization end to serve as a flag
indicating the event occurred. This change adds a second
AST_CEL_LOCAL_OPTIMIZE_BEGIN event for further detail.
Resolves: #52
UpgradeNote: The existing AST_CEL_LOCAL_OPTIMIZE can continue
to be used as-is and the AST_CEL_LOCAL_OPTIMIZE_BEGIN event
can be ignored if desired.
UserNote: The new AST_CEL_LOCAL_OPTIMIZE_BEGIN can be used
by itself or in conert with the existing
AST_CEL_LOCAL_OPTIMIZE to book-end local channel optimizaion.
InterLinked1 [Tue, 2 May 2023 14:05:12 +0000 (09:05 -0500)]
chan_dahdi: Add dialmode option for FXS lines. (#36)
Currently, both pulse and tone dialing are always enabled
on all FXS lines, with no way of disabling one or the other.
In some circumstances, it is desirable or necessary to
disable one of these, and this behavior can be problematic.
A new "dialmode" option is added which allows setting the
methods to support on a per channel basis for FXS (FXO
signalled lines). The four options are "both", "pulse",
"dtmf"/"tone", and "none".
Additionally, integration with the CHANNEL function is
added so that this setting can be updated for a channel
during a call.
Resolves: #35
ASTERISK-29992
UserNote: A "dialmode" option has been added which allows
specifying, on a per-channel basis, what methods of
subscriber dialing (pulse and/or tone) are permitted.
Additionally, this can be changed on a channel
at any point during a call using the CHANNEL
function.
Naveen Albert [Fri, 9 Dec 2022 14:18:26 +0000 (14:18 +0000)]
pbx_dundi: Add PJSIP support.
Adds PJSIP as a supported technology to DUNDi.
To facilitate this, we now allow an endpoint to be specified
for outgoing PJSIP calls. We also allow users to force a specific
channel technology for outgoing SIP-protocol calls.
Naveen Albert [Fri, 17 Feb 2023 13:45:16 +0000 (13:45 +0000)]
res_pjsip_stir_shaken: Fix JSON field ordering and disallowed TN characters.
The current STIR/SHAKEN signing process is inconsistent with the
RFCs in a couple ways that can cause interoperability issues.
RFC8225 specifies that the keys must be ordered lexicographically, but
currently the fields are simply ordered according to the order
in which they were added to the JSON object, which is not
compliant with the RFC and can cause issues with some carriers.
To fix this, we now leverage libjansson's ability to dump a JSON
object sorted by key value, yielding the correct field ordering.
Additionally, telephone numbers must have any leading + prefix removed
and must not contain characters outside of 0-9, *, and # in order
to comply with the RFCs. Numbers are now properly formatted as such.
chan_pjsip: fix music on hold continues after INVITE with replaces
In a three party scenario with INVITE with replaces, we need to
unhold the call, otherwise one party continues to get music on
hold, and the call is not properly bridged between them.
Naveen Albert [Tue, 28 Mar 2023 13:37:55 +0000 (13:37 +0000)]
voicemail.conf: Fix incorrect comment about #include.
A comment at the top of voicemail.conf says that #include
cannot be used in voicemail.conf because this breaks
the ability for app_voicemail to auto-update passwords.
This is factually incorrect, since Asterisk has no problem
updating files that are #include'd in the main configuration
file, and this does work in voicemail.conf as well.
Naveen Albert [Mon, 3 Apr 2023 14:16:59 +0000 (14:16 +0000)]
app_queue: Fix minor xmldoc duplication and vagueness.
The F option in the xmldocs for the Queue application
was erroneously duplicated, causing it to display
twice on the wiki. The two sections are now merged into one.
Additionally, the description for the d option was quite
vague. Some more details are added to provide context
as to what this actually does.
George Joseph [Tue, 28 Mar 2023 14:07:09 +0000 (08:07 -0600)]
test.c: Fix counting of tests and add 2 new tests
The unit test XML output was counting all registered tests as "run"
even when only a subset were actually requested to be run and
the "failures" attribute was missing.
* The "tests" attribute of the "testsuite" element in the
output XML now reflects only the tests actually requested
to be executed instead of all the tests registered.
* The "failures" attribute was added to the "testsuite"
element.
Also added 2 new unit tests that just pass and fail to be
used for CI testing.
Sean Bright [Tue, 21 Mar 2023 13:24:59 +0000 (09:24 -0400)]
ael: Regenerate lexers and parsers.
Various changes to ensure that the lexers and parsers can be correctly
generated when REBUILD_PARSERS is enabled.
Some notes:
* Because of the version of flex we are using to generate the lexers
(2.5.35) some post-processing in the Makefile is still required.
* The generated lexers do not contain the problematic C99 check that
was being replaced by the call to sed in the respective Makefiles so
it was removed.
* Since these files are generated, they will include trailing
whitespace in some places. This does not need to be corrected.
Mike Bradeen [Wed, 1 Mar 2023 22:39:15 +0000 (15:39 -0700)]
bridge_builtin_features: add beep via touch variable
Add periodic beep option to one-touch recording by setting
the touch variable TOUCH_MONITOR_BEEP or
TOUCH_MIXMONITOR_BEEP to the desired interval in seconds.
If the interval is less than 5 seconds, a minimum of 5
seconds will be imposed. If the interval is set to an
invalid value, it will default to 15 seconds.
A new test event PERIODIC_HOOK_ENABLED was added to the
func_periodic_hook hook_on function to indicate when
a hook is started. This is so we can test that the touch
variable starts the hook as expected.
Mike Bradeen [Mon, 13 Mar 2023 19:27:06 +0000 (13:27 -0600)]
res_mixmonitor: MixMonitorMute by MixMonitor ID
While it is possible to create multiple mixmonitor instances
on a channel, it was not previously possible to mute individual
instances.
This change includes the ability to specify the MixMonitorID
when calling the manager action: MixMonitorMute. This will
allow an individual MixMonitor instance to be muted via id.
This id can be stored as a channel variable using the 'i'
MixMonitor option.
As part of this change, if no MixMonitorID is specified in
the manager action MixMonitorMute, Asterisk will set the mute
flag on all MixMonitor spy-type audiohooks on the channel.
This is done via the new audiohook function:
ast_audiohook_set_mute_all.
This newly introduced periodic-announce-startdelay makes it possible to
configure the initial start delay of the first periodic announcement
after which periodic-announce-frequency takes over.
Naveen Albert [Sat, 4 Mar 2023 00:09:23 +0000 (00:09 +0000)]
app_dial: Fix DTMF not relayed to caller on unanswered calls.
DTMF frames are not handled in app_dial when sent towards the
caller. This means that if DTMF is sent to the calling party
and the call has not yet been answered, the DTMF is not audible.
This is now fixed by relaying DTMF frames if only a single
destination is being dialed.
George Joseph [Mon, 13 Mar 2023 19:35:07 +0000 (13:35 -0600)]
make_version: Strip svn stuff and suppress ref HEAD errors
* All of the code that used subversion has been removed.
* When Asterisk is checked out from a tag or commit instead
of one of the regular branches, git would emit messages like
"fatal: ref HEAD is not a symbolic ref" which weren't fatal
at all. Those are now suppressed.
Mike Bradeen [Mon, 6 Mar 2023 20:52:18 +0000 (13:52 -0700)]
cli: increase channel column width
For 'core show channels', the Channel name field is increased
to 64 characters and the Location name field is increased to
32 characters.
For 'core show channels verbose', the Channel name field is
increased to 80 characters, the Context is increased to 24
characters and the Extension is increased to 24 characters.
`rc.archlinux.asterisk`, which explicitly requests bash in its
shebang, uses the following command syntax:
${DAEMON} -rx "core stop now" > /dev/null 2&>1
The intent of which is to execute:
${DAEMON} -rx "core stop now"
While sending both stdout and stderr to `/dev/null`. Unfortunately,
because the `&` is in the wrong place, bash is interpreting the `2` as
just an additional argument to the `$DAEMON` command and not as a file
descriptor and proceeds to use the bashism `&>` to send stderr and
stdout to a file named `1`.
So we clean it up and just use bash's shortcut syntax.
Issue raised and a fix suggested (but not used) by peutch on GitHub¹.
Fabrice Fontaine [Sat, 25 Feb 2023 10:27:12 +0000 (11:27 +0100)]
main/iostream.c: fix build with libressl
Fix the following build failure with libressl by using SSL_is_server
which is available since version 2.7.0 and
https://github.com/libressl-portable/openbsd/commit/d7ec516916c5eaac29b02d7a8ac6570f63b458f7:
iostream.c: In function 'ast_iostream_close':
iostream.c:559:41: error: invalid use of incomplete typedef 'SSL' {aka 'struct ssl_st'}
559 | if (!stream->ssl->server) {
| ^~
ASTERISK-30107 #close
Fixes: - http://autobuild.buildroot.org/results/ce4d62d00bb77ba5b303cacf6be7e350581a62f9
Change-Id: Iea7f34970297f2fb50285d73462d0174ba7e9587
(cherry picked from commit f34d9d56189e9ad9663ddf4e20f350c6321b2993)
Mike Bradeen [Wed, 29 Mar 2023 18:49:51 +0000 (12:49 -0600)]
res_pjsip_pubsub: subscription cleanup changes
There are two main parts of the change associated with this
commit. These are driven by the change in call order of
pubsub_on_rx_refresh and pubsub_on_evsub_state by pjproject
when an in-dialog SUBSCRIBE is received.
First, the previous behavior was for pjproject to call
pubsub_on_rx_refresh before calling pubsub_on_evsub_state
when an in-dialog SUBSCRIBE was received that changes the
subscription state.
If that change was a termination due to a re-SUBSCRIBE with
an expires of 0, we used to use the call to pubsub_on_rx_refresh
to set the substate of the evsub to TERMINATE_PENDING before
pjproject could call pubsub_on_evsub_state.
This substate let pubsub_on_evsub_state know that the
subscription TERMINATED event could be ignored as there was
still a subsequent NOTIFY that needed to be generated and
another call to pubsub_on_evsub_state to come with it.
That NOTIFY was sent via serialized_pubsub_on_refresh_timeout
which would see the TERMINATE_PENDING state and transition it
to TERMINATE_IN_PROGRESS before triggering another call to
pubsub_on_evsub_state (which now would clean up the evsub.)
The new pjproject behavior is to call pubsub_on_evsub_state
before pubsub_on_rx_refresh. This means we no longer can set
the state to TERMINATE_PENDING to tell pubsub_on_evsub_state
that it can ignore the first TERMINATED event.
To handle this, we now look directly at the event type,
method type and the expires value to determine whether we
want to ignore the event or use it to trigger the evsub
cleanup.
Second, pjproject now expects the NOTIFY to actually be sent
during pubsub_on_rx_refresh and avoids the protocol violation
inherent in sending a NOTIFY before the SUBSCRIBE is
acknowledged by caching the sent NOTIFY then sending it
after responding to the SUBSCRIBE.
This requires we send the NOTIFY using the non-serialized
pubsub_on_refresh_timeout directly and let pjproject handle
the protocol violation.
George Joseph [Thu, 16 Feb 2023 16:05:30 +0000 (09:05 -0700)]
res_pjsip: Replace invalid UTF-8 sequences in callerid name
* Added a new function ast_utf8_replace_invalid_chars() to
utf8.c that copies a string replacing any invalid UTF-8
sequences with the Unicode specified U+FFFD replacement
character. For example: "abc\xffdef" becomes "abc\uFFFDdef".
Any UTF-8 compliant implementation will show that character
as a � character.
* Updated res_pjsip:set_id_from_hdr() to use
ast_utf8_replace_invalid_chars and print a warning if any
invalid sequences were found during the copy.
* Updated stasis_channels:ast_channel_publish_varset to use
ast_utf8_replace_invalid_chars and print a warning if any
invalid sequences were found during the copy.
Naveen Albert [Wed, 14 Dec 2022 16:00:51 +0000 (16:00 +0000)]
chan_iax2: Fix jitterbuffer regression prior to receiving audio.
ASTERISK_29392 (a security fix) introduced a regression by
not processing frames when we don't have an audio format.
Currently, chan_iax2 only calls jb_get to read frames from
the jitterbuffer when the voiceformat has been set on the pvt.
However, this only happens when we receive a voice frame, which
means that prior to receiving voice frames, other types of frames
get stalled completely in the jitterbuffer.
To fix this, we now fallback to using the format negotiated during
call setup until we've actually received a voice frame with a format.
This ensures we're always able to read from the jitterbuffer.
Nick French [Sat, 11 Feb 2023 12:58:43 +0000 (06:58 -0600)]
pjproject_bundled: Fix cross-compilation with SSL libs.
Asterisk makefiles auto-detect SSL library availability,
then they assume that pjproject makefiles will also autodetect
an SSL library at the same time, so they do not pass on the
autodetection result to pjproject.
This normally works, except the pjproject makefiles disables
autodetection when cross-compiling.
Fix by explicitly configuring pjproject to use SSL if we
have been told to use it or it was autodetected
cmaj [Sun, 8 Jan 2023 05:04:57 +0000 (22:04 -0700)]
res_phoneprov.c: Multihomed SERVER cache prevention
Phones moving between subnets on multi-homed server have their
initially connected interface IP cached in the SERVER variable,
even when it is not specified in the configuration files. This
prevents phones from obtaining the correct SERVER variable value
when they move to another subnet.
Mike Bradeen [Tue, 21 Feb 2023 20:25:28 +0000 (13:25 -0700)]
res_pjsip: Prevent SEGV in pjsip_evsub_send_request
contributed pjproject - patch to check sub->pending_notify
in evsub.c:on_tsx_state before calling
pjsip_evsub_send_request()
res_pjsip_pubsub - change post pjsip 2.13 behavior to use
pubsub_on_refresh_timeout to avoid the ao2_cleanup call on
the sub_tree. This is is because the final NOTIFY send is no
longer the last place the sub_tree is referenced.
Mike Bradeen [Mon, 6 Feb 2023 15:54:56 +0000 (08:54 -0700)]
app_senddtmf: Add option to answer target channel.
Adds a new option to SendDTMF() which will answer the specified
channel if it is not already up. If no channel is specified, the
current channel will be answered instead.
Mike Bradeen [Fri, 20 Jan 2023 22:50:44 +0000 (15:50 -0700)]
res_pjsip: Upgraded bundled pjsip to 2.13
Removed multiple patches.
Code chages in res_pjsip_pubsub due to changes in evsub.
Pjsip now calls on_evsub_state() before on_rx_refresh(),
so the sub tree deletion that used to take place in
on_evsub_state() now must take place in on_rx_refresh().
Additionally, pjsip now requires that you send the NOTIFY
from within on_rx_refresh(), otherwise it will assert
when going to send the 200 OK. The idea is that it will
look for this NOTIFY and cache it until after sending the
response in order to deal with the self-imposed message
mis-order. Asterisk previously dealt with this by pushing
the NOTIFY in on_rx_refresh(), but pjsip now forces us
to use it's method.
Changes were required to configure in order to detect
which way pjsip handles this as the two are not
compatible for the reasons mentioned above.
A corresponding change in testsuite is required in order
to deal with the small interal timing changes caused by
moving the NOTIFY send.
Naveen Albert [Thu, 6 Jan 2022 22:11:44 +0000 (22:11 +0000)]
app_signal: Add signaling applications
Adds the Signal and WaitForSignal
applications, which can be used for inter-channel
signaling in the dialplan.
Signal supports sending a signal to other channels
listening for a signal of the same name, with an
optional data payload. The signal is received by
all channels waiting for that named signal.
Naveen Albert [Sat, 12 Feb 2022 21:59:52 +0000 (21:59 +0000)]
func_json: Enhance parsing capabilities of JSON_DECODE
Adds support for arrays to JSON_DECODE by allowing the
user to print out entire arrays or index a particular
key or print the number of keys in a JSON array.
Additionally, adds support for recursively iterating a
JSON tree in a single function call, making it easier
to parse JSON results with multiple levels. A maximum
depth is imposed to prevent potentially blowing
the stack.
Also fixes a bug with the unit tests causing an empty
string to be printed instead of the actual test result.
Naveen Albert [Thu, 13 Oct 2022 13:45:26 +0000 (13:45 +0000)]
res_pjsip_session: Add overlap_context option.
Adds the overlap_context option, which can be used
to explicitly specify a context to use for overlap
dialing extension matches, rather than forcibly
using the context configured for the endpoint.
projects / thirdparty / asterisk.git / log
