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.
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.
Resolves: #120
ASTERISK-30333
ASTERISK-21741
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.
Resolves: #104
UserNote: New AMI action CoreShowChannelMap has been added.
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.
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.
When Asterisk is restarted it does not preserve paused reason for
members of realtime queues. This was fixed for non-realtime queues in
ASTERISK_25732
Resolves: #66
UpgradeNote: Add a new column to the queue_member table:
reason_paused VARCHAR(80) so the reason can be preserved.
UserNote: Make paused reason in realtime queues persist an
Asterisk restart. This was fixed for non-realtime
queues in ASTERISK_25732.
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.
Resolves: #98
ASTERISK-30330
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.
Resolves: #73
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.
Resolves: #100
ASTERISK-30349 #close
Commit 09e989f972
categorized the T option as not being compatible
with remote consoles, but they do affect verbose
messages with remote console. This fixes this.
Resolves: #102
This enables the test to work with CC=clang.
Without this the test for 6 args would fail with:
utils.c:99:12: error: static declaration of 'gethostbyname_r' follows non-static declaration
static int gethostbyname_r (const char *name, struct hostent *ret, char *buf,
^
/usr/include/netdb.h:177:12: note: previous declaration is here
extern int gethostbyname_r (const char *__restrict __name,
^
Fixing the expected return type to int sorts this out.
Signed-off-by: Jaco Kroon <jaco@uls.co.za>
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.
Resolves: #65
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.
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.
* Remove .gitreview and switch to pulling the main asterisk branch
version from configure.ac instead.
* Replace references to JIRA with GitHub.
* Other minor cleanup found along the way.
Resolves: #39
When using mediasec, requests sent after a 401 must still contain the
Security-Client header according to
draft-dawes-sipcore-mediasec-parameter.
Resolves: #48
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.
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.
ASTERISK-28109 #close
ASTERISK-28233 #close
Change-Id: I2e28e5a5d007bd49e3df113ad567b011103899bf
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.
ASTERISK-30407 #close
Change-Id: Iab76d39447c4b8cf133de85657dba02fda07f9a2
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.
ASTERISK-30428
Change-Id: I5675df11e739be5226b328f8828d4b8d81fbefb4
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.
ASTERISK-30479 #close
Change-Id: I3bf7d275849ab83f55f7fb6702a75a3077ee1df3
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.
ASTERISK-30486 #close
Change-Id: I6706cea708b5cc781f59f8652c2cb377e55aed7e
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.
Change-Id: Ia137814b5aeb0e1a44c75034bd3615c26021da69
