Removes some leftover build and config references to
modules that have since been removed from Asterisk.
ASTERISK-29935 #close
Change-Id: Iaefc73a23f4b2de3c6c14d928050135b6d0ef6af
When adding headers to an outgoing request the headers were cloned using
the dialog's pool when they should have been cloned using tdata's pool.
Under certain circumstances it was possible for the dialog object, and
its pool to be freed while tdata is still active and available. Thus the
cloned header "disappeared", and when tdata tried to later access it a
crash would occur.
This patch makes it so all added headers are cloned appropriately using
tdata's pool.
ASTERISK-29411 #close
ASTERISK-29535 #close
Change-Id: I9852025b5ee93ce1c038209150ee9dba1e0767c5
Several modules removal and deprecations occurred in 19.0.0 (initial
19 release), but associated UPGRADE files were not removed from
staging for some reason in the master branch.
This patch removes those files, and also removes a spurious leftover
header, chan_phone.h (associated module removed in 19).
Change-Id: Ib92142c846b45c882d6b2b6caca7225253c83add
This change removes patches which have been merged into
upstream and updates some existing ones. It also adds
some additional config_site.h changes to restore previous
behavior, as well as a patch to allow multiple Authorization
headers. There seems to be some confusion or disagreement
on language in RFC 8760 in regards to whether multiple
Authorization headers are supported. The RFC implies it
is allowed, as does some past sipcore discussion. There is
also the catch all of "local policy" to allow it. In
the case of Asterisk we allow it.
ASTERISK-29351
Change-Id: Id39ece02dedb7b9f739e0e37ea47d76854af7191
The PBX core uses the stack when it comes to includes, which
means that a context can only contain strictly fewer than
AST_PBX_MAX_STACK includes. If this is exceeded, then warnings
will be emitted for each number of includes beyond this if
searching for an extension in the including context, and if
the extension's inclusion is beyond the stack size, it will
simply not be found.
To address this, we now check if there are too many includes
in a context when the dialplan is reloaded so that if there
is an issue, the user is aware of at "compile time" as opposed
to "run time" only. Secondly, more details are printed out
when this message is encountered so it's clear what has happened.
ASTERISK-26719
Change-Id: Ia3700452e75a7af3391b3e82ee69f06a669f8958
make_xml_documentation was being called with the --validate
flag set when it shouldn't have been. This was causing
build failures if neither xmllint nor xmlstarlet were installed.
The correct behavior is to simply print a message that either
one of those tools should be installed for validation and
continue with the build.
ASTERISK-29988
Change-Id: Idc6c44114e7dd3fadae183a4e22f4fdba0b8a645
get_sourceable_makeopts wasn't handling variables with embedded
double quotes in them very well. One example was the DOWNLOAD
variable when curl was being used instead of wget. Rather than
trying to fix get_sourceable_makeopts, it's just been removed.
ASTERISK-29986
Reported by: Stefan Ruijsenaars
Change-Id: Idf2a90902228c2558daa5be7a4f8327556099cd2
The iax2 show netstats command previously didn't contain
enough spacing in the header to properly align the table
header with the table body. This caused column headers
to not align with the values on longer channel names.
Some spacing is added to account for the longest channel
names that display (before truncation occurs) so that
columns are always properly aligned.
ASTERISK-29895 #close
patches:
61205_misaligned2.patch submitted by Birger Harzenetter (license 5870)
Change-Id: I450ce6bb81157b9d6d149007e53b749f237b6d9f
There is work going on to update our OpenSSL usage to avoid the
deprecated functions but in the meantime make it possible to compile
in devmode.
Change-Id: Ib082eb8b3751f0185d8aa8fe127da664c93f0726
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
ASTERISK_22025 introduced a regression that shows
the host IP and port as the perceived IP and port
again, as opposed to showing the actual perceived
address. This fixes this by showing the correct
information.
ASTERISK-29048 #close
Change-Id: I0ad3e25bc6b449e83ce72ea5d1a1cdba72aa304a
Change RTP timer behavior for detecting RTP only after two-way
SDP channel establishment. Ignore detecting after receiving 183
with SDP or while direct media is used.
Make rtp_timeout and rtp_timeout_hold options consistent to rtptimeout
and rtpholdtimeout options in chan_sip.
ASTERISK-26689 #close
ASTERISK-29929 #close
Change-Id: I07326d5b9c40f25db717fd6075f6f3a8d77279eb
Use pkg-config to detect libxml2, falling back to xml2-config if the
former is not available.
This patch ensures Asterisk continues to build on systems without
xml2-config installed.
The patch also updates the associated 'configure' files.
ASTERISK-29970 #close
Change-Id: I3c90dfe0b0590486cbb8e6d426a7c5c4199410c0
Treat time_t's as entirely unique and use the POSIX API's for
converting to/from strings.
Lastly, a 64-bit integer formats as 20 digits at most in base10.
Don't need to have any 100 byte buffers to hold that.
ASTERISK-29674 #close
Signed-off-by: Philip Prindeville <philipp@redfish-solutions.com>
Change-Id: Id7b25bdca8f92e34229f6454f6c3e500f2cd6f56
When asterisk generates the RLMI part of NOTIFY request,
the asterisk uses the local contact uri instead of the URI to which
the SUBSCRIBE request is sent.
Because of this mismatch some IP phones (for example Cisco 5XX) ignore
this list.
According
https://datatracker.ietf.org/doc/html/rfc4662#section-5.2
The first mandatory <list> attribute is "uri", which contains the uri
that corresponds to the list. Typically, this is the URI to which
the SUBSCRIBE request was sent.
https://datatracker.ietf.org/doc/html/rfc4662#section-5.3
The "uri" attribute identifies the resource to which the <resource>
element corresponds. Typically, this will be a SIP URI that, if
subscribed to, would return the state of the resource.
This patch makes asterisk to generate URI using SUBSCRIBE request URI.
ASTERISK-29961 #close
Change-Id: I1fcfc08fd589677f40608c59a4e143c45ee05f6c
Adds documentation for all of the possible return values
for the DIALSTATUS variable in the Dial application.
ASTERISK-25716
Change-Id: Id22593f1f1f7ea86e5734cee49516ec50848e8c0
Using the length of a file found on the filesystem rather than the
file being requested could result in filenames whose names are
substrings of another to be erroneously matched.
We now ensure a complete comparison before returning a positive
result.
ASTERISK-29960 #close
Change-Id: Id3ffc77681b9b75b8569062f3d952a128a21c71a
Passing 0 as the last argument to strtoimax() or strtoumax() causes
octal and hexadecimal to be accepted which was not originally
intended. So we now force to only accept decimal.
ASTERISK-29950 #close
Change-Id: I93baf0f273441e8280354630a463df263a8c0edd
MUSL defines BUFSIZ as 1024 which is not reasonable for log messages.
More broadly, BUFSIZ is the amount of buffering stdio.h does, which
is arbitrary and largely orthogonal to what logging should accept
as the maximum message size.
ASTERISK-29928
Signed-off-by: Philip Prindeville <philipp@redfish-solutions.com>
Change-Id: Iaa49fbbab029c64ae3d95e4b18270e0442cce170
BackGround and WaitExten both accept options that are not
currently documented. This adds documentation for these
options to the xml documentation for each application.
ASTERISK-29967 #close
Change-Id: If812a9f1ccbba3e4d427a0e7a6dea923c2f905f7
This patch makes the Resource List Subscriptions (RLS) dynamic.
The asterisk updates the current subscriptions to reflect the changes
to the list on the subscriptions refresh. If list items are added,
removed, updated or do not exist anymore, the asterisk regenerates
the resource list.
ASTERISK-29906 #close
Change-Id: Icee8c00459a7aaa43c643d77ce6f16fb7ab037d3
The XML documentation for the SET MUSIC AGI
command is invalid, as the parameter does not
have a name and the on/off enum options for
the on/off argument are listed separately, which
is incorrect. The cumulative effect of these currently
is that the Asterisk Wiki documentation for SET MUSIC
is broken and external documentation generators crash
on SET MUSIC due to the malformed documentation.
These issues are corrected so that the documentation
can be successfully parsed as with other similar AGI
commands.
ASTERISK-29939 #close
ASTERISK-28891 #close
Change-Id: I8c3d59897531bcbc401cbc7b00c9e2829dcb35f8
Omit "unsupported column type 'text'" warning in logs while
using text-type column in the PgSQL backend.
ASTERISK-29924 #close
Change-Id: I48061a7d469426859670db07f1ed8af1eb814712
This adds a new AMI action called QueueWithdrawCaller.
This AMI action makes it possible to withdraw a caller from a queue,
in a safe and a generic manner.
This can be useful for retrieving a specific call and
dispatching it to a specific extension.
It works by signaling the caller to exit the queue application
whenever it can. Therefore, it is not guaranteed
that the call will leave the queue.
ASTERISK-29909 #close
Change-Id: Ic15aa238e23b2884abdcaadff2fda7679e29b7ec
ASTERISK_29853 added the ability to selectively disable
AMI events on a global basis, but the logic for this uses
strstr which means that events with names which are the prefix
of another event, if disabled, could disable those events as
well.
Instead, we account for this possibility to prevent this
undesired behavior from occuring.
ASTERISK_29853
Change-Id: Icccd1872602889806740971e4adf932f92466959
Added functions to open, close, and apply XML Stylesheets
to XML documents. Although the presence of libxslt was already
being checked by configure, it was only happening if xmldoc was
enabled. Now it's checked regardless.
Added ability to parse a string consisting of comma separated
name/value pairs into an ast_variable list. The reverse of
ast_variable_list_join().
Change-Id: I1e1d149be22165a1fb8e88e2903a36bba1a6cf2e
Added the missing xml-stylesheet and Xinclude namespace
declarations in pjsip_config.xml and pjsip_manager.xml.
Updated make_xml_documentation to show detailed errors when
xmlstarlet is the validator. It's now run once with the '-q'
option to suppress harmless/expected messages and if it actually
fails, it's run again without '-q' but with '-e' to show
the actual errors.
Change-Id: I4bdc9d2ea6741e8d2e5eb82df60c68ccc59e1f5e
Added:
Replace a variable in a list:
int ast_variable_list_replace_variable(struct ast_variable **head,
struct ast_variable *old, struct ast_variable *new);
Added test as well.
Create a "name=value" string from a variable list:
'name1="val1",name2="val2"', etc.
struct ast_str *ast_variable_list_join(
const struct ast_variable *head, const char *item_separator,
const char *name_value_separator, const char *quote_char,
struct ast_str **str);
Added test as well.
Allow the name of an XML element to be changed.
void ast_xml_set_name(struct ast_xml_node *node, const char *name);
Change-Id: I330a5f63dc0c218e0d8dfc0745948d2812141ccb
Moved the xmldoc build logic from the top-level Makefile into
its own script "make_xml_documentation" in the build_tools
directory.
Created a new utility script "get_sourceable_makeopts", also in
the build_tools directory, that dumps the top-level "makeopts"
file in a format that can be "sourced" from shell sscripts.
This allows scripts to easily get the values of common make
build variables such as the location of the GREP, SED, AWK, etc.
utilities as well as the AST* and library *_LIB and *_INCLUDE
variables.
Besides moving logic out of the Makefile, some optimizations
were done like removing "third-party" from the list of
subdirectories to be searched for documentation and changing some
assignments from "=" to ":=" so they're only evaluated once.
The speed increase is noticeable.
The makeopts.in file was updated to include the paths to
REALPATH and DIRNAME. The ./conifgure script was setting them
but makeopts.in wasn't including them.
So...
With this change, you can now place documentation in any"c"
source file AND you can now place it in a separate XML file
altogether. The following are examples of valid locations:
res/res_pjsip.c
Using the existing /*** DOCUMENTATION ***/ fragment.
res/res_pjsip/pjsip_configuration.c
Using the existing /*** DOCUMENTATION ***/ fragment.
res/res_pjsip/pjsip_doc.xml
A fully-formed XML file. The "configInfo", "manager",
"managerEvent", etc. elements that would be in the "c"
file DOCUMENTATION fragment should be wrapped in proper
XML. Example for "somemodule.xml":
<?xml version="1.0" encoding="UTF-8"?>
<!DOCTYPE docs SYSTEM "appdocsxml.dtd">
<docs>
<configInfo>
...
</configInfo>
</docs>
It's the "appdocsxml.dtd" that tells make_xml_documentation
that this is a documentation XML file and not some other XML file.
It also allows many XML-capable editors to do formatting and
validation.
Other than the ".xml" suffix, the name of the file is not
significant.
As a start... This change also moves the documentation that was
in res_pjsip.c to 2 new XML files in res/res_pjsip:
pjsip_config.xml and pjsip_manager.xml. This cut the number of
lines in res_pjsip.c in half. :)
Change-Id: I486c16c0b5a44d7a8870008e10c941fb19b71ade
Recap from earlier commit: If you have a development branch for a
major project that will receive gerrit reviews it'll probably be
named something like "development/16/newproject" or a work branch
based on that "development" branch. That will necessitate
setting "defaultbranch=development/16/newproject" in .gitreview.
The make_version script uses that variable to construct the
asterisk version however, which results in versions
like "GIT-development/16/newproject-ee582a8c7b" which is probably
not what you want. It also constructs the URLs for downloading
external modules with that version, which will fail.
Fast-forward:
The earlier attempt at adding a "basebranch" variable to
.gitreview didn't work out too well in practice because changes
were made to .gitreview, which is a checked-in file. So, if
you wanted to rebase your work branch on the base branch, rebase
would attempt to overwrite your .gitreview with the one from
the base branch and complain about a conflict.
This is a slighltly different approach that adds three methods to
determine the mainline branch:
1. --- MAINLINE_BRANCH from the environment
If MAINLINE_BRANCH is already set in the environment, that will
be used. This is primarily for the Jenkins jobs.
2. --- .develvars
Instead of storing the basebranch in .gitreview, it can now be
stored in a non-checked-in ".develvars" file and keyed by the
current branch. So, if you were working on a branch named
"new-feature-work" based on "development/16/new-feature" and wanted
to push to that branch in Gerrit but wanted to pull the external
modules for 16, you'd create the following .develvars file:
[branch "new-feature-work"]
mainline-branch = 16
The .gitreview file would still look like:
[gerrit]
defaultbranch=development/16/new-feature
...which would cause any reviews pushed from "new-feature-work" to
go to the "development/16/new-feature" branch in Gerrit.
The key is that the .develvars file is NEVER checked in (it's been
added to .gitignore).
3. --- Well Known Development Branch
If you're actually working in a branch named like
"development/<mainline_branch>/some-feature", the mainline branch
will be parsed from it.
4. --- .gitreview
If none of the earlier conditions exist, the .gitreview
"defaultbranch" variable will be used just as before.
Change-Id: I1cdeeaa0944bba3f2e01d7a2039559d0c266f8c9
Adds the lastcontext and lastexten channel fields to allow users
to access previous dialplan execution locations.
ASTERISK-29840 #close
Change-Id: Ib455fe300cc8e9a127686896ee2d0bd11e900307
Although there are 10 debugs levels, over time,
many current debug calls have come to use
inappropriately low debug levels. In particular,
a select few debug calls (currently all debug 1)
can result in thousands of debug messages per minute
for a single call.
This can adds a lot of noise to core debug
which dilutes the value in having different
debug levels in the first place, as these
log messages are from the core internals are
are better suited for higher debug levels.
Some debugs levels are thus adjusted so that
debug level 1 is not inappropriately overloaded
with these extremely high-volume and general
debug messages.
ASTERISK-29897 #close
Change-Id: I55a71598993552d3d64a401a35ee99474770d4b4
pbx.digium.com no longer accepts IAX2 calls and
there are no plans for it to come back.
Accordingly, nonworking IAX2 URIs are removed from
both the LICENSE file and the sample config.
ASTERISK-29923 #close
Change-Id: I257c54d4d812ed6b4bd4cbec2cd7ebe2b87b5bad
Adds the since tag to the documentation DTD so
that individual applications, functions, etc.
can now specify when they were added to Asterisk.
This tag is added at the individual application,
function, etc. level as opposed to at the module
level because modules can expand over time as new
functionality is added, and granularity only
to the module level would generally not be useful.
This enables the ability to more easily determine
when new functionality was added to Asterisk, down
to minor version as opposed to just by major version.
This makes it easier for users to write more portable
dialplan if desired to not use functionality that may
not be widely available yet.
ASTERISK-29896 #close
Change-Id: Ibbb35c702d8038bdc3fd0a944fbfa69384cc15d5
Currently, each module that uses libcurl duplicates the standard
Asterisk curl user agent.
This adds a global macro for the Asterisk user agent used for
curl requests to eliminate this duplication.
ASTERISK-29861 #close
Change-Id: I9fc37935980384b4daf96ae54fa3c9adb962ed2d
Currently, if VoiceMailMain is called with a mailbox, if that
mailbox doesn't exist, then the application silently falls back
to prompting the user for the mailbox, as if no arguments were
provided.
However, if a specific mailbox is requested and it doesn't exist,
then no warning at all is emitted.
This fixes this behavior to now warn if a specifically
requested mailbox could not be accessed, before falling back to
prompting the user for the correct mailbox.
ASTERISK-29920 #close
Change-Id: Ib4093b88cd661a2cabc5d685777d4e2f0ebd20a4
If Subscription refresh occurred between when the batched notification
was scheduled and the serialized notification was to be sent,
then new schedule notification task would never be added.
There are 2 threads:
thread #1. ast_sip_subscription_notify is called,
if notification_batch_interval then call schedule_notification.
1.1. The schedule_notification checks notify_sched_id > -1
not true, then
send_scheduled_notify = 1
notify_sched_id =
ast_sched_add(sched, sub_tree->notification_batch_interval, sched_cb....
1.2. The sched_cb pushes task serialized_send_notify to serializer
and returns 0 which means no reschedule.
1.3. The serialized_send_notify checks send_scheduled_notify if it's false
the just returns. BUT notify_sched_id is still set, so no more ast_sched_add.
thread #2. pubsub_on_rx_refresh is called
2.1 it pushes serialized_pubsub_on_refresh_timeout to serializer
2.2. The serialized_pubsub_on_refresh_timeout calls pubsub_on_refresh_timeout
which calls send_notify
2.3. The send_notify set send_scheduled_notify = 0;
The serialized_send_notify should always unset notify_sched_id.
ASTERISK-29904 #close
Change-Id: Ifc50c00b213c396509e10326a1ed89d8cf8c7875
Whereas BLFs allow to show a display name for each RLS entry,
the asterisk provides only the extension now.
This is not end user friendly.
This commit adds a new resource_list option, resource_display_name,
to indicate whether display name of resource or the resource name being
provided for RLS entries.
If this option is enabled, the Display Name will be provided.
This option is disabled by default to remain the previous behavior.
If the 'event' set to 'presence' or 'dialog' the non-empty HINT name
will be set as the Display Name.
The 'message-summary' is not supported yet.
ASTERISK-29891 #close
Change-Id: Ic5306bd5a7c73d03f5477fe235e9b0f41c69c681
Adds a simple sanity check for key names when users are
writing data to AstDB. This captures four cases indicating
malformed keynames that generally result in bad data going
into the DB that the user didn't intend: an empty key name,
a key name beginning or ending with a slash, and a key name
containing two slashes in a row. Generally, this is the
result of a variable being used in the key name being empty.
If a malformed key name is detected, a warning is emitted
to indicate the bug in the dialplan.
ASTERISK-29925 #close
Change-Id: Ifc08a9fe532a519b1b80caca1aafed7611d573bf
Adds two pieces of information to the core show settings command
which are useful in the context of getting backtraces.
The first is to display whether or not Asterisk would generate
a core dump if it were to crash.
The second is to show the current running directory of Asterisk.
ASTERISK-29866 #close
Change-Id: Ic42c0a9ecc233381aad274d86c62808d1ebb4d83
The configObject tag contains a default attribute which
allows the default value to be specified, if applicable.
This allows for the default value to show up specially on
the wiki in a way that is clear to users.
There are a couple places in the tree where default values
are included in the description as opposed to as attributes,
which means these can't be parsed specially for the wiki.
These are changed to use the attribute instead of being
included in the text description.
ASTERISK-29898 #close
Change-Id: I9d7ea08f50075f41459ea7b76654906b674ec755
mpg123 doesn't support HTTPS, but the MP3Player application
doesn't document this or warn the user about this. HTTPS
streams have become more common nowadays and users could
reasonably try to play them without being aware they should
use the HTTP stream instead.
This adds documentation to note this limitation. It also
throws a warning if users try to use the HTTPS stream to
tell them to use the HTTP stream instead.
ASTERISK-29900 #close
Change-Id: Ie3b029be5258c5a701f71ed3b1a7a80d1e03b827
Adds an option to the ReceiveMF application to allow specifying a
maximum number of digits.
Originally, this capability was not added to ReceiveMF as it was
with ReceiveSF because typically a ST digit is used to denote that
sending of digits is complete. However, there are certain signaling
protocols which simply transmit a digit (such as Expanded In-Band
Signaling) and for these, it's necessary to be able to read a
certain number of digits, as opposed to until receiving a ST digit.
This capability is added as an option, as opposed to as a parameter,
to remain compatible with existing usage (and not shift the
parameters).
ASTERISK-29877 #close
Change-Id: I4229167c9aa69b87402c3c2a9065bd8dfa973a0b
The disabledevents setting has been added to the general section
in manager.conf, which allows users to specify events that
should be globally disabled and not sent to any AMI listeners.
This allows for processing of these AMI events to end sooner and,
for frequent AMI events such as Newexten which users may not have
any need for, allows them to not be processed. Additionally, it also
cleans up core debug as previously when debug was 3 or higher,
the debug was constantly spammed by "Analyzing AMI event" messages
along with a complete dump of the event contents (often for Newexten).
ASTERISK-29853 #close
Change-Id: Id42b9a3722a1f460d745cad1ebc47c537fd4f205
When tps_shutdown is called as part of the cleanup process there is a
chance that one of the taskprocessors that references the
tps_singletons object is still running. The change is to allow for
tps_shutdown to check tps_singleton's container count and give the
running taskprocessors a chance to finish. If after
AST_TASKPROCESSOR_SHUTDOWN_MAX_WAIT (10) seconds there are still
container references we shutdown anyway as this is most likely a bug
due to a taskprocessor not being unreferenced.
ASTERISK-29365
Change-Id: Ia932fc003d316389b9c4fd15ad6594458c9727f1
There are a lot of Queue AMI actions and Queue applications
which do not load queue and queue members from Realtime.
AMI actions
QueuePause - if queue not in memory - response "Interface not found".
QueueStatus/QueueSummary - if queue not in memory - empty response.
Applications:
PauseQueueMember - if queue not in memory
Attempt to pause interface %s, not found
UnpauseQueueMember - if queue not in memory
Attempt to unpause interface xxxxx, not found
This patch adds a new function load_realtime_queues
which loads queue and queue members for desired queue
or all queues and all members if param 'queuename' is NULL or empty.
Calls the function load_realtime_queues when needed.
Also this patch fixes leak of ast_config in function set_member_value.
Also this patch fixes incorrect LOG_WARNING when pausing/unpausing
already paused/unpaused member.
The function ast_update_realtime returns 0 when no record modified.
So 0 is not an error to warn about.
ASTERISK-29873 #close
ASTERISK-18416 #close
ASTERISK-27597 #close
Change-Id: I554ee0eebde93bd8f49df7f84b74acb21edcb99c
This code was needlessly complex and would fail to properly delimit
the response message if LOW_MEMORY was defined.
Change-Id: Iae50bf09ef4bc34f9dc4b49435daa76f8b2c5b6e
Naveen Albert
Kevin Harwell
Joshua C. Colp
George Joseph
Sean Bright
Marcel Wagner
Boris P. Korzun
Hugh McMaster
Philip Prindeville
Alexei Gradinari
Kfir Itzhak
Mike Bradeen