Adds the m option to the Queue application, which allows a
music on hold class to be specified at runtime which will
override the class configured in queues.conf.

This option functions like the m option to Dial.

ASTERISK-29876 #close

Change-Id: Ie25a48569cf8755c305c9438b1ed292c3adcf8d7

Currently, if a user tries to access a non-dynamic
MeetMe conference and the conference is not found,
the call simply silent hangs up. There is no indication
to the user that anything went wrong at all.

This changes the relevant debug message to a warning
so that the user is notified of this invalidity.

ASTERISK-29954 #close

Change-Id: Iebcfae3755d00f2150d676ee211c57bc59530048

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

ASTERISK-29980 #close

Change-Id: I7b347665822ea2774dd322276c09be67914d2065

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

ASTERISK-29353

Change-Id: I4ea43eda1691565563a4c03ef37166952d211b2b

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

Refactors temp file utility function into file.c.

ASTERISK-29809 #close

Change-Id: Ife478708c8f2b127239cb73c1755ef18c0bf431b