On a write error to an AMI session a flag was set to
indicate that the write error had occurred, with the
expected result being that the session be terminated.
This was not actually happening and instead writing
would continue to be attempted.
This change adds a check for the write error and causes
the session to actually terminate.
ASTERISK-29948
Change-Id: Icaf5d413d4c0d5dc78292a17287fecc8720a31a5
Add framework to connect to, and read and write protocol based
messages from and to an external application using an Asterisk
External Application Protocol (AEAP). This has been divided into
several abstractions:
1. transport - base communication layer (currently websocket only)
2. message - AEAP description and data (currently JSON only)
3. transaction - links/binds requests and responses
4. aeap - transport, message, and transaction handler/manager
This patch also adds an AEAP implementation for speech to text.
Existing speech API callbacks for speech to text have been completed
making it possible for Asterisk to connect to a configured external
translator service and provide audio for STT. Results can also be
received from the external translator, and made available as speech
results in Asterisk.
Unit tests have also been created that test the AEAP framework, and
also the speech to text implementation.
ASTERISK-29726 #close
Change-Id: Iaa4b259f84aa63501e5fd2a6fb107f900b4d4ed2
Patch provided inline by Yury Kirsanov on the linked issue and
approved by Josh Colp.
ASTERISK-29253 #close
Change-Id: I5b9ccc67ebf06e875ed061d9e7fc21f47b0a4e1f
Added support for a basic AEAP configuration read from aeap.conf.
Also added 2 CLI commands for showing individual configurations as
well as all of them: aeap show server <id> and aeap show servers.
Only one configuration option is required at the moment, and that one is
server_url. It must be a websocket URL. The other option, codecs, is
optional and will be used over the codecs specified on the endpoint if
provided.
https://wiki.asterisk.org/wiki/pages/viewpage.action?pageId=45482453
Change-Id: I567ac5148c92b98d29d2ad83421b416b75ffdaa3
(cherry picked from commit 45a1977de4)
The async_operations setting on a transport configures how
many simultaneous incoming packets the transport can handle
when multiple threads are polling and waiting on the transport.
As we only use a single thread this was needlessly creating
incoming packets when set to a non-default value, wasting memory.
ASTERISK-30006
Change-Id: I1915973ef352862dc2852a6ba4cfce2ed536e68f
There was an issue with the conditional where STIR/SHAKEN would be
enabled even when not configured. It has been changed to ensure that if
a profile does not exist and stir_shaken is not set in pjsip.conf, then
the conditional will return from the function without performing
STIR/SHAKEN operations.
ASTERISK-30024
Change-Id: I41286a3d35b033ccbfbe4129427a62cb793a86e6
When executing dial, the topology of the incoming channel is cloned and
used for the outgoing channel. This creates issues when an incoming
stream is sendonly or recvonly as the stream state of the outgoing
channel will be the same as the stream state of the incoming channel.
Now the stream state is flipped for the outgoing stream in
dial_exec_full if the incoming stream topology is recvonly or sendonly.
ASTERISK-29655
Reported by: Michael Auracher
ASTERISK-29638
Reported by: Michael Auracher
Change-Id: I294dc834ac9a5f048b101b691669959e9df630e1
Chrome has added more attributes, causing the limit to be
exceeded. This raises it up some more.
ASTERISK-30015
Change-Id: I964957c005c4e6f7871b15ea1ccd9b4659c7ef32
Adds a new configuration option, stir_shaken_profile, in pjsip.conf that
can be specified on a per endpoint basis. This option will reference a
stir_shaken_profile that can be configured in stir_shaken.conf. The type
of this option must be 'profile'. The stir_shaken option can be
specified on this object with the same values as before (attest, verify,
on), but it cannot be off since having the profile itself implies wanting
STIR/SHAKEN support. You can also specify an ACL from acl.conf (along
with permit and deny lines in the object itself) that will be used to
limit what interfaces Asterisk will attempt to retrieve information from
when reading the Identity header.
ASTERISK-29476
Change-Id: I87fa61f78a9ea0cd42530691a30da3c781842406
Some databases depending on their configuration using backslashes
for escaping. When combined with the use of ' this can result in
a broken func_odbc query.
This change adds a SQL_ESC_BACKSLASHES dialplan function which can
be used to escape the backslashes.
This is done as a dialplan function instead of being always done
as some databases do not require this, and always doing it would
result in incorrect data being put into the database.
ASTERISK-29838
Change-Id: I152bf34899b96ddb09cca3e767254d8d78f0c83d
Put checks in place to limit how much we will actually download, as well
as a check for the data we receive at the start to ensure it begins with
what we would expect a certificate to begin with.
ASTERISK-29872
Change-Id: Ifd3c6b8bd52b8b6192a04166ccce4fc8a8000b46
The ReceiveMF and ReceiveSF applications currently always
return 0, even if a channel has hung up. The call will still
end but generally applications are expected to return -1 if
the channel has hung up.
We now return -1 if a hangup occured to bring this behavior
in line with this norm. This has no functional impact, but
merely increases conformity with how these modules interact
with the PBX core.
ASTERISK-29951 #close
Change-Id: I234d755050ab8ed58f197c6925b968ba26b14033
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
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
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
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
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
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
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
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
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
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
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
(cherry picked from commit 37ece75677)
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
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
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
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
Adds documentation for all of the possible return values
for the DIALSTATUS variable in the Dial application.
ASTERISK-25716
Change-Id: Id22593f1f1f7ea86e5734cee49516ec50848e8c0
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
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
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
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
