This target requires specifying CONFIG_SRC=path_to_configs. This can be
used to install custom configs for the Asterisk build while still
performing directory replacements on asterisk.conf.
Modify internal INSTALL_CONFIGS so first argument requires full path to
the config sources relative to Asterisk source root.
Change-Id: Idcd841df3c8d5bfe23d566bb9e2e448e9df4f8ab
When converting from a json object to an ast variables list the conversion
algorithm was doing a complete traversal of the entire variables list for
every item appended from the json structure.
This patch makes it so the list is no longer traversed for each new ast
variable being appended.
Change-Id: I8bf496a1fc449485150d6db36bfc0354934a3977
When publishing a device state the change can be marked as being
cachable or not. If it is not cached the change is just published
to all interested and not stored away for later query. This was not
fully taken into account when publishing in stasis. The act of
publishing would create a topic for the device even if it may be
ephemeral.
This change makes it so messages which are not cached won't create
a topic for the device. If a topic does already exist it will be
published to but otherwise the change will only be published to
the device state all topic.
ASTERISK-27591
Change-Id: I18da0e8cbb18e79602e731020c46ba4101e59f0a
Update the bundled jansson Makefile to do nothing during Asterisk
install, use a target that is not phony to initiate the jansson make and
install.
Change-Id: I7643cc3d39af9feba8fc0da676b646efc5f8b3bb
If a SIP MESSAGE is triggered for an endpoint that is currently not registered
- and therefore has no valid contact associated - an error message was logged.
Since this is a valid request in a valid use cases this is now changed to a
warning, as discussed with Matt Fredrickson on the asterisk-dev mailing list.
Change-Id: I55eb62d2712818a58c7532119dec288bd98cf0c0
Remove AST_DEVMODE from the AST_BUILDOPTS list and the AST_BUILDOPTS_SUM
calculation as it no longer affects API/ABI compatibility.
Change-Id: Id5bd6dfade173a53b3a49f715586b86e3fb24acb
The --test-command argument has now been split, unit tests now use
`--unittest-command` and the testsuite uses --testsuite-command.
This will make it easier to create a script which run everything by
forwarding the same arguments to all CI scripts.
Change-Id: Ia54aa4848eaffbdf13175fcda40fc0b23080ad71
The "xmldoc dump" cli command was simply concatenating xml documents
into the output file. The resulting file had multiple "xml"
processing instructions and multiple root elements which is illegal.
Normally this isn't an issue because Asterisk has only 1 main xml
documentation file but codec_opus has its own file so if it's
downloaded and you do "xmldoc dump", the result is invalid.
* Added 2 new functions to xml.c:
ast_xml_copy_node_list creates a copy of a list of children.
ast_xml_add_child_list adds a list to an existing list.
* Modified handle_dump_docs to create a new output document and
add to it the children from each input file. It then dumps the
new document to the output file.
Change-Id: I3f182d38c75776aee76413dadd2d489d54a85c07
Use .gitreview defaultbranch setting to determine the mainline branch.
This allows the script to be used against other directories which might
not be on the same defaultbranch. This can be used by CI scripts to
report the testsuite version being used:
./build_tools/make_version ${TESTSUITE_DIR}
Change-Id: Ifdad4a9d8a26138c41bc6b630ecc3e34ea1c2758
In the past there was an assertion in the ast_sched_del function
and in order to ensure it was useful the calling function name,
line number, and filename had to be passed in. This cause the ABI
to be different between dev mode and non-dev mode.
This assertion is no longer present so the special logic can be
removed to make it the same between them both.
Change-Id: Icbc69c801e357d7004efc5cf2ab936d9b83b6ab8
A change recently went in which disabled the built-in PJSIP
keepalive. This defaulted to 90 seconds and kept TCP/TLS
connections alive. Disabling this functionality has resulted
in a behavior change of not doing keepalives by default resulting
in TCP/TLS connections dropping for some people.
This change makes our default keepalive interval 90 seconds
to match the previous behavior and preserve it.
ASTERISK-27978
Change-Id: Ibd9a45f3cbe5d9bb6d2161268696645ff781b1d6
* Use the replacement function ast_sip_push_task_wait_servant() instead of
the deprecated ast_sip_push_task_synchronous().
Change-Id: I145b550ba7054640c7faa3b644e63137f505c612
