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
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
When realtime text packets are to be sent, the text is accumulated in a
buffer and sent regularly by a timer. It can happen that commands such as
a backspace, CR, or LF get merged with regular text. This breaks some
UAs.
The proposed change:
* We test if the current packet contains a command. If so we send the
buffer immediately.
* We test if the buffer contained a command. If so we send the buffer
immediately.
* We accumulate the text (or the command) in the buffer.
ASTERISK-27970
Change-Id: Ifbe993311410fa855cb8aa4a12084db75f413462
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
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
Keep track if ICE candidates were in the SDP offer & only put them
in the corresponding SDP answer if the offer condaind ICE candidates
ASTERISK-27957 #close
Change-Id: Idf2597ee48e9a287e07aa4030bfa705430a13a92
Previously, Asterisk did not tell its bundled PJProject about this configure
parameter. Therefore, PJProject used the platform provided OpenSSL always.
ASTERISK-27880
Change-Id: Iea545aec854dd0e2c061c69bb118a76ce56c5dc6
Fixes issue where error msg
"Use of before/init after destruction"
was being printed on disabled messages
in dev mode. With this
fix if message is disabled
a warning will print.
ASTERISK-25548
Change-Id: Ie0d866d1cbc60c16dbef08bc65e99505c3c1adfa
A problem I've seen countless times is a global or system section
for PJSIP not getting applied. This is inevitably the result of
the "type=" line missing. This change alleviates that problem.
The ability to specify an explicit section name has been
added to res_sorcery_config. If the configured section
name matches this and there are no unknown things configured
the section is taken as being for the given type.
Both the PJSIP "global" and "system" types now support this
so you can just name your section "global" or "system" and it
will be matched and used, even without a "type=" line.
ASTERISK-27972
Change-Id: Ie22723663c1ddd24f869af8c9b4c1b59e2476893
SRTP SDES key lifetime support was added in ASTERISK_17899.
In that addition, the minimum key lifetime to be accepted was
set at the 10 hours @ 20ms/packet = 1800000 packets.
The firmware in the obi1xx ATA uses a hardcoded lifetime of
2^20 packets.
Lower the limit to 2^20 to support a wider field of clients.
ASTERISK-27967 #close
Change-Id: I81a0703c595a0c9101dfdf02300149a3cc39bf94
Asterisk patched the pjproject source to avoid crashing when pjproject
sip_msg headers are encountered with NULL vptr's, but the patch also
output error messages for some valid headers which simply did not need
to be added to the message itself, such as hidden route headers.
pjproject has since applied a similar patch to their baseline to avoid
crashes, but their version also avoids the spurious error logging.
Lets use their patch instead.
ASTERISK-27961 #close
Change-Id: I2ddbd82c8da10e0dcc9807a48089d1f3c2d6e389
Asterisk modules that use PJPROJECT services have their compiler
optimization and possibly their symbolic debug options overridden by the
PJPROJECT configure script selected settings.
* We need to filter-out any -O and -g options in PJ_CFLAGS before echoing
out the result so the PJPROJECT_INCLUDE variable does not override the
Asterisk module settings when using bundled PJPROJECT.
NOTE: This patch only has an effect when using bundled PJPROJECT.
ASTERISK-27563
Change-Id: If124169735ecf572ad1535cd43bff94cb44d5b30
When `--without-libcurl` is used PBX_CURL is never set. Set default
value 0 so the proper value is passed to menuselect.
Change-Id: I03e2842a00899cbca2dbde52bb1f6636d54bae1e
