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
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
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
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