Sean Bright [Tue, 21 Aug 2018 18:50:33 +0000 (14:50 -0400)]
app_queue: Silence GCC 8 compiler warning
I'm only seeing an error in 14+, so I assume it is due to different
compiler options:
app_queue.c: In function ‘handle_queue_add_member’:
app_queue.c:10234:19: error: ‘%d’ directive writing between 1 and 11
bytes into a region of size 3 [-Werror=format-overflow=]
sprintf(num, "%d", state);
^~
app_queue.c:10234:18: note: directive argument in the range
[-2147483648, 99]
sprintf(num, "%d", state);
^~~~
Richard Mudgett [Sat, 11 Aug 2018 00:28:45 +0000 (19:28 -0500)]
res_pjsip_t38.c: Fix crash if already saw a final T.38 reINVITE response.
We were still getting crashes after the first fix. Somehow we receive a
non-2xx final response before we get a 200 final response. With the
failure response we had already cleaned up and destroyed some data
structures. When the unexpected 200 response comes in we crash.
* Add protection code to prevent processing another final T.38 reINVITE
response.
Richard Mudgett [Thu, 16 Aug 2018 02:31:45 +0000 (21:31 -0500)]
pbx_dundi.c: Misc memory management fixes when destroying peers
* In destroy_peer(), fixed memory leaks of lookup history strings and
qualify transactions when destroying peers.
* In destroy_peer(), fixed leaving the registerexpire scheduled callback
active when a peer is destroyed on a reload. The reload marks and sweeps
peers so any peers not explicitly configured get destroyed. Peers created
dynamically from the '*' peer will not exist until they re-register after
the reload. These destroyed peers caused memory corruption when the
registerexpire timer expired.
* Made build_peer() not schedule any callbacks on the '*' peer
(empty_eid). It is a special peer that is cloned to dynamically created
peers so it doesn't actually get involved in any message transactions.
* Made do_register_expire() remove the dundi/dpeers AstDB entry when a
peer registration expires.
* Fix deep_copy_peer() to not copy some things that cannot be copied to
the cloned peer structure. Timers, message transactions, and lookup
history are specific to a peer instance.
* Made set_config() lock around processing the mappings configuration.
* Reordered unload_module() to handle load_module() declining the load due
to error.
Richard Mudgett [Wed, 15 Aug 2018 23:14:52 +0000 (18:14 -0500)]
pbx_dundi: Fix debug frame decode string.
* Fixed a typo in the name of the REGREQ frame decode string array.
* Fixed off by one range check indexing into the frame decode string
array.
* Removed some unneeded casts associated with the decode string array.
Richard Mudgett [Wed, 15 Aug 2018 19:44:48 +0000 (14:44 -0500)]
res_rtp_asterisk.c: Fix unused variable warnings
Compiling without SRTP support installed resulted in some unused variable
warnings. These warnings also showed that the srtp variable was obtained
and passed around some functions but not really used even when a system
has SRTP installed.
George Joseph [Thu, 16 Aug 2018 17:08:21 +0000 (11:08 -0600)]
CI: Add https credentials to gerrit checkouts
If the review to be tested is in a project with restricted access,
we need to use the jenkins user's gerrit https credentials when we
do the checkout or the checkout will fail.
Joshua Colp [Tue, 14 Aug 2018 12:29:18 +0000 (09:29 -0300)]
res_pjsip_caller_id: Add "party" parameter to RPID header.
This change adds the "party" parameter to the Remote-Party-ID header
which indicates which party the header information is applicable
to. In Asterisk this is determined on whether we are the calling
or called party. This is added to improve interoperability with some
implementations.
Ben Ford [Tue, 7 Aug 2018 15:57:29 +0000 (10:57 -0500)]
res_pjsip/rtp: No joint capabilities between streams.
When a conference contained a mixture of audio/video and audio-only
users, a NOTICE message would pop up stating there are no joint
capabilities between streams. This happens because streams can never be
removed, but they can be in a REMOVED state. If we have the scenario
where user A joins with audio/video, user B joins with audio-only, and
user C joins with audio/video, then user A leaves, the message would
be triggered. That removed stream is still in the SDP, but Asterisk
would pass it through, causing it to be seen as a ulaw stream. A check
has been added for removed streams, setting their status to REMOVED when
handling negotiated SDPs.
Also addressed an issue where user A joins, then user B joins but does
not receive video until much later. Full frames were not being sent,
causing some PLI from the browser. Because the video was flowing in one
direction, the browser sets the SSRC to 1, but Asterisk was dropping the
PLI because of that. Added a check to see if the SSRC is 1 or not, which
sends full frames and allows video to flow between user A and user B.
This should only happen when dealing with PSFB or FUR, and in the case
of PSFB, only for PLI.
Ivan Poddubny [Sun, 12 Aug 2018 16:04:42 +0000 (18:04 +0200)]
app_queue: set QUEUESTATUS to LEAVEEMPTY instead of CONTINUE
When a call leaves a queue on leaveempty condition, QUEUESTATUS
must be set to LEAVEEMPTY, no matter whether Queue was executed with or
without the "c" (continue) option.
The regression was introduced in the fix for ASTERISK_25665.
The following fix (ASTERISK_27065) was incomplete, as QUEUESTATUS was
overwritten in case when "c" is set, regardless of what was the cause
for leaving the queue.
Corey Farrell [Thu, 9 Aug 2018 20:25:41 +0000 (16:25 -0400)]
Sample configs: Fix pjsip.conf syntax error.
It is valid for a config file to be empty or contain only comments, but
not valid for a config value to be set when no uncommented context
exists. This caused an error to be loged numerous times during start
when loading the default pjsip.conf.
Enable coverage with `./tests/CI/buildAsterisk.sh --coverage`. This
will cause Asterisk to be compiled with coverage support. It also
initializes 'before' coverage data for all sources. Accept
--tested-only to disable modules which are not run by any test.
Enabling coverage also sets tested-only true by default. To build
everything with coverage enabled use `--coverage --tested-only=0`.
./tests/CI/processCoverage.sh is used to process the coverage and
generate HTML reports.
Fix utils/check_expr2 which failed to compiled with coverage enabled.
Add status output 5 times per stage of astobj2_test_perf to ensure
remote CLI does not timeout when compiled with coverage. Remote CLI
disconnects if no output is received for 60 seconds. When coverage is
enabled it takes about 70 seconds for my laptop to run the stages of
this test, so with the change a message is printed every 14 seconds.
Joshua Colp [Mon, 6 Aug 2018 11:36:22 +0000 (08:36 -0300)]
stasis: Reduce calculation of stasis message type hash.
When the stasis cache is used a hash is calculated for
retrieving or inserting messages. This change calculates
a hash when the message type is initialized that is then
used each time needed. This ensures that the hash is
calculated only once for the message type.
Alexander Traud [Mon, 30 Jul 2018 12:49:08 +0000 (14:49 +0200)]
pjproject_bundled: Fix for Solaris builds. Do not undef s_addr.
The authors of PJProject undef s_addr because of some issue in Microsoft
Windows. However in Oracle Solaris, s_addr is not a structure member, but
defined to map to the real structure member.
Salah Ahmed [Thu, 2 Aug 2018 19:37:16 +0000 (14:37 -0500)]
dialplan_functions: wrong srtp use status report of a dialplan function
If asterisk offer an endpoint with SRTP and that endpoint respond
with non srtp, in that case channel(rtp,secure,audio) reply wrong
status.
Why delete flag AST_SRTP_CRYPTO_OFFER_OK while check identical remote_key:
Currently this flag has being set redundantly. In either case identical
or different remote_key this flag has being set. So if we
don't set it while we receive identical remote_key or non SRTP SDP
response then we can take decision of srtp use by using that flag.
Alexander Traud [Mon, 30 Jul 2018 11:05:34 +0000 (13:05 +0200)]
pjproject_bundled: Find shared libraries in root --with-ssl=PATH.
The script configure from Teluu expects shared libraries (.so) in a subfolder
called 'lib', when --with-xyz=PATH is specified. However for OpenSSL, the
default location is the root of the source folder = PATH. Furthermore, Asterisk
supports both, 'lib' and root. For consistency and because Asterisk is using
(only) OpenSSL in PJProject, it is enhanced to support both locations, just
like Asterisk.
Joshua Colp [Wed, 1 Aug 2018 14:45:04 +0000 (14:45 +0000)]
res_pjsip_registrar: Improve performance on inbound handling.
This change removes a sorcery lookup for retrieving all
contacts at the end of the registration process by keeping
track of the contacts that are added/updated/deleted.
This ensures at the end of the process the container of
contacts we have is the current state.
Pool usage has also been reduced by allocating one for
usage throughout the handling of a REGISTER and resetting
it to a clean state. This ensures that in most cases
we allocate once and just reuse it.
thirdparty/pjproject: fix deadlock in response retransmissions
The tdata containing the response can be shared by both the dialog
object and the tsx object. In order to prevent the race condition
between the tsx retransmission and the dialog sending a response,
clone the tdata before modifying it for the dialog send response.
Alexander Traud [Sat, 28 Jul 2018 16:49:17 +0000 (18:49 +0200)]
BuildSystem: Enable ncurses for menuselect in Solaris 11.
The check for the library ncurses should use not use the header <curses.h> but
<ncurses.h>, because on some platforms <curses.h> is not a drop-in replacement
for <ncurses.h>: For example in Solaris, the symbol initscr is a typedef in
<curses.h> to a symbol which does not exist in the library ncurses (initscr32).
Simply use <ncurses.h> when you link to ncurses.
Furthermore in Solaris, the header <ncurses.h> is in a subdirectory
/usr/include/ncurses and not available via pkg-config.
Alexander Traud [Sat, 28 Jul 2018 13:08:40 +0000 (15:08 +0200)]
BuildSystem: Enable Jansson in Solaris 11.
In Solaris, the header <jansson.h> is in /usr/include/jansson. To find
Jansson even in such a subdirectory, the tool pkg-config is queried via
AST_PKG_CONFIG_CHECK. For those platforms, which do not list Jansson via
pkg-config, the previous check remains and is executed thereafter.
Because the check for the NetBSD Editline library uses the tool pkg-config
conditionally PKG_PROG_PKG_CONFIG must be used. Because that check happens
earlier than Jansson, it must be placed in front of that.
The script configure does some pre-checks for the script configure of the
Asterisk internal NetBSD Editline library. The check for the library ncurses
should use not use the header <curses.h> but <ncurses.h>, because on some
platforms <curses.h> is not a drop-in replacement for <ncurses.h>: For example
in Solaris, the symbol initscr is a typedef in <curses.h> to a symbol which
does not exist in the library ncurses (initscr32). Simply use <ncurses.h> when
you link to ncurses.
Richard Mudgett [Tue, 24 Jul 2018 18:44:41 +0000 (13:44 -0500)]
res_pjsip_endpoint_identifier_ip.c: Added regex support to match_header
This patch adds regular expression support to make the identify section's
match_header option more useful when attempting to match complex headers
like the 'To' or 'From' headers. The 'From' header has variable
components such as the tag parameter that you cannot predict. To specify
a regular expression put slashes around the regular expression in place of
the header value.
* Added regex support to match_header so you could match a 'To' header
among other complex headers.
Fixed reported crashes when trying to match special headers like 'Contact'.
The identify section's match_header method used code that assumed you were
matching a generic header. Any other type of header could cause a crash
if the header structure variant did not match the generic header enough.
* Made use code that will work for any header type instead of code
specific to generic headers.
Other fixes while in the area:
* Made check all headers of the requested name.
* Added some more sanity checks to the configured identify matching
options when applying the configuration.
res_pjsip_pubsub: Treat "prune_on_boot" as a yes / no.
The alembic for the PJSIP subscription persistence table has the
"prune_on_boot" field as a boolean. While in Asterisk we are
tolerant of many different definitions of true and false in the
database we only accept "yes" and "no". This change makes the
field treated as a yes/no instead of an integer, thus storing
"yes" and "no" instead of "1" and "0".
res_rtp_asterisk: Avoid merging command and regular T.140 text packets
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.
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.
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.
devicestate: Don't create topic when change isn't cached.
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.
res_pjsip: Change log message from error to warning for valid use cases
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.
projects / thirdparty / asterisk.git / log
