Corey Farrell [Tue, 7 Nov 2017 23:07:35 +0000 (18:07 -0500)]
CLI: Create ast_cli_completion_vector.
This is a rewrite of ast_cli_completion_matches using a vector to build
the list. The original function calls the vector version, NULL
terminates the vector and extracts the elements array.
One change in behavior the results are now sorted and deduplicated. This
will solve bugs where some duplicate checking was done before the list
was sorted.
Sean Bright [Thu, 16 Nov 2017 22:18:16 +0000 (17:18 -0500)]
res_pjsip_transport_websocket: Give transport a meaningful description
We were not \0 terminating this string, so any attempt to print it would
in the best case show an empty string and in the worst case potentially
crash.
Richard Mudgett [Wed, 15 Nov 2017 00:00:55 +0000 (18:00 -0600)]
audiohook.c: Fix freeing a frame and still using it.
Memory corruption happened to the media frame caches when an audio hook
freed a frame when it shouldn't. I think the freed frame was because a
jitter buffer interpolated a missing frame and the audio hook
unconditionally freed it.
* Made audiohook.c:audio_audiohook_write_list() not free an interpolated
frame if it is the same frame as what was passed into the routine.
* Made plc.c:normalise_history() use memmove() instead of memcpy() on a
memory block that could overlap. Found by valgrind investigating this
issue.
George Joseph [Wed, 15 Nov 2017 18:10:16 +0000 (11:10 -0700)]
app_record: Don't set RECORD_STATUS chan var until file is closed
We've been calling pbx_builtin_setvar_helper to set the
RECORD_STATUS variable before actually closing the recorded file.
If a client is watching VarSet events and tries to do something with
the file when a RECORD_STATUS event is seen, they might attempt to
do so while the file it's still open.
We now delay calling pbx_builtin_setvar_helper until after we close
the file.
George Joseph [Tue, 7 Nov 2017 14:25:59 +0000 (07:25 -0700)]
ast_coredumper: Add ability to use directory other than /tmp
The OUTPUTDIR environment variable can now be set either in the
environment itself or in ast_debug_tools.conf. If set, it's used
for all work products instead of /tmp.
Also added the --tarball-config option that includes the contents
of /etc/asterisk when either --tarball-coredumps or --tarball-results
are used.
Joshua Colp [Mon, 13 Nov 2017 13:14:23 +0000 (13:14 +0000)]
pjsip / hep: Provide correct local address for Websockets.
Previously for PJSIP the local address of WebSocket connections
was set to the remote address. For logging purposes this is
not particularly useful.
The WebSocket API has been extended to allow the local
address to be queried and this is used in PJSIP to set the
local address to the correct value.
The PJSIP HEP support has also been tweaked so that reliable
transports always use the local address on the transport
and do not try to (wrongly) guess. As they are connection
based it is impossible for the source to be anything else.
Corey Farrell [Mon, 13 Nov 2017 22:20:27 +0000 (17:20 -0500)]
core: Fix configuration of remote console socket path.
The remote console socket path is the combination of asterisk.conf
settings astrundir from [directories] and astctl from [files].
Unconditionally combine the two strings after processing all values
to ensure we end up with the correct socket path.
George Joseph [Fri, 10 Nov 2017 16:37:16 +0000 (09:37 -0700)]
bundled_pjproject: sip_parser: Fix return code in pjsip_find_msg
The default return code for pjsip_find_msg was PJ_SUCCESS so if
a Content-Length header wasn't found at all, pjsip_find_msg was
returning PJ_SUCCESS instead of PJSIP_EMISSINGHDR.
Also added the volatile keyword to a few variables that are used
both inside and outside the PJ_TRY/PJ_CATCH block.
Sean Bright [Thu, 9 Nov 2017 14:21:38 +0000 (09:21 -0500)]
sorcery: Add ast_sorcery_retrieve_by_prefix()
Some consumers of the sorcery API use ast_sorcery_retrieve_by_regex
only so that they can anchor the potential match as a prefix and not
because they truly need regular expressions.
Rather than using regular expressions for simple prefix lookups, add
a new operation - ast_sorcery_retrieve_by_prefix - that does them.
Patches against 13 and 15 have a compatibility layer needed to
maintain ABI that is not needed in master.
Richard Mudgett [Sat, 11 Nov 2017 19:01:47 +0000 (13:01 -0600)]
core: Add cache_media_frames debugging option.
The media frame cache gets in the way of finding use after free errors of
media frames. Tools like valgrind and MALLOC_DEBUG don't know when a
frame is released because it gets put into the cache instead of being
freed.
* Added the "cache_media_frames" option to asterisk.conf. Disabling the
option helps track down media frame mismanagement when using valgrind or
MALLOC_DEBUG. The cache gets in the way of determining if the frame is
used after free and who freed it. NOTE: This option has no effect when
Asterisk is compiled with the LOW_MEMORY compile time option enabled
because the cache code does not exist.
To disable the media frame cache simply disable the cache_media_frames
option in asterisk.conf and restart Asterisk.
Joshua Colp [Fri, 10 Nov 2017 13:06:38 +0000 (13:06 +0000)]
pjsip: Add patch to allow all transports to be destroyed.
If a transport is created with the same transport type, source
IP address, and source port as one that already exists the old
transport is moved into a linked list called "tp_list".
If this old transport is later shutdown it will not be destroyed
as the process checks whether the transport is valid or not. This
check does not look at the "tp_list" when making the determination
causing the transport to not be destroyed.
This change updates the logic to query not just the main storage
method for transports but also the "tp_list".
Corey Farrell [Thu, 9 Nov 2017 19:24:46 +0000 (14:24 -0500)]
Build System: Disable parallel make in the root Makefile.
This ensures that the root Makefile runs only a single target at a time.
SUBMAKE will still honor requested parallelism, so 'make -j8' will build
one directory at a time but allow 8 jobs at once when building a sub
directory.
This will fix some display glitches related to rebuild of XML
documentation. It will also prevent some edge case errors where
bundled pjproject needs to be rebuild before other parts of Asterisk.
Kevin Harwell [Thu, 19 Oct 2017 18:35:16 +0000 (13:35 -0500)]
AST-2017-011 - res_pjsip_session: session leak when a call is rejected
A previous commit made it so when an invite session transitioned into a
disconnected state destruction of the Asterisk pjsip session object was
postponed until either a transport error occurred or the event timer
expired. However, if a call was rejected (for instance a 488) before the
session was fully established the event timer may not have been initiated,
or it was canceled without triggering either of the session finalizing states
mentioned above.
Really the only time destruction of the session should be delayed is when a
BYE is being transacted. This is because it's possible in some cases for the
session to be disconnected, but the BYE is still transacting.
This patch makes it so the session object always gets released (no more
memory leak) when the pjsip session is in a disconnected state. Except when
the method is a BYE. Then it waits until a transport error occurs or an event
timeout.
cdr_object_update_party_b_userfield_cb() could overrun the fixed buffer if
the supplied string is too long. The long string could be supplied by
external means using the CDR(userfield) function.
This may seem reminiscent to AST-2017-001 (ASTERISK_26897) and it is. The
earlier patch fixed the buffer overrun for Party A's userfield while this
patch fixes the same thing for Party B's userfield.
George Joseph [Thu, 19 Oct 2017 18:53:32 +0000 (12:53 -0600)]
AST-2017-009: pjproject: Add validation of numeric header values
Parsing the numeric header fields like cseq, ttl, port, etc. all
had the potential to overflow, either causing unintended values to
be captured or, if the values were subsequently converted back to
strings, a buffer overrun. To address this, new "strto" functions
have been created that do range checking and those functions are
used wherever possible in the parser.
* Created pjlib/include/limits.h and pjlib/include/compat/limits.h
to either include the system limits.h or define common numeric
limits if there is no system limits.h.
* Created strto*_validate functions in sip_parser that take bounds
and on failure call the on_str_parse_error function which prints
an error message and calls PJ_THROW.
* Updated sip_parser to validate the numeric fields.
* Fixed an issue in sip_transport that prevented error messages
from being properly displayed.
* Added "volatile" to some variables referenced in PJ_CATCH blocks
as the optimizer was sometimes optimizing them away.
* Fixed length calculation in sip_transaction/create_tsx_key_2543
to account for signed ints being 11 characters, not 9.
ASTERISK-27319
Reported by: Youngsung Kim at LINE Corporation
Richard Mudgett [Mon, 6 Nov 2017 22:37:49 +0000 (16:37 -0600)]
res_pjsip_registrar.c: Fix named AOR and pjproject group deadlock.
One of the patches for ASTERISK_27147 introduced a deadlock regression.
When the connection oriented transport shut down, the code attempted to
remove the associated contact. However, that same transport had just
requested a registration that we hadn't responded to yet. Depending
upon timing we could deadlock.
* Made send the REGISTER response after we completed processing the
request contacts and released the named AOR lock to avoid the deadlock.
Corey Farrell [Mon, 6 Nov 2017 18:51:32 +0000 (13:51 -0500)]
CLI: Remove unused internal command.
The internal CLI command "_command complete" was last used by Asterisk
0.2.0. Since then we've been using "_command nummatches" and "_command
matchesarray".
Joshua Colp [Mon, 6 Nov 2017 16:29:33 +0000 (16:29 +0000)]
res_pjsip_pubsub: Ensure remote URI contains URI only.
This change makes it so that any user of the pubsub
API that requests the remote URI receives only the URI.
Previously the entire string was returned, which could
contain a display name.
When (v)asprintf() fails, the state of the allocated buffer is undefined.
The library had better not leave an allocated buffer as a result or no one
will know to free it. The most likely way it can return failure is for an
allocation failure. If the printf conversion fails then you actually have
a threading problem which is much worse because another thread modified
the parameter values.
* Made __ast_asprintf()/__ast_vasprintf() set the returned buffer to NULL
on failure. That is much more useful than either an uninitialized pointer
or a pointer that has already been freed. Many uses won't have to check
for failure to ensure that the buffer won't be double freed or prevent an
attempt to free an uninitialized pointer.
* stasis.c: Fixed memory leak in multi_object_blob_to_ami() allocated by
ast_asprintf().
* ari/resource_bridges.c:ari_bridges_play_helper(): Remove assignment to
the wrong thing which is now not needed even if assigning to the right
thing.
Sean Bright [Mon, 6 Nov 2017 14:05:56 +0000 (09:05 -0500)]
res_pjsip: Ignore empty TLS configuration
When using realtime, fields that are not explicitly set by an
administrator are still presented to sorcery as empty strings. Handle
this case explicitly.
In this particular case, if any of these fields are required for TLS
support, their existence should be validated in the 'apply' handler once
we have a complete transport definition.
Alexander Traud [Mon, 6 Nov 2017 09:18:24 +0000 (10:18 +0100)]
tcptls: Print notice when TLS is enabled but not configured.
Asterisk can be compiled without a SSL/TLS library, without the Development
Headers of OpenSSL. However, if TLS (SIP) or Secure-WebSockets (WebRTC) was
enabled in a configuration file, Asterisk did not notice the user. Asterisk
failed silently, only the corresponding TCP ports were not open.
Corey Farrell [Mon, 6 Nov 2017 09:21:12 +0000 (04:21 -0500)]
configure: Add autoconf check for libopusfile.
This check is being added to make it easier for end-users of third party
open source Opus modules. This was removed by ASTERISK-26426 but only
the module needed to be removed.
projects / thirdparty / asterisk.git / log
