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In all non-pbx modules, AST_MODULE_LOAD_FAILURE has been changed
to AST_MODULE_LOAD_DECLINE. This prevents asterisk from exiting
if a module can't be loaded. If the user wishes to retain the
FAILURE behavior for a specific module, they can use the "require"
or "preload-require" keyword in modules.conf.
A new API was added to logger: ast_is_logger_initialized(). This
allows asterisk.c/check_init() to print to the error log once the
logger subsystem is ready instead of just to stdout. If something
does fail before the logger is initialized, we now print to stderr
instead of stdout.
Change-Id: I5f4b50623d9b5a6cb7c5624a8c5c1274c13b2b25
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* changes:
stun.c: Fix ast_stun_request() erratic timeout.
sorcery.c: Speed up ast_sorcery_retrieve_by_id()
res_pjsip: Fix pointer use after unref.
res_pjsip_sdp_rtp.c: Don't use deprecated transport struct member.
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If ast_stun_request() receives packets other than a STUN response then we
could conceivably never exit if we continue to receive packets with less
than three seconds between them.
* Fix poll timeout to keep track of the time when we sent the STUN
request. We will now send a STUN request every three seconds regardless
of how many other packets we receive while waiting for a response until we
have completed three STUN request transmission cycles.
Change-Id: Ib606cb08585e06eb50877f67b8d3bd385a85c266
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Return early if ast_sorcery_retrieve_by_id() is not passed an id to find.
Also eliminated the RAII_VAR() usage in the function.
Change-Id: I871dbe162a301b5ced8b4393cec27180c7c6b218
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Temporarily running out of file descriptors should not terminate the
listener thread. Otherwise, when there becomes more file descriptors
available, nothing is listening.
* Added EMFILE exception to abnormal thread exit.
* Added an abnormal TCP/TLS listener exit error message.
* Closed the TCP/TLS listener socket on abnormal exit so Asterisk does not
appear dead if something tries to connect to the socket.
ASTERISK-26903 #close
Change-Id: I10f2f784065136277f271159f0925927194581b5
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ast_cdr_setuserfield wrote to a fixed length field using strcpy. This could
result in a buffer overrun when called from chan_sip or func_cdr. This patch
adds a maximum bytes written to the field by using ast_copy_string instead.
ASTERISK-26897 #close
patches:
0001-CDR-Protect-from-data-overflow-in-ast_cdr_setuserfie.patch submitted
by Corey Farrell (license #5909)
Change-Id: Ib23ca77e9b9e2803a450e1206af45df2d2fdf65c
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If DESTDIR is set, don't call ldconfig. Assume that DESTDIR is used to
create a binary archive. The ldconfig call should be delegated to the
archive postinst script. This fixes the case where fakeroot wraps 'make
install' causing $EUID to be 0 even though it doesn't have permission to
call ldconfig.
The previous logic in configure.ac to detect and correct libdir
has been removed as it was not completely accurate. CentOS 64-bit
users should again specifiy --libdir=/usr/lib64 when configuring
to prevent install to /usr/lib.
Updated Makefile:check-old-libdir to check for orphans in
lib64 when installing to lib as well as orphans in lib when installing
to lib64.
Updated Makefile and main/Makefile uninstall targets to remove the
orphans using the new logic.
ASTERISK-26705
Change-Id: I51739d4a03e60bff38be719b8d2ead0007afdd51
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The ao2_global_obj_release() function holds an exclusive lock on the
global object while it is being dereferenced. Any destructors that
run during this time that call ao2_global_obj_ref() will deadlock
because a read lock is required.
Instead, we make the global object inaccessible inside of the write
lock and only dereference it once we have released the lock. This
allows the affected destructors to fail gracefully.
While this doesn't completely solve the referenced issue (the error
message about not being able to create an IQ continues to be shown)
it does solve the backtrace spew that accompanied it.
ASTERISK-21009 #close
Reported by: Marcello Ceschia
Change-Id: Idf40ae136b5070dba22cb576ea8414fbc9939385
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ASTERISK-25490 #close
Change-Id: I1c5fc0942c33c96d62b24203aad0f1e1a1a0131f
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This has not worked for some time and is no longer actively maintained.
Change-Id: I5110b0db69c152761b58fa025cb0a53b0e544d99
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The CDR code previously did not allow the user field to be set
from the 'h' extension in the dialplan. This change removes that
limitation and allows it to be set.
ASTERISK-26818
Change-Id: I0fed8a79b5e408bac4e30542b8f33a61c5ed9aa6
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references." into 13
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We aren't validating that the URI we just parsed is a SIP/SIPS one before
trying to access the user, host, and port members of a possibly uninitialized
structure.
Also update the MessageSend documentation to indicate what 'from' formats are
accepted.
ASTERISK-26484 #close
Reported by: Vinod Dharashive
Change-Id: I476b5cc5f18a7713d0ee945374f2a1c164857d30
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Fixed a bug in function "ast_audiohook_write_frame" that checked the
variable other_factory_samples and only flushed the factories, so they
would be in sync, when other_factory_samples > 0. When there is not any
rtp incoming the variable other_factory_samples will be 0, and although
the result of "our_factory_ms - other_factory_ms" may be very large,
this led to the record file not syncing.
ASTERISK-26875 #close
Reported-by: Aaron An
Tested-by: Aaron An
Change-Id: Ia4d890fb8fc1636a7188502bab35f555685aea22
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POSIX does not require getprotobyname() to be thread safe and some
implementations use static memory which causes issues when multiple
threads are used.
Further, our usage of it today is just to ultimately get IPPROTO_TCP
for calls to setsockopt(). So instead we just use IPPROTO_TCP directly.
Change-Id: I2e14e58674808f7ce99b2f5e900d0f90d0d8da48
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Dereferencing struct ast_autochan.chan without first calling
ast_autochan_channel_lock() is unsafe because the pointer could change at
any time due to a masquerade. Unfortunately, ast_autochan_channel_lock()
itself uses struct ast_autochan.chan unsafely and can result in a deadlock
if the original channel happens to get destroyed after a masquerade in
addition to the pointer getting changed.
The problem is more likely to happen with v11 and earlier because
masquerades are used to optimize out local channels on those versions.
However, it could still happen on newer versions if the channel is
executing a dialplan application when the channel is transferred or
redirected. In this situation a masquerade still must be used.
* Added a lock to struct ast_autochan to safely be able to use
ast_autochan.chan while trying to get the channel lock in
ast_autochan_channel_lock(). The locking order is the channel lock then
the autochan lock. Locking in the other direction requires deadlock
avoidance.
* Fix unsafe ast_autochan.chan usages in app_mixmonitor.c.
* Fix unsafe ast_autochan.chan usages in app_chanspy.c.
* app_chanspy.c: Removed unused autochan parameter from next_channel().
ASTERISK-26867
Change-Id: Id29dd22bc0f369b44e23ca423d2f3657187cc592
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Forgetting to indicate an exten is a pattern can cause a crash if the
"pattern" has a character set range. e.g., "9999[3-5]" The crash is due
to a buffer overwrite because the '-' exten eye-candy wasn't removed as
expected and overran the allocated space.
The buffer overwrite is fixed two ways in this patch.
1) Fix ext_strncpy() to distinguish between pattern and non-pattern
extens. Now '-' characters are removed when they are eye-candy and not
when they are part of a pattern character set. Since the function is
private to pbx.c, the return value now returns the number of bytes written
to the destination buffer instead of the strlen() of the final buffer so
the callers that care don't need to add one.
2) Fix callers to ext_strncpy() to supply the correct available buffer
size of the destination buffer.
ASTERISK-26668
Change-Id: I555d97411140e47e0522684062d174fbe32aa84a
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This patch demotes the ERROR message that is displayed when a
nonexistent item is removed from the Stasis cache. The genesis of this
demotion is due to chan_sip's realtime peers and their interaction with
Asterisk's core ast_endpoint code, but ostensibly it could happen from
other channel drivers as well.
Since Mark Michelson already did an excellent job of explaining on this
issue, it is quoted here for posterity:
"Internally, when a realtime peer is retrieved, Asterisk creates an
ast_endpoint structure. When that peer is destroyed, the ast_endpoint is
destroyed as well. Part of the destruction of the ast_endpoint involves
clearing the Stasis cache of all information about that endpoint. The
problem here is that the act of creating the ast_endpoint is not enough
to actually put any information in the Stasis cache. Instead, something
has to happen, such as a state change, in order for the Stasis cache to
have any information about that endpoint. When a device registers,
chan_sip creates an ast_endpoint structure, processes the REGISTER, and
then destroys the ast_endpoint. When the ast_endpoint is destroyed,
there is nothing to destroy in the Stasis cache, so an error message is
emitted. When you use rtcachefriends, ast_endpoint structures persist
for the lifetime of the module and so you do not see this error
message."
ASTERISK-25237 #close
Change-Id: I53cebc6b4a897a1ab9564182b75c177780feff70
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* say.c Changed 'digits/and' to 'vm-and' for en_GB
ASTERISK-26598 #close
Change-Id: If1b713e5daea6f952b339f139178d292a6c4fcfe
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* manager.c:manager_state_cb() Fix potential use of uninitialized hint[]
if a hint does not exist for the requested extension. Ran into this when
developing a testsuite test. The AMI event ExtensionStatus came out with
the hint header value containing garbage. The AMI event PresenceStatus
also had the same issue.
* manager.c:action_extensionstate() no need to completely initialize the
hint[]. Only initialize the first element.
* pbx.c:ast_add_hint() Remove unnecessary assignment.
* chan_sip.c: Eliminate an unneeded hint[] local variable. We only care
about the return value of ast_get_hint() there.
Change-Id: Ia9a8786f01f93f1f917200f0a50bead0319af97b
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... and clean them both up on uninstall.
We've fixed the issue where 'make install' was installing to
/usr/lib on 64-bit systems that use /usr/lib64. Now we need
to clean up the remnants in /usr/lib.
* 'make install' now prints a warning if DESTDIR/ASTLIBDIR
contains 'lib64' and libasterisk* shared libraries or modules
are also found in DESTDIR/ASTLIBDIR with 'lib64' transformed
to 'lib'.
* 'make uninstall' ALWAYS cleans up both DESTDIR/ASTLIBDIR and
DESTDIR/ASTLIBDIR with 'lib64' transformed to 'lib'.
ASTERISK-26705
Change-Id: I6edddeb3c07a51e7c7ba7cac3c05e4bf3ec3f01f
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On some platforms a multiarch approach is used for libraries.
The build system does not take this into account and still
places libraries into the lib directory if no --libdir is
specified to configure. On initial startup this results in
libasteriskssl.so not being found, as it is not in the multiarch
lib directory. To make matters worse, options were being passed
to ldconfig on both Linux and FreeBSD that actually prevented
the rebuild of the cache.
* Fedora has a /usr/share/config.site that automatically tells
autoconf to use /usr/lib64 but CentOS does not. This logic was
copied to configure.ac and modified so systems like Ubuntu,
which still use /usr/lib for 64-bit systems, aren't affected.
Now that we have them in the correct directory...
In order for the system loader to find libasteriskssl and
libasteriskpj, one of 3 things has to happen...
- The linker cache must be rebuilt including the directory
where the libasterisk* libraries were installed. Only root
can rebuild the cache. This was busted.
- We have to link the asterisk binary with an rpath pointing
to the directrory where the libasterisk* libraries were
installed. This makes things very complicated and will happen
over the collective dead bodies of everyone who's had to
package a distribution with an rpath.
- Finally, you can start asterisk with LD_LIBRARY_PATH set to the
directrory where the libasterisk* libraries were installed.
There are no other options. So...
* The invokation of ldconfig has been moved from main/Makefile
to ASTTOPDIR/Makefile, the options have been removed, and
DESTDIR/ASTLIBDIR appended. If you aren't root, you will be
warned after the "Asterisk Installation Compete" banner that
you must re-run 'make install' as root, manually run
'ldconfig DESTDIR/ASTLIBDIR' as root, or run asterisk with
LD_LIBRARY_PATH.
ASTERISK-26705
Change-Id: I2a64b7c33a7d3e9bde20f47e3d3ab771977af982
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This reverts commit d90430953c508670a67de68de400fef44f5e9fba.
Change-Id: I758fe7ea0408f83a6df8e1774310d69f482700f6
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On some platforms a multiarch approach is used for libraries.
The build system does not take this into account and still
places libraries into the lib directory if no --libdir is
specified to configure. On initial startup this results in
libasteriskssl.so not being found, as it is not in the multiarch
lib directory.
This change does the minimally invasive thing and executes
ldconfig so that the libraries in the lib directory are found
and their location cached. By doing so Asterisk starts up fine.
If DESTDIR is specified, however, the old logic is executed as
the install process may not have permission to alter the ldconfig
cache.
ASTERISK-26705
Change-Id: If4eca46ac510c6fea5568256280ffdb3888d7bb4
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This reverts commit e910dbab90ef3d628955c49f441b2c9dda1f222c.
Change-Id: I242aa0a965a79738dc898299959c6d2e020c86bd
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Change-Id: I0ddf01cd3c10d3b6666d7bf68d4e206a37f4fbdb
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On some platforms a multiarch approach is used for libraries.
The build system does not take this into account and still
places libraries into the lib directory if no --libdir is
specified to configure. On initial startup this results in
libasteriskssl.so not being found, as it is not in the multiarch
lib directory.
This change does the minimally invasive thing and executes
ldconfig so that the libraries in the lib directory are found
and their location cached. By doing so Asterisk starts up fine.
ASTERISK-26705
Change-Id: I6d30b6427e9d5e69470e11327c7ff203fa7da519
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ASTERISK-26794 #close
Change-Id: I9cbc3b6b6a8aab590f5ccde9c262a98e4d5253a1
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OpenSSL 1.1 requires no explicit initialization. The hacks in the
library are not needed. They also happen to fail running Asterisk.
ASTERISK-26109 #close
Change-Id: I3b3efd5d80234a4c45a8ee58dcfe25b15d9ad100
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OpenSSL 1.1 introduced TLS_client_method() and deprecated the previous
version-specific methods (such as TLSv1_client_method(). Other than
being simpler to use and more correct (gain support for TLS newer that
TLS1, in our case), the older ones produce a deprecation warning that
fails the build in dev-mode.
ASTERISK-26109 #close
Change-Id: I257b1c8afd09dcb0d96cda3a41cb9f7a15d0ba07
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Use OPENSSL_VERSION_NUMBER instead of OPENSSL_API_COMPAT to detect
the openssl 1.1 API.
ASTERISK-26109 #close
Change-Id: I4e448f55ef516aedf6ad154037c35577a421a458
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* app_minivm: Use built-in completion facilities to complete optional
arguments.
* app_voicemail: Use built-in completion facilities to complete
optional arguments.
* app_confbridge: Add missing colons after 'Usage' text.
* chan_alsa: Use built-in completion facilities to complete optional
arguments.
* chan_sip: Use built-in completion facilities to complete optional
arguments. Add completions for 'load' for 'sip show user', 'sip show
peer', and 'sip qualify peer.'
* chan_skinny: Correct and extend completions for 'skinny reset' and
'skinny show line.'
* func_odbc: Correct completions for 'odbc read' and 'odbc write'
* main/asterisk: Correct and extend completions for 'core show file
version.'
* main/astmm: Use built-in completion facilities to complete arguments
for 'memory' commands.
* main/bridge: Correct completions for 'bridge kick.'
* main/ccss: Use built-in completion facilities to complete arguments
for 'cc cancel' command.
* main/cli: Add 'all' completion for 'channel request hangup.' Correct
completions for 'core set debug channel.' Correct completions for 'core
show calls.'
* main/pbx_app: Remove redundant completions for 'core show
applications.'
* main/pbx_hangup_handler: Remove unused completions for 'core show
hanguphandlers all.'
* res_sorcery_memory_cache: Add completion for 'reload' argument of
'sorcery memory cache stale' and properly implement.
Change-Id: Iee58c7392f6fec34ad9d596109117af87697bbca
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The ast_waitfor_nandfds operation will manipulate the flags
of channels passed in. This was previously done without
the channel lock being held. This could result in incorrect
values existing for the flags if another thread manipulated
the flags at the same time.
This change locks the channel during flag manipulation.
ASTERISK-26788
Change-Id: I2c5c8edec17c9bdad4a93291576838cb552ca5ed
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In Asterisk 11, if the 'Originate' AMI command failed to connect the provided
Channel while in extension mode, a 'failed' extension would be looked up and
run. This was, I believe, unintentionally removed in 51b6c49. This patch
restores that behavior.
This also adds an enum for the various 'synchronous' modes in an attempt to
make them meaningful.
ASTERISK-26115 #close
Reported by: Nasir Iqbal
Change-Id: I8afbd06725e99610e02adb529137d4800c05345d
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We shouldn't unlock the channel after starting a snapshot staging because
another thread may interfere and do its own snapshot staging.
* app_dial.c:dial_exec_full() made hold the channel lock while setting up
the outgoing channel staging. Made hold the channel lock after the called
party answers while updating the caller channel staging.
* chan_sip.c:sip_new() completed the channel staging on off-nominal exit.
Also we need to use ast_hangup() instead of ast_channel_unref() at that
location.
* channel.c:__ast_channel_alloc_ap() added a comment about not needing to
complete the channel snapshot staging on off-nominal exit paths.
* rtp_engine.c:ast_rtp_instance_set_stats_vars() made hold the channel
locks while staging the channels for the stats channel variables.
Change-Id: Iefb6336893163f6447bad65568722ad5d5d8212a
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When performing an SRV lookup using the ast_srv_lookup function it
did not properly handle the situation where 0 records are returned.
If this happened it would wrongly assume that at least one record
was present.
This change fixes the code so it will exit early if an error occurs
or if 0 records are returned.
ASTERISK-26772
patches:
srv_lookup.patch submitted by nappsoft (license 6822)
Change-Id: I09b19081c74e0ad11c12bf54a257243b1bcb2351
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into 13
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If an audiohook is placed on a channel that does not require transcoding,
muting that hook will cause the underlying frames to be muted as well.
The original patch is from David Woolley but I have modified slightly.
ASTERISK-21094 #close
Reported by: David Woolley
Patches:
ASTERISK-21094-Patch-1.8-1.txt (license #5737) patch uploaded
by David Woolley
Change-Id: Ib2b68c6283e227cbeb5fa478b2d0f625dae338ed
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