Occasionally under load we'll attempt to send a final NOTIFY on a
subscription that's already been terminated and a SEGV will occur
down in pjproject's evsub_destroy function. This is a result of a
race condition between all the paths that can generate a notify
and/or destroy the underlying pjproject evsub object:
* The client can send a SUBSCRIBE with Expires: 0.
* The client can send a SUBSCRIBE/refresh.
* The subscription timer can expire.
* An extension state can change.
* An MWI event can be generated.
* The pjproject transaction timer (timer_b) can expire.
Normally when our pubsub_on_evsub_state is called with a terminate,
we push a task to the serializer and return at which point the dialog
is unlocked. This is usually not a problem because the task runs
immediately and locks the dialog again. When the system is heavily
loaded though, there may be a delay between the unlock and relock
during which another event may occur such as the subscription timer
or timer_b expiring, an extension state change, etc. These may also
cause a terminate to be processed and if so, we could cause pjproject
to try to destroy the evsub structure twice. There's no way for us to
tell that the evsub was already destroyed and the evsub's group lock
can't tolerate this and SEGVs.
The remedy is twofold.
* A patch has been submitted to Teluu and added to the bundled
pjproject which adds add/decrement operations on evsub's group lock.
* In res_pjsip_pubsub:
* configure.ac and pjproject-bundled's configure.m4 were updated
to check for the new evsub group lock APIs.
* We now add a reference to the evsub group lock when we create
the subscription and remove the reference when we clean up the
subscription. This prevents evsub from being destroyed before
we're done with it.
* A state has been added to the subscription tree structure so
termination progress can be tracked through the asyncronous tasks.
* The pubsub_on_evsub_state callback has been split so it's not doing
double duty. It now only handles the final cleanup of the
subscription tree. pubsub_on_rx_refresh now handles both client
refreshes and client terminates. It was always being called for
both anyway.
* The serialized_on_server_timeout task was removed since
serialized_pubsub_on_rx_refresh was almost identical.
* Missing state checks and ao2_cleanups were added.
* Some debug levels were adjusted to make seeing only off-nominal
things at level 1 and nominal or progress things at level 2+.
ASTERISK-26099 #close
Reported-by: Ross Beer.
Change-Id: I779d11802cf672a51392e62a74a1216596075ba1
Some configure scripts used both AC_HELP_STRING and its replacement
AS_HELP_STRING. For consistency and to avoid obsolete warnings, those were
changed to AS_HELP_STRING.
ASTERISK-26046
Change-Id: I8aad4fd2bdee40aa2a31ce3339a1eb33ff4f5b0f
When running on a system that does not support or use AST_UNDEFINED_SANITIZER
or AST_LEAK_SANITIZER, the configure script would incorrectly set those
constants to a blank value, e.g., 'AST_UNDEFINED_SANITIZER='. This would
cause menuselect to error out, complaining that a blank value is not a
valid option. This patch corrects the issue by setting the value to 0 if
the options that those constants enable/disable is not found.
Change-Id: Ib39814aaf940f308d500c1e026edb3d70de47fba
The pjsua and pjsystest apps are now built only if TEST_FRAMEWORK is set.
The python bindings are now built only if TEST_FRAMEWORK is set and a
python development package is installed.
libresample was also disabled.
ASTERISK-25993 #close
Reported-by: Joshua Colp
Change-Id: If4e91c503a02f113d5b71bc8b972081fa3ff6f03
For all OSes:
* Disabled third-party codecs in pjproject and added
'--disable-speex-codec --disable-speex-aec --disable-gsm-codec' to the
configure options since we don't use the pjsip codec capability.
FreeBSD:
* Added FreeBSD support to install_prereq.
* Changed pjproject/configure.m4 to use $GNU_MAKE instead of hardcoding "make".
* Added __progname and environ to asterisk.exports.in.
* Reverted the use of ldconfig to create shared library symlinks to ln.
* Only enable epoll in pjproject if `uname -s` is Linux.
* Added a patch to pjproject to take the name of the 'make' command from
an environment variable if supplied. This is needed for the python bindings.
(merged by Teluu into pjproject trunk 5/3/2016)
FreeBSD support isn't complete. Still some general issues regarding
make/gmake having nothing to do with pjproject. With some handholding it DOES
build successfully.
CentOS:
Added 'patch' and 'bzip2' to install_prereq PACKAGES_RH.
CentOS 6/7 32/64 build and run the pjsip testsuite successfully.
Ubuntu:
No changes required.
Ubuntu 15/16 32/64 build and run the pjsip testsuite successfully.
Debian:
No changes required.
Debian 6/7/8 32/64 build and run the pjsip testsuite successfully.
There will utimately be a follow-up patch to create an install_prereq for
the testsuite as I've discovered a few missing requirements.
ASTERISK-25968 #close
Change-Id: I5756a07facfc63798115a5e73a8709382fe9259c
PortAudio should no longer be required
PJSIP_MAX_PKT_LEN is now 6000
Older autoconf issue fixed. (CentOS 6)
Change-Id: I463fa9586cbe7c6b3b603289f535bd8e361611dd
Older versions of PJSIP do not have the proto field on the TLS transport
setting structure. This change adds a configure check so even if it is
not present we will still be able to build.
Change-Id: Ibf3f47befb91ed1b8194bf63888baa6fee05aba9
Background here:
http://lists.digium.com/pipermail/asterisk-dev/2016-January/075266.html
From CHANGES:
* To help insure that Asterisk is compiled and run with the same known
version of pjproject, a new option (--with-pjproject-bundled) has been
added to ./configure. When specified, the version of pjproject specified
in third-party/versions.mak will be downloaded and configured. When you
make Asterisk, the build process will also automatically build pjproject
and Asterisk will be statically linked to it. Once a particular version
of pjproject is configured and built, it won't be configured or built
again unless you run a 'make distclean'.
To facilitate testing, when 'make install' is run, the pjsua and pjsystest
utilities and the pjproject python bindings will be installed in
ASTDATADIR/third-party/pjproject.
The default behavior remains building with the shared pjproject
installation, if any.
Building:
All you have to do is include the --with-pjproject-bundled option on
the ./configure command line (and remove any existing --with-pjproject
option if specified). Everything else is automatic.
Behind the scenes:
The top-level Makefile was modified to include 'third-party' in the
list of MOD_SUBDIRS.
The third-party directory was created to contain any third party
packages that may be needed in the future. Its Makefile automatically
iterates over any subdirectories passing on targets.
The third-party/pjproject directory was created to house the pjproject
source distribution. Its Makefile contains targets to download, patch
configure, generate dependencies, compile libs, apps and python bindings,
sanitized build.mak and generate a symbols list.
When bootstrap.sh is run, it automatically includes the configure.m4
file in third-party/pjproject. This file has a macro to download and
conifgure pjproject and get and set PJPROJECT_INCLUDE, PJPROJECT_DIR
and PJPROJECT_BUNDLED. It also tests for the capabilities like
PJ_TRANSACTION_GRP_LOCK by parsing preprocessor output as opposed to
trying to compile. Of course, bootstrap.sh is only run once and the
configure file is incldued in the patch.
When configure is run with the new options, the macro in configure.m4
triggers the download, patch, conifgure and tests. No compilation is
performed at this time. The downloaded tarball is cached in /tmp so
it doesn't get downloaded again on a distclean.
When make is run in the top-level Asterisk source directory, it will
automatically descend all the subdirectories in third_party just as it
does for addons, apps, etc. The top-level Makefile makes sure that
the 'third-party' is built before 'main' so that dependencies from the
other directories are built first.
When main does build, a new shared library (libasteriskpj) is created that
links statically to the pjproject .a files and exports all their symbols.
The asterisk binary links to that, just as it does with libasteriskssl.
When Asterisk is installed, the pjsua and pjsystest apps, and the pjproject
python bindings are installed in ASTDATADIR/third-party/pjproject. This
will facilitate testing, including running the testsuite which will be
updated to check that directory for the pjsua module ahead of the system
python library.
Modules should continue to depend on pjproject if they use pjproject APIs
directly. They should not care about the implementation. No changes to any
res_pjsip modules were made.
Change-Id: Ia7a60c28c2e9ba9537c5570f933c1ebcb20a3103
Pjproject has deprecated pjsip_dlg_create_uas in 2.5 and replaced it with
pjsip_dlg_create_uas_and_inc_lock which, as the name implies, automatically
increments the lock on the returned dialog. To account for this, configure.ac
now detects the presence of pjsip_dlg_create_uas_and_inc_lock and res_pjsip.c
has an #ifdef HAVE_PJSIP_DLG_CREATE_UAS_AND_INC_LOCK to decide whether to use
the original call or the new one. If the new one was used, the ref count is
decremented before returning.
ASTERISK-25751 #close
Reported-by Josh Colp
Change-Id: I1be776b94761df03bd0693bc7795a75682615ca8
In older versions of the compiler was not sanitizes.
Compilers other than GCC can not support the Usan and TSAN
or have other options for *FLAGS.
ASTERISK-25767 #close
Reported by: Badalyan Vyacheslav
Tested by: Badalyan Vyacheslav
Change-Id: Iefce6608221fa87884b82ae3cb5649b7b1804916
The SSL_OP_NO_TLSv1_1 and SSL_OP_NO_TLSv1_2 defines did not exist prior
to OpenSSL version 1.0.1. A recent commit attempts to, by default, set
these options, which can cause problems on systems with older OpenSSL
installations.
This commit adds a configure script check for those defines and will not
attempt to make use of those if they do not exist. We will print a
warning urging the user to upgrade their OpenSSL installation if those
defines are not present.
Change-Id: I6a2eb9a43fd0738b404d8f6f2cf4b5c22d9d752d
* Add 'check-alembic' target to root Makefile.
* Create build_tools/make_check_alembic to do the actual checks.
ASTERISK-25685
Change-Id: Ibb3cae7d1202ac23dc70b0f3b5801571ad46b004
Versions of libunbound before v1.4.21 do not compile with Asterisk.
However, since v1.4.21 has a configure script bug that fails to detect the
ldns library (which is fixed in v1.4.22) and v1.4.22 is not an easily
detectable version we will require v1.5.0 as a minimum version of the
library to work with Asterisk.
ASTERISK-25108 #close
Reported by: Richard Mudgett
Change-Id: Ieb228bfb01467573fc121c7356a9dde27128894d
clock_gettime() is, unfortunately, not portable. But I did like that
over our usual `ts.tv_nsec = tv.tv_usec * 1000` copy/paste code we
usually do when we want a timespec and all we have is ast_tvnow().
This patch adds ast_tsnow(), which mimics ast_tvnow(), but returns a
timespec. If clock_gettime() is available, it will use that. Otherwise
ast_tsnow() falls back to using ast_tvnow().
Change-Id: Ibb1ee67ccf4826b9b76d5a5eb62e90b29b6c456e
This will add ECDH support to Asterisk. It will
detect auto ECDH support in OpenSSL
(1.0.2b and above) during ./configure. If this is
available, it will use it,
otherwise it will fall back to prime256v1 (this
behavior is consistent with
other projects such as Apache and nginx).
This fixes WebRTC being broken in Firefox 38+ due
to Firefox now only supporting
ciphers with perfect forward secrecy.
ASTERISK-25265 #close
Change-Id: I8c13b33a2a79c0bde2e69e4ba6afa5ab9351465b
GCC 4.7 Manual:
http://gcc.gnu.org/onlinedocs/gcc-4.7.4/gcc/Function-Attributes.html
weakref ("target")
A weak reference is an alias that does not by itself require a definition
to be given for the target symbol.
ASTERISK-22559 #close
Reported by: Ibercom
Change-Id: I36a136cae947b65187a697533416f9ff9a0b8cdf
Created autoconf/ast_check_raii.m4: contains AST_CHECK_RAII which
checks compiler requirements for RAII:
gcc: -fnested-functions support
clang: -fblocks (and if required -lBlocksRuntime)
The original check was implemented in configure.ac and now has it's
own file. This function also sets C_COMPILER_FAMILY to either gcc or
clang for use by makefile
Created autoconf/ast_check_strsep_array_bounds.m4 (contains
AST_CHECK_STRSEP_ARRAY_BOUNDS):
which checks if clang is able to handle the optimized strsep & strcmp
functions (linux). If not, the standard libc implementation should be
used instead. Clang + the optimized macro's work with:
strsep(char *, char []), but not with strsepo(char *, char *).
Instead of replacing all the occurences throughout the source code,
not using the optimized macro version seemed easier
See 'define __strcmp_gc(s1, s2, l2) in bits/string2.h':
llvm-comment: Normally, this array-bounds warning are suppressed for
macros, so that unused paths like the one that accesses __s1[3] are
not warned about. But if you preprocess manually, and feed the
result to another instance of clang, it will warn about all the
possible forks of this particular if statement. Instead of switching
of this optimization, another solution would be to run the preproces-
sing step with -frewrite-includes, which should preserve enough
information so that clang should still be able to suppress the diag-
nostic at the compile step later on.
See also "https://llvm.org/bugs/show_bug.cgi?id=20144"
See also "https://llvm.org/bugs/show_bug.cgi?id=11536"
Makefile.rules: If C_COMPILER_FAMILY=clang then add two warning
suppressions:
-Wno-unused-value
-Wno-parentheses-equality
In an earlier review (reviewboard: 4550 and 4554), they were deemed a
nuisace and less than benefitial.
configure.ac:
Added AST_CHECK_RAII() see earlier
Added AST_CHECK_STRSEP_ARRAY_BOUNDS() see earlier
Removed moved content
ASTERISK-24917
Change-Id: I12ea29d3bda2254ad3908e279b7effbbac6a97cb
This change adds the following:
1. A query set implementation. This is an API that allows queries to be executed in parallel and once all have completed a callback is invoked.
2. Unit tests for the query set implementation.
3. An external PJSIP resolver which uses the DNS core API to do NAPTR, SRV, AAAA, and A lookups.
For the resolver it will do NAPTR, SRV, and AAAA/A lookups in parallel. If NAPTR or SRV
are available it will then do more queries. And so on. Preference is NAPTR > SRV > AAAA/A,
with IPv6 preferred over IPv4. For transport it will prefer TLS > TCP > UDP if no explicit
transport has been provided. Configured transports on the system are taken into account to
eliminate resolved addresses which have no hope of completing.
ASTERISK-24947 #close
Reported by: Joshua Colp
Change-Id: I56cb03ce4f9d3d600776f36928e0b3e379b5d71e
This change adds an abstracted core DNS API which resembles the API described
here[1]. The API provides a pluggable mechanism for resolvers and also a
consistent view for records. Both synchronous and asynchronous queries are
supported.
This change also adds a res_resolver_unbound module which uses the libunbound
library to provide resolution.
Unit tests have also been written for all of the above to confirm the API and
functionality.
ASTERISK-24834 #close
Reported by: Matt Jordan
ASTERISK-24836 #close
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/4474/
Review: https://reviewboard.asterisk.org/r/4512/
[1] https://wiki.asterisk.org/wiki/display/AST/Asterisk+DNS+API
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@433370 65c4cc65-6c06-0410-ace0-fbb531ad65f3
RAII_VAR, which is used extensively in Asterisk to manage reference counted
resources, uses a GCC extension to automatically invoke a cleanup function
when a variable loses scope. While this functionality is incredibly useful
and has prevented a large number of memory leaks, it also prevents Asterisk
from being compiled with clang.
This patch updates the RAII_VAR macro such that it can be compiled with clang.
It makes use of the BlocksRuntime, which allows for a closure to be created
that performs the actual cleanup.
Note that this does not attempt to address the numerous warnings that the clang
compiler catches in Asterisk.
Much thanks for this patch goes to:
* The folks on StackOverflow who asked this question and Leushenko for
providing the answer that formed the basis of this code:
http://stackoverflow.com/questions/24959440/rewrite-gcc-cleanup-macro-with-nested-function-for-clang
* Diederik de Groot, who has been extremely patient in working on getting this
patch into Asterisk.
Review: https://reviewboard.asterisk.org/r/4370/
ASTERISK-24133
ASTERISK-23666
ASTERISK-20399
ASTERISK-20850 #close
Reported by: Diederik de Groot
patches:
RAII_CLANG.patch uploaded by Diederik de Groot (License 6600)
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This patch addresses compilation errors on OS X. It's been a while, so
there's quite a few things.
* Fixed __attribute__ decls in route.h to be portable.
* Fixed htonll and ntohll to work when they are defined as macros.
* Replaced sem_t usage with our ast_sem wrapper.
* Added ast_sem_timedwait to our ast_sem wrapper.
* Fixed some GCC 4.9 warnings using sig*set() functions.
* Fixed some format strings for portability.
* Fixed compilation issues with res_timing_kqueue (although tests still fail
on OS X).
* Fixed menuconfig /sbin/launchd detection, which disables res_timing_kqueue
on OS X).
ASTERISK-24539 #close
Reported by: George Joseph
ASTERISK-24544 #close
Reported by: George Joseph
Review: https://reviewboard.asterisk.org/r/4327/
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The Asterisk 13 configure.ac checks for HAS_WORKING_SEMAPHORE but does not have
an option for cross-compiling so it fails with an exit. Since we're cross-
compiling, we can't exactly go looking for the header. The semaphore.h header
is relatively common:
* It's part of the POSIX standard
* It's part of GNU C Library
As such, we assume that it will be present when cross-compiling.
As such, this patch defaults "HAS_WORKING_SEMAPHORE" to "1" if cross-compiling
is detected.
If you're cross-compiling to a platform that doesn't support this, then make
sure you re-define this to 0.
ASTERISK-24663 #close
Reported by: abelbeck
patches:
asterisk-13-anonymous-semaphores.patch uploaded by abelbeck (License 5903)
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The 'pjsip_get_dest_info' function is used to determine if the signaling transport
of the dialog is secure or not. This function was added in PJSIP 2.3 and does not
exist in earlier versions.
This configure check allows Asterisk to build and run with older versions at the
loss of the 'secure' argument for the PJSIP CHANNEL dialplan function. Usage of
this argument will require upgrading to PJSIP 2.3.
ASTERISK-24665 #close
Reported by: Mark Michelson
Review: https://reviewboard.asterisk.org/r/4329/
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gcc on the ARM platform defaults 'char' to 'unsigned char' whereas Intel and
SPARC default to 'signed char'. This is only an issue in the rare cases where
negative values are assigned to a 'char' but this this patch insures
compatibility by detecting platforms that default to 'unsigned' and adding an
'-fsigned-char' flag to _ASTCFLAGS.
If compiling for ARM (native or cross-compile) be sure to run ./bootstrap.sh
and ./configure to regenerate the build files. You shouldn't have to do this
for Intel or SPARC.
Tested-by: George Joseph
Review: https://reviewboard.asterisk.org/r/4091/
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When configuring Asterisk to build against a version of pjproject installed
in a non-standard location, the checks for "PJSIP Transaction Group Lock
Support" and "PJSIP Media Stream Replacement Support" fail. This is
because these secondary checks are not taking the CFLAGS and LIBS returned
by the pkg-config check into account.
Review: https://reviewboard.asterisk.org/r/3830
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@419077 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The commit that added libxml2 support didn't fully check for the libxml2
development script in the Asterisk configure file. As a result, Asterisk could
be configured, then fail on menuselect. This patch fixes it so that Asterisk
should detect the libxml2 dependency failure first.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@418850 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This is the final patch in adding menuselect to Asterisk.
- The first patch (r418832) added menuselect along with mxml
- The second patch (r418833) removed mxml from menuselect
This patch adds support for libxml2 to menuselect, and makes libxml2 a
required library for Asterisk.
Note that the libxml2 portion of this patch was written by Sean Bright,
and was made available on a team branch:
http://svn.digium.com/svn/menuselect/team/seanbright/libxml2/
Review: https://reviewboard.asterisk.org/r/3773/
ASTERISK-20703 #close
patches:
some_mysterious_team_branch uploaded by seanbright (License 5060)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@418834 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The previous patch (r418034) fixed the 'glitch' that the channels/h323
Makefile no longer existed. Unfortunately, removing the entire line was a bit
of a blunder, as it meant that build_tools/menuselect-deps was never
generated. Hilarity ensued when actually trying to compile.
But hey! At least configure worked.
This patch fixes *that* glitch, and removes some more of the vestiges of h323.
(It had tendrils in the main Makefile? Crazy.)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@418035 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The patch for ASTERISK-23905 that added PFS support in Asterisk depends on the
elliptic curve library support being present in OpenSSL. As it turns out, some
versions of OpenSSL don't have this library - notably the version running on
our build agents.
This patch fixes the build by providing a configure check for the specific
library calls that the PFS patch relies on.
Review: https://reviewboard.asterisk.org/r/3709/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@417900 65c4cc65-6c06-0410-ace0-fbb531ad65f3
ast_ext_tool_check.m4 isn't handling cases where a path to a package is
provided (E.G. --with-mysqlclient=/some/sysroot) and the package has a config
tool (E.G. mysql_config) and the package has its own subdirectories in include
or lib. For example, mysql's libraries are in ${MYSQLCLIENT_DIR}/usr/lib/mysql
but ast_ext_tool_check sets MYSQLCLIENT_LIB to ${MYSQLCLIENT_DIR}/usr/lib.
libxml2 has the same problem with its includes. They're in
${LIBXML2_DIR}/usr/include/libxml2 not directly in ${LIBXML2_DIR}/usr/include.
Both cause configure to fail and there are others in the same boat.
The problem is caused by logic in ast_ext_tool_check that overrides the result
of the config tool's --cflags and --libs options if package_DIR is set.
This patch prepends package_DIR (if specified) to the -L and -I results from
the package's config tool instead of overriding them.
A regenerated ./configure and include/asterisk/autoconfig.h.in are included
but can be regenerated by running ./bootstrap.sh at any time.
Tested by: George Joseph
Tested by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/3550/
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* SS7 support now requires libss7 v2.0 or later. The new libss7 is not
backwards compatible.
* Added SS7 support for connected line and redirecting.
* Most SS7 CLI commands are reworked as well as new SS7 commands added.
See online CLI help.
* Added several SS7 config option parameters described in
chan_dahdi.conf.sample.
* ISUP timer support reworked and now requires explicit configuration.
See ss7.timers.sample.
Special thanks to Kaloyan Kovachev for his support and persistence in
getting the original patch by adomjan updated and ready for release.
SS7-27 #close
Reported by: adomjan
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@416416 65c4cc65-6c06-0410-ace0-fbb531ad65f3
When overlap dialing is enabled, the lack of inband audio available
information in the SETUP_ACKNOWLEDGE events causes an interoperability
problem with SIP. sig_pri doesn't know if there is dialtone present when
a SETUP_ACKNOWLEDGE is received so it assumes it is there and posts an
AST_CONTROL_PROGRESS frame. The SIP channel driver then sends out a 183
Session Progress and blocks the desired 180 Ringing message when the
ALERTING message comes in.
* Made the configure script detect if the installed version of libpri
supports the SETUP_ACKNOWLEDGE enhancements.
* Using the new API, made generate an AST_CONTROL_PROGRESS frame on an
incoming SETUP_ACKNOWLEDGE message when the message indicates inband audio
is present instead of assuming that dialtone is present.
* Using the new API, made SETUP_ACKNOWLEDGE send out an inband audio
available indication only if dialtone is expected. The change also makes
the fallback behaviour of sending the PROGRESS message better by sending
it only if dialtone is expected.
* Changed receiving a PROCEEDING message to not generate an
AST_CONTROL_PROGRESS frame if the progress indication ie indicates
non-end-to-end-ISDN. This helps interoperability with SIP.
* Changed sending a PROCEEDING message in response to an
AST_CONTROL_PROCEEDING frame to not indicate inband audio available. It
was silly to do so anyway because the channel driver doesn't know if
inband audio is even available. This helps interoperability with SIP.
This patch and a corresponding change in libpri work together to allow
Asterisk to control the inband audio available progress indication ie on
the SETUP_ACKNOWLEDGE message when dialtone is present.
AST-1338 #close
Reported by: Tyler Stewart
Review: https://reviewboard.asterisk.org/r/3521/
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There are cases in Asterisk where it might be desirable to lock
a short critical code section but not incur the context switch
and yield penalty of a mutex or rwlock. The primary spinlock
implementations execute exclusively in userspace and therefore
don't incur those penalties. Spinlocks are NOT meant to be a
general replacement for mutexes. They should be used only for
protecting short blocks of critical code such as simple compares
and assignments. Operations that may block, hold a lock, or
cause the thread to give up it's timeslice should NEVER be
attempted in a spinlock.
The first use case for spinlocks is in astobj2 - internal_ao2_ref.
Currently the manipulation of the reference counter is done with
an ast_atomic_fetchadd_int which works fine. When weak reference
containers are introduced however, there's an additional comparison
and assignment that'll need to be done while the lock is held.
A mutex would be way too expensive here, hence the spinlock.
Given that lock contention in this situation would be infrequent,
the overhead of the spinlock is only a few more machine instructions
than the current ast_atomic_fetchadd_int call.
ASTERISK-23553 #close
Review: https://reviewboard.asterisk.org/r/3405/
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SIP transaction group lock support has been backported into our pjproject. Since the code
now internally uses a group lock the code is now changed to unlock it if present. Note
that the act of finding the transaction is what actually returns it locked.
For further information about group locks check out the wiki page at:
http://trac.pjsip.org/repos/wiki/Group_Lock
(issue ASTERISK-22818)
Reported by: Matt Jordan
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When running configure, libiodbc2 development headers will fulfill the
requirement for ODBC development headers, but will not function
properly. This adds a warning when libiodbc2 development headers are
detected instead of unixodbc development headers.
(closes issue ASTERISK-22459)
Reported by: Patrick Maille
Tested by: Walter Doekes
Patches:
issueA22459_warn_when_using_iodbc.patch uploaded by Walter Doekes (License 5674)
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This introduces usage of an additional libxslt cleanup function,
xsltCleanupGlobals, when the configure script detects that it is
available. Early versions of the library did not include this function.
(closes issue ASTERISK-22570)
Reported by: Corey Farrell
Patches:
xsltCleanupGlobals.patch uploaded by Corey Farrell (License 5909)
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r399887 | dlee | 2013-09-26 10:41:47 -0500 (Thu, 26 Sep 2013) | 1 line
Minor performance bump by not allocate manager variable struct if we don't need it
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r400138 | dlee | 2013-09-30 10:24:00 -0500 (Mon, 30 Sep 2013) | 23 lines
Stasis performance improvements
This patch addresses several performance problems that were found in
the initial performance testing of Asterisk 12.
The Stasis dispatch object was allocated as an AO2 object, even though
it has a very confined lifecycle. This was replaced with a straight
ast_malloc().
The Stasis message router was spending an inordinate amount of time
searching hash tables. In this case, most of our routers had 6 or
fewer routes in them to begin with. This was replaced with an array
that's searched linearly for the route.
We more heavily rely on AO2 objects in Asterisk 12, and the memset()
in ao2_ref() actually became noticeable on the profile. This was
#ifdef'ed to only run when AO2_DEBUG was enabled.
After being misled by an erroneous comment in taskprocessor.c during
profiling, the wrong comment was removed.
Review: https://reviewboard.asterisk.org/r/2873/
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r400178 | dlee | 2013-09-30 13:26:27 -0500 (Mon, 30 Sep 2013) | 24 lines
Taskprocessor optimization; switch Stasis to use taskprocessors
This patch optimizes taskprocessor to use a semaphore for signaling,
which the OS can do a better job at managing contention and waiting
that we can with a mutex and condition.
The taskprocessor execution was also slightly optimized to reduce the
number of locks taken.
The only observable difference in the taskprocessor implementation is
that when the final reference to the taskprocessor goes away, it will
execute all tasks to completion instead of discarding the unexecuted
tasks.
For systems where unnamed semaphores are not supported, a really
simple semaphore implementation is provided. (Which gives identical
performance as the original taskprocessor implementation).
The way we ended up implementing Stasis caused the threadpool to be a
burden instead of a boost to performance. This was switched to just
use taskprocessors directly for subscriptions.
Review: https://reviewboard.asterisk.org/r/2881/
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r400180 | dlee | 2013-09-30 13:39:34 -0500 (Mon, 30 Sep 2013) | 28 lines
Optimize how Stasis forwards are dispatched
This patch optimizes how forwards are dispatched in Stasis.
Originally, forwards were dispatched as subscriptions that are invoked
on the publishing thread. This did not account for the vast number of
forwards we would end up having in the system, and the amount of work it
would take to walk though the forward subscriptions.
This patch modifies Stasis so that rather than walking the tree of
forwards on every dispatch, when forwards and subscriptions are changed,
the subscriber list for every topic in the tree is changed.
This has a couple of benefits. First, this reduces the workload of
dispatching messages. It also reduces contention when dispatching to
different topics that happen to forward to the same aggregation topic
(as happens with all of the channel, bridge and endpoint topics).
Since forwards are no longer subscriptions, the bulk of this patch is
simply changing stasis_subscription objects to stasis_forward objects
(which, admittedly, I should have done in the first place.)
Since this required me to yet again put in a growing array, I finally
abstracted that out into a set of ast_vector macros in
asterisk/vector.h.
Review: https://reviewboard.asterisk.org/r/2883/
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r400181 | dlee | 2013-09-30 13:48:57 -0500 (Mon, 30 Sep 2013) | 28 lines
Remove dispatch object allocation from Stasis publishing
While looking for areas for performance improvement, I realized that an
unused feature in Stasis was negatively impacting performance.
When a message is sent to a subscriber, a dispatch object is allocated
for the dispatch, containing the topic the message was published to, the
subscriber the message is being sent to, and the message itself.
The topic is actually unused by any subscriber in Asterisk today. And
the subscriber is associated with the taskprocessor the message is being
dispatched to.
First, this patch removes the unused topic parameter from Stasis
subscription callbacks.
Second, this patch introduces the concept of taskprocessor local data,
data that may be set on a taskprocessor and provided along with the data
pointer when a task is pushed using the ast_taskprocessor_push_local()
call. This allows the task to have both data specific to that
taskprocessor, in addition to data specific to that invocation.
With those two changes, the dispatch object can be removed completely,
and the message is simply refcounted and sent directly to the
taskprocessor.
Review: https://reviewboard.asterisk.org/r/2884/
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With the new work in Asterisk 12, there are some uses of the
optional_api that are prone to failure. The details are rather involved,
and captured on [the wiki][1].
This patch addresses the issue by removing almost all of the magic from
the optional API implementation. Instead of relying on weak symbol
resolution, a new optional_api.c module was added to Asterisk core.
For modules providing an optional API, the pointer to the implementation
function is registered with the core. For modules that use an optional
API, a pointer to a stub function, along with a optional_ref function
pointer are registered with the core. The optional_ref function pointers
is set to the implementation function when it's provided, or the stub
function when it's now.
Since the implementation no longer relies on magic, it is now supported
on all platforms. In the spirit of choice, an OPTIONAL_API flag was
added, so we can disable the optional_api if needed (maybe it's buggy on
some bizarre platform I haven't tested on)
The AST_OPTIONAL_API*() macros themselves remained unchanged, so
existing code could remain unchanged. But to help with debugging the
optional_api, the patch limits the #include of optional API's to just
the modules using the API. This also reduces resource waste maintaining
optional_ref pointers that aren't used.
Other changes made as a part of this patch:
* The stubs for http_websocket that wrap system calls set errno to
ENOSYS.
* res_http_websocket now properly increments module use count.
* In loader.c, the while() wrappers around dlclose() were removed. The
while(!dlclose()) is actually an anti-pattern, which can lead to
infinite loops if the module you're attempting to unload exports a
symbol that was directly linked to.
* The special handling of nonoptreq on systems without weak symbol
support was removed, since we no longer rely on weak symbols for
optional_api.
[1]: https://wiki.asterisk.org/wiki/x/wACUAQ
(closes issue ASTERISK-22296)
Reported by: Matt Jordan
Review: https://reviewboard.asterisk.org/r/2797/
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This patch moves the RESTful URL's around to more appropriate
locations for release.
The /stasis URL's are moved to /ari, since Asterisk REST Interface was
a more appropriate name than Stasis-HTTP. (Most of the code still has
stasis_http references, but they will be cleaned up after there are no
more outstanding branches that would have merge conflicts with such a
change).
A larger change was moving the ARI events WebSocket off of the shared
/ws URL to its permanent home on /ari/events. The Swagger code
generator was extended to handle "upgrade: websocket" and
"websocketProtocol:" attributes on an operation.
The WebSocket module was modified to better handle WebSocket servers
that have a single registered protocol handler. If a client
connections does not specify the Sec-WebSocket-Protocol header, and
the server has a single protocol handler registered, the WebSocket
server will go ahead and accept the client for that subprotocol.
(closes issue ASTERISK-21857)
Review: https://reviewboard.asterisk.org/r/2621/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393528 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The library that provides UUID support varies greatly from system to
system. On most Linux distros, it's in libuuid. On OpenBSD, it's in
libe2fs-uuid. On OS X, it is in libsystem.
This patch plays hide-and-seek with UUID support, looking for it in the
three places we know about. It also corrects the Makefile so that it uses
the configured library name and include path.
(closes issue ASTERISK-21816)
Reported by: Brad Latus (snuffy)
Tested by: Brad Latus (snuffy)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@390352 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The pimp_my_sip branch is being merged at this point because
it offers basic functionality, and from an API standpoint, things
are complete.
SIP work is *not* feature-complete; however, with the completion
of the SUBSCRIBE/NOTIFY API, all APIs (except a PUBLISH API) have
been created, and thus it is possible for developers to attempt
to create new SIP work.
API documentation can be found in the doxygen in the code, but
usability documentation is still lacking.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@386540 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch started out simply as fixing the bouncing tests introduced
in r382685, but required some other changes to give it a decent
implementation.
To fix the bouncing tests, the UserEvent and Newexten AMI events
needed to be refactored to dispatch via Stasis. Dispatching directly
to AMI resulted in those events sometimes getting ahead of the
associated Newchannel events, which would understandably confuse anyone.
I found that instead of creating a zillion different message types and
structures associated with them, it would be preferable to define a
message type that has a channel snapshot and a blob of structured data
with a small bit of additional information. The JSON object model
provides a very nice way of representing structured data, so I went
with that.
* Move JSON support from res_json.c to main/json.c
* Made libjansson-dev a required dependency
* Added an ast_channel_blob message type, which has a channel
snapshot and JSON blob of data.
* Changed UserEvent and Newexten events so that they are dispatched
via ast_channel_blob messages on the channel's topic.
* Got rid of the ast_channel_varset message; used ast_channel_blob
instead.
* Extracted the manager functions converting Stasis channel events to
AMI events into manager_channel.c.
(issue ASTERISK-21096)
Review: https://reviewboard.asterisk.org/r/2381/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@383579 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Building Asterisk on Raspbian with hard-float support fails as it uses the
string 'linux-gnueabihf' for host os, as opposed to 'linux-gnueabi'. This patch
modifies the configure script for Asterisk such that it will match on any
string beginning with 'linux-gnueabi', as opposed to requiring an explicit
match.
(closes issue ASTERISK-21006)
Reported by: Christian Hesse
Tested by: Christian Hesse
patches:
linux-gnueabihf.patch uploaded by Christian Hesse (license 6459)
linux-gnueabihf-autoconf.patch uploaded by Christian Hesse (license 6459)
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With ptlib 2.10.9, the configure script fails due to grep returning multiple
matches for the pattern it searches for. This patch updates the pattern
matching to return only the actual version for the symbol searched for,
PTLIB_VERSION.
(closes issue ASTERISK-20980)
Reported by: Stefan Reuter
patches:
ASTERISK-20980-1.patch uploaded by Stefan Reuter (license 5339)
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* This allows us to remove some special-case build logic.
* 10.5 is down to less that 8% of the OS X market share. 10.4 is down to
under 2%.
* Apple is no longer releasing security updates for 10.5 and earlier.
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This provides a JSON API by pulling in and wrapping the Jansson JSON
library[1]. The Asterisk API basically mirrors the Jansson
functionality, with a few minor tweaks.
* Some names have been asteriskified to protect the innocent.
* Jansson provides both reference-stealing and reference-borrowing
versions of several API's. The Asterisk API is exclusively
reference-stealing for operations that put elements into arrays and
objects.
* No support for doubles, since we usually don't need that.
* Coming along for the ride is the ast_test_validate macro, which made
the unit tests much easier to write.
[1]: http://www.digip.org/jansson/
(issue ASTERISK-20887)
(closes issue ASTERISK-20888)
Review: https://reviewboard.asterisk.org/r/2264/
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This provides a common API for dealing with unique identifiers.
The API provides methods to create, parse, copy, and stringify UUIDs.
An accompanying unit test is provided that tests all operations.
(closes issue ASTERISK-20726)
reported by Matt Jordan
Review: https://reviewboard.asterisk.org/r/2217
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Make git more attractive for managing work-in-progress. Especially
convenient when a potential patch set needs to be tested on multiple
platforms since one can use git to keep all the test environments in sync
independent of a subversion server.
Now the Asterisk version will show the exact git SHA5 that was used when
building (still appended by "M" if there are local modifications) from a
git clone of the Asterisk repository so the developer can more easily know
what is actually under test.
You will now get this:
$ asterisk -V
Asterisk GIT-1698298
Instead of this:
$ asterisk -V
Asterisk UNKNOWN__and_probably_unsupported
This has zero impact for those not using git with the exception of an
extra test in the configure script to gather git's path. This is
necessary to prevent "sudo make install" from failing since git may not be
in the path in make's shell environment.
(closes issue ASTERISK-20483)
Reported by: Shaun Ruffell
Patches:
0001-build_tools-Allow-Asterisk-to-report-git-SHAs-in-ver.patch (license #5417) patch uploaded by Shaun Ruffell
Modified
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The autoconf configuration system had a test for DOT but not for Doxygen. I added the test for Doxygen and did an overhaul of the Makefile check to a much simpler process.
(issue ASTERISK-20259)
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As mentioned on the review for this, WebRTC has moved towards choosing
DTLS-SRTP as the mechanism for key exchange for SRTP. This commit adds
support for this but makes it available for normal SIP clients as well.
Testing has been done to ensure that this introduces no regressions with
existing behavior and also that it functions as expected.
Review: https://reviewboard.asterisk.org/r/2113/
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In order to use nested functions on some versions of GCC (e.g. GCC on OS X),
the -fnested-functions flag must be passed to the compiler. This patch adds
detection logic to ./configure to add the flag if necessary. It also adds
a comment to utils.h as to why the nested function needs a prototype.
(closes issue ASTERISK-20399)
Reported by: David M. Lee
Review: https://reviewboard.asterisk.org/r/2102/
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This patch changes the Asterisk configure script and build system to detect
the presence of the NetBSD editline library (libedit) on the system. If it is
found, it will be used in preference to the version included in the Asterisk
source tree.
(closes issue ASTERISK-18725)
Reported by: Jeffrey C. Ollie
Review: https://reviewboard.asterisk.org/r/1528/
Patches:
0001-Allow-linking-building-against-an-external-editline.patch uploaded by jcollie (license #5373) (heavily modified by kpfleming)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370481 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This patch adds the core changes necessary to support AMI event documentation
in the source files of Asterisk, and adds documentation to those AMI events
defined in the core application modules. Event documentation is built from
the source by two new python scripts, located in build_tools:
get_documentation.py and post_process_documentation.py.
The get_documentation.py script mirrors the actions of the existing AWK
get_documentation scripts, except that it will scan the entirety of a source
file for Asterisk documentation. Upon encountering it, if the documentation
happens to be an AMI event, it will attempt to extract information about the
event directly from the manager event macro calls that raise the event. The
post_process_documentation.py script combines manager event instances that
are the same event but documented in multiple source files. It generates
the final core-[lang].xml file.
As this process can take longer to complete than a typical 'make all', it
is only performed if a new make target, 'full', is chosen.
Review: https://reviewboard.asterisk.org/r/1967/
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AST_PKG_CONFIG_CHECK: Similar to AST_EXT_LIB_CHECK, but simply uses
pkg-config data.
This simple version only uses pkg-config(1)'s tests.
This commit also uses the macro to test for GTK2 and GMIME (instead of
the current direct usage of pkg-config).
Review: https://reviewboard.asterisk.org/r/1906/
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Passing -Wshadow to gcc enables shadow warnings. From the gcc manual:
Warn whenever a local variable or type declaration shadows another
variable, parameter, type, or class member (in C++), or whenever a
built-in function is shadowed.
Asterisk will not currently compile with this option set, but a number of bugs
have been discovered by enabling this flag on specific files. The long-term
goal is to eliminate all of the suspect code that causes this warning to be
emitted.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@358622 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The principle difference between libvorbisfile v1.1.2 and newer (at least
v1.2.0) is the addition of the predefined callbacks OV_CALLBACKS_xxx in
vorbis/vorbisfile.h used for ov_open_callbacks().
* Updated the configure script to detect if libvorbisfile.h declares
OV_CALLBACKS_NOCLOSE.
* Copied the declaration of OV_CALLBACKS_NOCLOSE from v1.2.0 to allow
v1.1.2 to compile.
(closes issue ASTERISK-19370)
Reported by: Jonn Taylor
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Ogg/vorbis was fairly useless as a voicemail file format because it did
not implement the seek and tell format callbacks among other problems.
Since we were already using the libvorbis and libvorbisenc libraries we
can use libvorbisfile as it is also part of the vorbis library package.
* Made use the libvorbisfile to handle the ogg/vorbis file stream. The
format_ogg_vorbis.c is now mostly a wrapper around libvorbisfile.
(closes issue ASTERISK-16926)
Reported by: sque
Patches:
ogg_vorbis_use_libvorbisfile.patch (license #6108) patch uploaded by sque
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This patch removes res_ais and introduces a new module, res_corosync.
The OpenAIS project is deprecated and is now just a wrapper around
Corosync. This module provides the same functionality using the same
core infrastructure, but without the use of the deprecated components.
Technically res_ais could have been used with an AIS implementation other
than OpenAIS, but that is the only one I know of that was ever used.
Review: https://reviewboard.asterisk.org/r/1700/
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When Asterisk is used with various third-party libraries (CURL, PostgresSQL,
many others) that have the ability themselves to use OpenSSL, it is possible
for conflicts to arise in how the OpenSSL libraries are initialized and
shutdown. This patch addresses these conflicts by 'wrapping' the important
functions from the OpenSSL libraries in a new shared library that is part
of Asterisk itself, and is loaded in such a way as to ensure that *all*
calls to these functions will be dispatched through the Asterisk wrapper
functions, not the native functions.
This new library is optional, but enabled by default. See the CHANGES file
for documentation on how to disable it.
Along the way, this patch also makes a few other minor changes:
* Changes MODULES_DIR to ASTMODDIR throughout the build system, in order to
more closely match what is used during run-time configuration.
* Corrects some errors in the configure script where AC_CHECK_TOOLS was used
instead of AC_PATH_PROG.
* Adds a new variable for linker flags in the build system (DYLINK), used for
producing true shared libraries (as opposed to the dynamically loadable
modules that the build system produces for 'regular' Asterisk modules).
* Moves the Makefile bits that handle installation and uninstallation of the
main Asterisk binary into main/Makefile from the top-level Makefile.
* Moves a couple of useful preprocessor macros from optional_api.h to
asterisk.h.
Review: https://reviewboard.asterisk.org/r/1006/
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r339720 | rmudgett | 2011-10-06 17:58:40 -0500 (Thu, 06 Oct 2011) | 27 lines
Merged revisions 339719 via svnmerge from
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r339719 | rmudgett | 2011-10-06 17:47:50 -0500 (Thu, 06 Oct 2011) | 20 lines
Fix regression in configure script for libpri capability checks.
JIRA AST-598 added the PRI_L2_PERSISTENCE option to fix BRI PTMP TE layer
2 persistence issues with some telcos. ASTERISK-18535 attempted to fix
the unexpected requirement that libpri *must* have that feature to work
with Asterisk. The AST_EXT_LIB_SETUP_DEPENDENT lines made the PRI
optional features required. Unfortunately, I thought
AST_EXT_LIB_SETUP_DEPENDENT didn't do anything useful for libpri and
deleted those lines for libpri. The result was the HAVE_PRI_xxx defines
that control the ability to use optional libpri features were also
deleted.
* Created AST_EXT_LIB_SETUP_OPTIONAL configuration macro to allow optional
features in a library that the source code could take advantage of if the
code supports the feature.
(closes issue ASTERISK-18687)
Reported by: Norbert
Tested by: rmudgett
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r336734 | tilghman | 2011-09-19 15:29:40 -0500 (Mon, 19 Sep 2011) | 18 lines
Merged revisions 336733 via svnmerge from
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r336733 | tilghman | 2011-09-19 15:27:03 -0500 (Mon, 19 Sep 2011) | 11 lines
Various changes to allow 1.8 to compile on Mac OS X Lion (10.7)
* Makefile workaround for 10.6 extended to work on 10.7 and later.
* Now uses the 'weak' symbol for Lion systems, which no longer support
'weak_import'
Closes ASTERISK-17612.
Closes ASTERISK-18213.
Tested by: tilghman, oej.
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r335912 | rmudgett | 2011-09-14 13:31:15 -0500 (Wed, 14 Sep 2011) | 20 lines
Merged revisions 335911 via svnmerge from
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r335911 | rmudgett | 2011-09-14 13:21:35 -0500 (Wed, 14 Sep 2011) | 13 lines
Remove unnecessary libpri dependency checks in the configure script.
Using the --with-pri option with the configure script generated an error
about not having PRI_L2_PERSISTENCE if you did not have the absolute
latest libpri SVN checkout installed.
The AST_EXT_LIB_SETUP_DEPENDENT macro in the configure.ac script seems to
be for libraries that are dependent upon other libraries and not
necessarily for optional/added features within a library.
(closes issue ASTERISK-18535)
Reported by: Michael Keuter
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r332369 | tilghman | 2011-08-17 14:24:59 -0500 (Wed, 17 Aug 2011) | 17 lines
Merged revisions 332355 via svnmerge from
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r332355 | tilghman | 2011-08-17 14:21:36 -0500 (Wed, 17 Aug 2011) | 10 lines
Re-add support for spaces in pathnames, including now spaces in DESTDIR.
This was initially added to 1.8 prior to release, primarily to support the
standard paths on Mac OS X, but was partially reverted recently in Subversion,
due to the lack of support for spaces in DESTDIR. This commit restores support
for the standard paths on Mac OS X, and also includes support for spaces in
DESTDIR.
(closes issue ASTERISK-18290)
Reported by: pabelanger
Review: https://reviewboard.asterisk.org/r/1326/
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r332265 | rmudgett | 2011-08-17 11:01:29 -0500 (Wed, 17 Aug 2011) | 33 lines
Merged revisions 332264 via svnmerge from
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r332264 | rmudgett | 2011-08-17 10:51:08 -0500 (Wed, 17 Aug 2011) | 26 lines
Outgoing BRI calls fail when using Asterisk 1.8 with HA8, HB8, and B410P cards.
France Telecom brings layer 2 and layer 1 down on BRI lines when the line
is idle. When layer 1 goes down Asterisk cannot make outgoing calls and
the HA8 and HB8 cards also get IRQ misses.
The inability to make outgoing calls is because the line is in red alarm
and Asterisk will not make calls over a line it considers unavailable.
The IRQ misses for the HA8 and HB8 card are because the hardware is
switching clock sources from the line which just brought layer 1 down to
internal timing.
There is a DAHDI option for the B410P card to not tell Asterisk that layer
1 went down so Asterisk will allow outgoing calls: "modprobe wcb4xxp
teignored=1". There is a similar DAHDI option for the HA8 and HB8 cards:
"modprobe wctdm24xxp bri_teignored=1". Unfortunately that will not clear
up the IRQ misses when the telco brings layer 1 down.
* Add layer 2 persistence option to customize the layer 2 behavior on BRI
PTMP lines. The new option has three settings: 1) Use libpri default
layer 2 setting. 2) Keep layer 2 up. Bring layer 2 back up when the peer
brings it down. 3) Leave layer 2 down when the peer brings it down.
Layer 2 will be brought up as needed for outgoing calls.
JIRA AST-598
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There were some bugs in the very ancient version of Berkeley DB that Asterisk
used. Instead of spending the time tracking down the bugs in the Berkeley code
we move to the much better documented SQLite 3.
Conversion of the old astdb happens at runtime by running the included
astdb2sqlite3 utility. The ast_db API with SQLite 3 backend should behave
identically to the old Berkeley backend, but in the future we could offer a
much more robust interface.
We do not include the SQLite 3 library in the source tree, but instead rely
upon the distribution-provided libraries. SQLite is so ubiquitous that this
should not place undue burden on administrators.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@326589 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The NEC SV8300 rejects the Q931_IE_TIME_DATE for Q.SIG.
Add option to specify if and how much of the current time is put in
Q931_IE_TIME_DATE.
* Send date/time ie never.
* Send date/time ie date only.
* Send date/time ie date and hour.
* Send date/time ie date, hour, and minute.
* Send date/time ie date, hour, minute, and second.
* Send date/time ie default: Libpri will send date and hhmm only when in
NT PTMP mode to support ISDN phones.
(closes issue #19221)
Reported by: kenner
JIRA SWP-3396
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The "controlling user number" is always the number of the voice mail box
which is identical with the subscriber number itself. This number which
is listed in the ISDN phone MWI menu cannot be called back to contact the
voice mail box. The controlling user number should be made configurable.
JIRA ABE-2738
JIRA SWP-2846
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Recent versions of GCC have a tuning option value of 'native', which causes
the compiler to optimize the build for the CPU the compile is performed on.
Since most people are building Asterisk on the machine they plan to run it on,
the configure script and build system will now use this value unless a different
value is specified by the user in CFLAGS when the configure script is executed.
In addition, this value will be used for building the GSM and LPC10 codecs as
well, in preference to the logic that has been in their Makefiles forever to
optimize for certain types of CPUs.
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r309808 | tilghman | 2011-03-06 18:54:42 -0600 (Sun, 06 Mar 2011) | 14 lines
Merged revisions 309251 via svnmerge from
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r309251 | tilghman | 2011-03-01 19:06:02 -0600 (Tue, 01 Mar 2011) | 7 lines
Revert previous 2 commits, and instead conditionally redefine the same macro used in flex 2.5.35 that clashed with our workaround.
Not surprisingly, the workaround was exactly the same code as was provided by
the Flex maintainers, albeit in two different places, in different macros.
This should fix the FreeBSD builds, which have an older version of Flex.
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r309035 | tilghman | 2011-02-28 05:10:28 -0600 (Mon, 28 Feb 2011) | 15 lines
Merged revisions 309033-309034 via svnmerge from
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r309033 | tilghman | 2011-02-28 04:43:12 -0600 (Mon, 28 Feb 2011) | 4 lines
A later version of flex already includes the fwrite workaround code, which if used twice causes a compilation error.
Detect whether Flex will compile without the workaround; if so, suppress our workaround code.
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r309034 | tilghman | 2011-02-28 05:07:52 -0600 (Mon, 28 Feb 2011) | 2 lines
Clarify meaning, removing double negative (stupid!)
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The display ie handling can be controlled independently in the send and
receive directions with the following options:
* Block display text data.
* Use display text in SETUP/CONNECT messages for name.
* Use display text for COLP name updates (FACILITY/NOTIFY as appropriate).
* Pass arbitrary display text during a call. Sent in INFORMATION
messages. Received from any message that the display text was not used as
a name.
If the display options are not set then the options default to legacy
behavior.
The arbitrary display text is exchanged between bridged channels using the
AST_FRAME_TEXT frame type.
To send display text from the dialplan use the SendText() application when
the arbitrary display text option is enabled.
JIRA SWP-2688
JIRA ABE-2693
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r304466 | qwell | 2011-01-27 11:03:01 -0600 (Thu, 27 Jan 2011) | 23 lines
Merged revisions 304465 via svnmerge from
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r304465 | qwell | 2011-01-27 11:01:24 -0600 (Thu, 27 Jan 2011) | 16 lines
Merged revisions 304464 via svnmerge from
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r304464 | qwell | 2011-01-27 10:57:46 -0600 (Thu, 27 Jan 2011) | 9 lines
Fix default prefix=/usr regression on non-Linux systems.
This partially reverts a change made in branches/1.4/ r267759, which will
cause issue #17013 to be reopened. This issue was pointed out by a user
on #asterisk, who helpfully discovered that paths were being set incorrectly.
To truly understand what was wrong, one should run:
svn diff --force -c<this revision> configure
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r301221 | pabelanger | 2011-01-09 16:40:34 -0500 (Sun, 09 Jan 2011) | 21 lines
Merged revisions 301220 via svnmerge from
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r301220 | pabelanger | 2011-01-09 16:38:24 -0500 (Sun, 09 Jan 2011) | 14 lines
SOUND_CACHE_DIR now defaults to empty
Sounds files included in the Asterisk tarball were being ignored and
re-downloaded. Users wanting to cache the files can still override the setting
using the --with-sounds-cache option.
(closes issue #18589)
Reported by: pabelanger
Patches:
issue18589.patch uploaded by pabelanger (license 224)
Tested by: pabelanger
Review: https://reviewboard.asterisk.org/r/1074/
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r298960 | tilghman | 2010-12-17 17:52:04 -0600 (Fri, 17 Dec 2010) | 20 lines
Merged revisions 298957 via svnmerge from
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r298957 | tilghman | 2010-12-17 17:30:55 -0600 (Fri, 17 Dec 2010) | 13 lines
Merged revisions 298905 via svnmerge from
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r298905 | tilghman | 2010-12-17 15:40:56 -0600 (Fri, 17 Dec 2010) | 6 lines
Let Asterisk find better backtrace information with libbfd.
The menuselect option BETTER_BACKTRACES, if enabled, will use libbfd to search
for better symbol information within both the Asterisk binary, as well as
loaded modules, to assist when using inline backtraces to track down problems.
Review: https://reviewboard.asterisk.org/r/1055/
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Already had the pthread ID which is not the same. The most obvious enhancement
is in the "core show threads" output. As stated in the utils header, if the
platform isn't supported -1 is reported (instead of the process ID previously).
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r296534 | tilghman | 2010-11-29 01:28:44 -0600 (Mon, 29 Nov 2010) | 20 lines
Merged revisions 296533 via svnmerge from
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r296533 | tilghman | 2010-11-29 01:27:09 -0600 (Mon, 29 Nov 2010) | 13 lines
I love standards. There are so many to choose from. Except when there isn't one.
Linux and *BSD disagree on the elements within the ucred structure. Detect
which one is in use on the system.
(closes issue #18384)
Reported by: bjm
Patches:
cred-diffs uploaded by bjm (license 473)
20101127__issue18384__1.6.2.diff.txt uploaded by tilghman (license 14)
20101127__issue18384__1.8.diff.txt uploaded by tilghman (license 14)
Tested by: tilghman, bjm
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r285931 | tilghman | 2010-09-09 20:25:50 -0500 (Thu, 09 Sep 2010) | 21 lines
Merged revisions 285930 via svnmerge from
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r285930 | tilghman | 2010-09-09 20:16:32 -0500 (Thu, 09 Sep 2010) | 14 lines
Merged revisions 285889 via svnmerge from
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r285889 | tilghman | 2010-09-09 19:13:45 -0500 (Thu, 09 Sep 2010) | 7 lines
Fix Mac OS X build.
This also fixes a rather grievous calculation error for the offset of
ast_fdset, which was masked on Linux and FreeBSD, because these platforms
check the first 256 FDs regardless of the bitmask setting (due to backwards
compatibility).
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r282200 | twilson | 2010-08-13 11:00:02 -0500 (Fri, 13 Aug 2010) | 10 lines
Detect when libsrtp cannot be linked in a shared library
The libsrtp build system currently does not produce a shared library
or a static library compiled with -fPIC, so on 64-bit systems it is
possible that we will get a compile error if libsrtp is installed and
res_srtp is selected in menuselect.
This patch attempts to detect this situation and provide the user with
instructions to work around the problem.
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r282201 | twilson | 2010-08-13 11:02:20 -0500 (Fri, 13 Aug 2010) | 2 lines
Whitespace fix :-/
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r279953 | russell | 2010-07-27 16:16:05 -0500 (Tue, 27 Jul 2010) | 5 lines
Add --enable-coverage option to configure script.
This option enables the proper compiler flags for tracking code coverage, which
is useful along side automated testing.
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r279504 | mmichelson | 2010-07-26 11:04:09 -0500 (Mon, 26 Jul 2010) | 14 lines
Allow for systems without locale support to be usable.
A recent change to SIP URI comparison code added a locale-specific
string comparison to the mix, and certain systems do not support
such functions. This fix allows for those systems to still use
Asterisk 1.8
(closes issue #17697)
Reported by: pprindeville
Patches:
asterisk-trunk-bugid17697.patch uploaded by pprindeville (license 347)
Tested by: mmichelson
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Theoretically the ./configure script is a pure bourne-shell script.
Practically it may be run by bash if /bin/sh is not good enough. But we should not count on it. See bug report for the gory details.
(closes issue #17485)
Patches:
0001-remove-bashism-from-ast_check_pwlib.m4.patch uploaded by tzafrir (license 46)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@270184 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This way the libraries can be found even if they are in
non-standard locations.
(closes issue #16155)
Reported by: jcollie
Patches:
0008-change-configure.ac-to-look-for-pkg-config-gmime-2.0.patch uploaded by jcollie (license 412)
Tested by: jsmith, tilghman, pabelanger
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@270042 65c4cc65-6c06-0410-ace0-fbb531ad65f3
After 5 years in mantis and over a year on reviewboard, SRTP support is finally
being comitted. This includes generic CHANNEL dialplan functions that work for
getting the status of whether a call has secure media or signaling as defined
by the underlying channel technology and for setting whether or not a new
channel being bridged to a calling channel should have secure signaling or
media. See doc/tex/secure-calls.tex for examples.
Original patch by mikma, updated for trunk and revised by me.
(closes issue #5413)
Reported by: mikma
Tested by: twilson, notthematrix, hemanshurpatel
Review: https://reviewboard.asterisk.org/r/191/
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Add the ability to announce a call to an endpoint when there are no B
channels available. A call waiting call is a SETUP message with no B
channel selected.
Relevant specification: EN 300 056, EN 300 057, EN 300 058
For DAHDI/ISDN channels, the CHANNEL() dialplan function now supports the
"no_media_path" option.
* Returns "0" if there is a B channel associated with the call.
* Returns "1" if no B channel is associated with the call. The call is
either on hold or is a call waiting call.
If you are going to allow incoming call waiting calls then you need to use
CHANNEL(no_media_path) do determine if you must drop a call to accept the
new call.
Review: https://reviewboard.asterisk.org/r/568/
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Added ability to send and receive ETSI Explicit Call Transfer (ECT)
messages to eliminate tromboned calls.
Note: Asterisk already supported initiating the transfer of calls to
eliminate tromboned calls to libpri so there was nothing to do for the
asterisk portion.
Review: https://reviewboard.asterisk.org/r/520/
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Newer versions of libical (which we require) store the header file in a
libical/ subfolder and include an ical.h file that does a #warning for
deprecation and then #includes <libical/ical.h>. Since we now test for
libical/ical.h, we can change the #includes back to <libical/ical.h> and
remove the test which specifically adds /usr/include/libical as an include
directory.
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This uses a modified version of pabelanger's patch that checks for NTLM support
instead, which was added in 0.29.0 which is what is required for
res_calendar_ews.
(closes issue #17391)
Reported by: loloski
Patches:
issue17391.patch.v2 uploaded by pabelanger (license 224)
Tested by: twilson
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@265793 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This ensures cross-platform compatibility, even among Linux distributions,
which don't always put headers in the same place.
(closes issue #17391)
Reported by: loloski
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@265747 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r264248 | tilghman | 2010-05-19 12:41:29 -0500 (Wed, 19 May 2010) | 17 lines
Internal timing is now on by default, if you're using DAHDI 2.3 or above.
The reason for ensuring DAHDI 2.3 or above is that this version ensures that
a timer is always available, whereas in previous versions, it was possible
for DAHDI to be loaded, but have no drivers to actually generate timing. If
internal_timing was turned on in this circumstance, a complete lack of audio
would result. This is the reason why internal_timing was not on by default.
However, now that DAHDI ensures the availability of a timer, there is no
reason for this setting to be off (and in fact, it solves a great many initial
user problems).
(closes issue #15932)
Reported by: dimas
Patches:
20100519__issue15932.diff.txt uploaded by tilghman (license 14)
Tested by: tilghman
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@264249 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This will save a considerable amount of CPU on the BSDs, including Mac OS X,
as it eliminates several places in the code that we previously used a busy
loop. Additionally, this adds a res_timing interface, using kqueue timers.
Review: https://reviewboard.asterisk.org/r/543/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@262852 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Revision -r1489 of the libpri 1.4 branch corrected a deviation from Q.931
Section 5.3.2. However, this resulted in an unexpected behaviour change
to the upper layer (Asterisk).
This change uses pri_hangup_fix_enable() to follow Q.931 Section 5.3.2
call hangup better if the version of libpri supports it.
(issue #17104)
Reported by: shawkris
Tested by: rmudgett
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@262569 65c4cc65-6c06-0410-ace0-fbb531ad65f3
We nicely detect the right flags on each system for building Asterisk with
pthreads, then ignore it for every other build option that requires us to
build with pthreads. This caused some items to return a false negative.
Also cleanup some minor naming issues that caused "library library" redundancy
in the output.
(closes issue #17303)
Reported by: stuarth
Patches:
20100507__issue17303.diff.txt uploaded by tilghman (license 14)
Tested by: stuarth
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@261913 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r259748 | qwell | 2010-04-28 14:17:38 -0500 (Wed, 28 Apr 2010) | 7 lines
Remove usage of `id` since it isn't useful and was causing breakge.
Solaris `id` doesn't support the -u argument. Instead of figuring out how to
fix this to work on Solaris, I decided to check why it was necessary and where
else it was used. It was only used in one place, and it hasn't been needed
for a very long time (I question whether it was ever needed).
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@259760 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r259352 | qwell | 2010-04-27 14:29:26 -0500 (Tue, 27 Apr 2010) | 5 lines
Support the silly OSes that don't have ar and strip.
Since AC_PATH_TOOL is equiv to AC_CHECK_TOOL when path isn't specified, and
AC_PATH_TOOLS doesn't exist, we'll just switch to AC_CHECK_TOOLS.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@259353 65c4cc65-6c06-0410-ace0-fbb531ad65f3
These changes add the ability to run 'make asterisk.txt' just like the existing
'make asterisk.pdf' commands to generate a text document from the TeX files we
have in the doc/tex/ directory. I've also updated a few of the .tex files because
they weren't properly escaping certain characters so they would show up as Unicode
characters (like [U+021C]). Made changes to the configure scripts so it would
detect the catdvi program which is required to convert the .dvi file generated
by latex.
I've also added a few lines to the build_tools/prep_tarball script so that the
text documentation gets generated and added to future tarballs of Asterisk
releases.
(closes issue #17220)
Reported by: lmadsen
Patches:
asterisk.txt.patch uploaded by lmadsen (license 10)
asterisk.txt.patch-v4 uploaded by pabelanger (license 224)
Tested by: lmadsen, pabelanger
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@258351 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Detect all platforms that don't like that, either, and ensure that when documentation is
missing, we pass a non-NULL pointer when outputting the corresponding documentation.
(closes issue #16689)
Reported by: bklang
Patches:
20100209__issue16689__with_tests.diff.txt uploaded by tilghman (license 14)
Review: https://reviewboard.asterisk.org/r/497/
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@246030 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r242520 | tilghman | 2010-01-24 00:33:01 -0600 (Sun, 24 Jan 2010) | 8 lines
Only rebuild bison and flex source files on demand, if bison and flex are detected by the configure script.
Changed after discussion on the -dev list about possible unnecessary build
failures, due to checkouts/untars causing these special source files to
possibly be newer than their resulting C files. This should additionally
ensure that nobody need learn about extra Makefile arguments to ensure the
proper files get rebuilt when changes are made to these special source files.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@242521 65c4cc65-6c06-0410-ace0-fbb531ad65f3