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path: root/include/asterisk/autoconfig.h.in
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2015-08-10Merge "Replaces clock_gettime() with ast_tsnow()"Joshua Colp
2015-08-07Replaces clock_gettime() with ast_tsnow()David M. Lee
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
2015-07-29res/res_rtp_asterisk: Add ECDH supportMark Duncan
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
2015-05-03Update configure.ac/Makefile for clangDiederik de Groot
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
2015-04-15res_pjsip: Add external PJSIP resolver implementation using core DNS API.Joshua Colp
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
2015-03-25dns: Add core DNS API + unit tests and res_resolver_unbound module + unit tests.Joshua Colp
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
2015-01-26Various fixes for OS XDavid M. Lee
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/ ........ Merged revisions 431092 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@431093 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2015-01-16Add support for the ca_list_path option for PJSIP transports.Mark Michelson
This allows for a path to be specified that has a collection of CA certificates in it. ASTERISK-24575 #close Reported by cloos Patches: pj-ca-path-trunk.diff uploaded by cloos (License #5956) Review: https://reviewboard.asterisk.org/r/4344 ........ Merged revisions 430709 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430713 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2015-01-13chan_pjsip: Add configure check for 'pjsip_get_dest_info' function.Joshua Colp
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/ ........ Merged revisions 430546 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430547 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-10-27configure: Add autoconf check for libopus.Sean Bright
Because opus transcoding support cannot be included in the standard Asterisk distribution, a few codec_opus implementations have popped up. To make it easier for people to drop in opus support in their own installations, this patch adds configure checks for libopus. Review: https://reviewboard.asterisk.org/r/4106/ ........ Merged revisions 426234 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@426235 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-10-19build: Force -fsigned-char on platforms where the default for char is unsignedGeorge Joseph
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/ ........ Merged revisions 425964 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 425965 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@425966 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-07-17menuselect: Add libxml2 support (Patch 3)Matthew Jordan
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
2014-07-03main/tcptls: Add checks for OpenSSL Elliptic Curve supportMatthew Jordan
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
2014-06-20build: Allow autoconf/ast_ext_tool_check to handle cross-compiling better.George Joseph
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/ ........ Merged revisions 416929 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 416930 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 416931 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@416935 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-05-13chan_dahdi/sig_pri: Prevent unnecessary PROGRESS events when overlap dialing ↵Richard Mudgett
is enabled. 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/ ........ Merged revisions 413714 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 413765 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 413771 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@413772 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-04-23This patch adds support for spinlocks in Asterisk.George Joseph
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/ ........ Merged revisions 412976 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@412977 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-03-05Corrected cross-platform stat nanosecond codeDavid M. Lee
When nanosecond time resolution was added for identifying config file changes, it didn't cover all of the myriad of ways that one might obtain nanosecond time resolution off of struct stat. Rather than complicate the #if even further figuring out one system from the next, this patch directly tests for the three struct members I know about today, and #ifdef's accordingly. Review: https://reviewboard.asterisk.org/r/3273/ ........ Merged revisions 409833 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 409834 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 409835 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@409836 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-01-28cdr_radius, cel_radius: build agains libfreeradius-clientKevin Harwell
Asterisk's RADIUS module currently build against libradiusclient-ng, but this project has been superseeded by libfreeradius-client. The API is 99% compatible except that the header name has changed, the library name has changed, and the configuration file location has changed. (closes issue ASTERISK-22980) Reported by: Jeremy Lainé Patches: freeradius-client.patch uploaded by sharky (license 6561) ........ Merged revisions 406801 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 406802 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 406803 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@406825 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-12-03res_pjsip_session: Add support for PJMEDIA_SDP_NEG_ALLOW_MEDIA_CHANGE flag.Joshua Colp
Newer versions of PJSIP have changed to using a flag for the PJMEDIA_SDP_NEG_ALLOW_MEDIA_CHANGE instead of a define. This adds a configure check to detect the presence of the flag and use it if found. ........ Merged revisions 403329 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403330 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-11-26Fix a configure issue with PJSIP transaction group lock detection.Joshua Colp
The configure check did not use the provided paths for pjproject if provided when looking for transaction group lock support. ........ Merged revisions 403160 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403161 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-11-16res_pjsip: Add support for building against pjproject with SIP transaction ↵Joshua Colp
group lock support. 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 ........ Merged revisions 402864 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@402865 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-10-03Detect and use xsltCleanupGlobals when availableKinsey Moore
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) ........ Merged revisions 400384 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@400385 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-09-30Multiple revisions 399887,400138,400178,400180-400181David M. Lee
........ 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 ........ 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/ ........ 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/ ........ 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/ ........ 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/ ........ Merged revisions 399887,400138,400178,400180-400181 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@400186 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-08-30optional_api: Fix linking problems between modules that export global symbolsDavid M. Lee
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/ ........ Merged revisions 397989 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397990 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-08-23Add the bucket API.Joshua Colp
Bucket is a URI based API for the creation, retrieval, updating, and deletion of "buckets" and files contained within them. Review: https://reviewboard.asterisk.org/r/2715/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397600 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-08-01Fix documentation replication issuesKinsey Moore
This prevents XML documentation duplication by expanding channel and bridge snapshot tags into channel and bridge snapshot parameter sets with a given prefix or defaulting to no prefix. This also prevents documentation from becoming fractured and out of date by keeping all variations of the documentation in template form such that it only needs to be updated once and keeps maintenance to a minimum. Review: https://reviewboard.asterisk.org/r/2708/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@395985 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-07-03Shuffle RESTful URL's around.David M. Lee
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
2013-06-03Correct autoconf script for finding UUID support.David M. Lee
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
2013-05-15Use srtp_shutdown when availableKinsey Moore
This allows the SRTP library to be shut down properly when the functionality is offered by libsrtp. Review: https://reviewboard.asterisk.org/r/2538/ (closes issue ASTERISK-21719) ........ Merged revisions 388768 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 388769 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@388770 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-04-25Merge the pimp_my_sip branch into trunk.Mark Michelson
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
2013-03-22Move more channel events to Stasis; move res_json.c to main/json.c.David M. Lee
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
2013-03-12Switch to using external pjproject libraries.Jason Parker
ICE/STUN/TURN support in res_rtp_asterisk is also now optional. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@382900 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-01-29Update configure script to be compatible with ptlib 2.10.9Matthew Jordan
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) ........ Merged revisions 380297 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 380298 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@380299 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-01-19Add builtin roundf() for systems lacking it.Walter Doekes
(closes issue ASTERISK-16854) Review: https://reviewboard.asterisk.org/r/2276 Reported-by: Ovidiu Sas ........ Merged revisions 379547 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 379548 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@379549 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2013-01-11Add JSON API for Asterisk.David M. Lee
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/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@378915 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-12-13Bail configure if it can't find libuuid.David M. Lee
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@377981 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-12-13Remove compile time check HAVE_DEV_URANDOM.Russell Bryant
The code was doing a runtime check, anyway. The compile time check isn't always valid (cross-compiling, packages). Review: https://reviewboard.asterisk.org/r/2245/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@377977 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-12-13Fixed configure.ac to look for proper uuid.h fileDavid M. Lee
Introduced in r377846, the configure script was looking for uuid.h instead of uuid/uuid.h. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@377972 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-12-11Add UUID support to Asterisk.Mark Michelson
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 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@377846 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-10-18build_tools: Allow Asterisk to report git SHAs in version string.Richard Mudgett
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 ........ Merged revisions 375189 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 375190 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged revisions 375191 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@375192 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-09-20Add support for DTLS-SRTP to res_rtp_asterisk and chan_sip.Joshua Colp
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/ ........ Merged revisions 373229 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@373234 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-09-18Add -fnested-functions compile flag, if needed.David M. Lee
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/ ........ Merged revisions 373119 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@373120 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-08-09Use better libss7 detection test and move libpri compile test.Richard Mudgett
........ Merged revisions 371012 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 371013 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@371030 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-07-25Enable usage of system-provided NetBSD editline library if available.Kevin P. Fleming
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
2012-07-23Enable usage of system-provided iLBC library.Kevin P. Fleming
The WebRTC version of the iLBC codec is now package as a library and is available on some platforms. This patch allows codec_ilbc to be built against that library if it is present. Review: https://reviewboard.asterisk.org/r/1964/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370407 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-04-28res_corosync: Fix build against corosync 2.0.Russell Bryant
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@364444 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-03-13Remove chan_usbradio and app_rpt.Russell Bryant
These modules are being maintained outside of the tree and have been for a long time now, so it doesn't make sense to keep them here. Review: https://reviewboard.asterisk.org/r/1764/ ........ Merged revisions 359050 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 359051 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@359052 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-02-16Fix compile problem when old version of libvorbisfile v1.1.2 is used.Richard Mudgett
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 ........ Merged revisions 355608 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 355620 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@355621 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-02-14Re-commit the verbose branch.Tilghman Lesher
This change permits each verbose destination (consoles, logger) to have its own concept of what the verbosity level is. The big feature here is that the logger will now be able to capture a particular verbosity level without condemning each console to need to suffer that level of verbosity. Additionally, a stray 'core set verbose' will no longer change what will go to the log. Review: https://reviewboard.asterisk.org/r/1599/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@355413 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-02-14Fix voicemail problems when using ogg/vorbis.Richard Mudgett
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 ........ Merged revisions 355365 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 355375 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@355376 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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