asterisk

Commit Graph

Author	SHA1	Message	Date
Richard Mudgett	a7e9ee3a7b	Fix CCSS CLI commands and logger level not unregistered. (issue ASTERISK-20649) Reported by: Corey Farrell Patches: ccss-cleanup-all.patch (license #5909) patch uploaded by Corey Farrell ........ Merged revisions 377037 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 377038 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@377039 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Joshua Colp	5935aaa6ec	Fix an RTP instance reference count leak in chan_motif. When setting up an RTP instance the RTCP portion of the instance keeps a reference to the instance itself. In order to release this reference and stop RTCP the stop API call must be called before destroying the instance. (closes issue ASTERISK-20751) Reported by: joshoa git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@377021 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Joshua Colp	1d83d32489	Tweak extension used for incoming calls received on Motif. Based on feedback from numerous individuals this patch tweaks incoming calls to first look for an extension with the name of the endpoint. If no such extension exists the call will silently fall back to the "s" extension as it previously did. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@376983 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	b8a4eff072	chan_misdn: Fix sending RELEASE_COMPLETE in response to SETUP. Fix sending a RELEASE_COMPLETE in response to a SETUP if chan_misdn does not have a B channel available to assign to the call. (closes issue ABE-2869) Reported by: Guenther Kelleter Patches: setup-reject_2.diff (license #6372) patch uploaded by Guenther Kelleter Modified ........ Merged revision 376949 from https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier ........ Merged revisions 376950 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 376951 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@376952 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Sean Bright	258edecea8	Minor spelling fix to the VOLUME documentation. ........ Merged revisions 376919 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 376920 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@376921 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Mark Michelson	da951d0855	Fix potential crashes during SIP attended transfers. The principal behind this patch is simple. During a transfer, we manipulate channels that are owned by a separate thread than the one we currently are running in, so it makes sense that we need to grab a reference to the channels so that they cannot disappear out from under us. In the wild, crashes were sometimes seen when the transferring party would hang up the call before the transfer target answered the call. The most common place to see the crash occur was when attempting to send a connected line update to the transferer channel. (closes issue ASTERISK-20226) Reported by Jared Smith Patches: ASTERISK-20226.patch uploaded by Mark Michelson (License #5049) Tested by: Jared Smith ........ Merged revisions 376901 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 376916 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@376917 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	0c6a476bb3	chan_local: Fix local_pvt ref leak in local_devicestate(). Regression introduced by ASTERISK-20390 fix. (closes issue ASTERISK-20769) Reported by: rmudgett Tested by: rmudgett ........ Merged revisions 376868 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 376869 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@376870 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	903a942b85	Fix compile error. (issue ASTERISK-20724) ........ Merged revisions 376864 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 376865 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@376866 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Michael L. Young	bb38f97269	Improve Code Readability And Fix Setting natdetected Flag For 1.8, 10, 11 and trunk we are are improving the code readability. For 11 and trunk, auto nat detection was added. The natdetected flag was being set to 1 when the host address in the VIA header did not specifiy a port. This patch fixes this by setting the port on the temporary sock address used to SIP_STANDARD_PORT in order for the sock address comparison to work properly. (closes issue ASTERISK-20724) Reported by: Michael L. Young Patches: asterisk-20724-set-port-v2.diff uploaded by Michael L. Young (license 5026) Review: https://reviewboard.asterisk.org/r/2206/ ........ Merged revisions 376834 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 376835 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@376836 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Pedro Kiefer	ed6c432874	Fix chan_sip websocket payload handling Websocket by default doesn't return an ast_str for the payload received. When converting it to an ast_str on chan_sip the last character was being omitted, because ast_str functions expects that the given length includes the trailing 0x00. payload_len only has the actual string length without counting the trailing zero. For most cases this passed unnoticed as most of SIP messages ends with \r\n. (closes issue ASTERISK-20745) Reported by: Iñaki Baz Castillo Review: https://reviewboard.asterisk.org/r/2219/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@376822 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	c223b79688	Add MALLOC_DEBUG atexit unreleased malloc memory summary. * Adds the following CLI commands to control MALLOC_DEBUG reporting of unreleased malloc memory when Asterisk is shut down. memory atexit list on memory atexit list off memory atexit summary byline memory atexit summary byfunc memory atexit summary byfile memory atexit summary off * Made check all remaining allocated region blocks atexit for fence violations. * Increased the allocated region hash table size by about three times. It still isn't large enough considering the number of malloced blocks Asterisk uses. * Made CLI "memory show allocations anomalies" use regions_check_all_fences(). Review: https://reviewboard.asterisk.org/r/2196/ ........ Merged revisions 376788 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 376789 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@376790 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	b6101cfec7	Enhance MALLOC_DEBUG CLI commands. * Fixed CLI "memory show allocations" misspelling of anomalies option. The command will still accept the original misspelling. * Miscellaneous tweaks to CLI "memory show allocations" command output format. * Made CLI "memory show summary" summarize by line number instead of by function if a filename is given. * Made CLI "memory show summary" sort its output by filename or function-name/line-number depending upon request. * Miscellaneous tweaks to CLI "memory show summary" command output format. ........ Merged revisions 376758 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 376759 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@376760 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Jonathan Rose	f4e00404e7	manager: Make challenge work with allowmultiplelogin=no Prior to this patch, challenge would yield a multiple logins error if used without providing the username (which isn't really supposed to be an argument to challenge) if allowmultiplelogin was set to no because allowmultiplelogin finds a user with a zero length login name. This check is simply disabled for the challenge action when the username is empty by this patch. (closes issue ASTERISK-20677) Reported by: Vladimir Patches: challenge_action_nomultiplelogin.diff uploaded by Jonathan Rose (license 6182) ........ Merged revisions 376725 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 376726 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@376727 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	8a32488929	Fix extension matching with the '-' char. The '-' char is supposed to be ignored by the dialplan extension matching. Unfortunately, it's treatment is not handled consistently throughout the extension matching code. * Made the old exten matching code consistently ignore '-' chars. * Made the old exten matching code consistently handle case in the matching. * Made ignore empty character sets. * Fixed ast_extension_cmp() to return -1, 0, or 1 as documented. The only user of it in pbx_lua.c was testing for -1. It was originally returning the strcmp() value for less than which is not usually going to be -1. * Fix character set sorting if the sets have the same number of characters and start with the same character. Character set [0-9] now sorts before [02-9a] as originally intended. * Updated some extension label and priority already in use warnings to also indicate if the extension is aliased. (closes issue ASTERISK-19205) Reported by: Philippe Lindheimer, Birger "WIMPy" Harzenetter Tested by: rmudgett Review: https://reviewboard.asterisk.org/r/2201/ ........ Merged revisions 376688 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 376689 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@376690 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	37b50fe4e5	Remove unnecessary channel module references. * Removed call to ast_module_user_hangup_all() in res_config_mysql.c since it is effectively a noop. No channels can attach a reference to that module. * Removed call to ast_module_user_hangup_all() in app_celgenuserevent.c. The caller of unload_module() has already called it. * Removed redundant channel module references in pbx_dundi.c. The registered dialplan function callback dispatchers for the read/read2/write callbacks already reference the module before calling. * pbx_dundi: Moved unregistering CLI commands, DUNDi switch, and dialplan functions to the first thing the unload_module() does. This will reduce the chance of new channels using DUNDi services while the module is being torn down. ........ Merged revisions 376657 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 376658 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@376659 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	224dd2f60a	Made AST_LIST_REMOVE() simpler and use better names. * Update doxygen of AST_LIST_REMOVE(). ........ Merged revisions 376627 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 376628 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@376629 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	c4f013c5c8	Re-initialize logmsgs mutex upon logger initialization to prevent lock errors Similar to the patch that moved the fork earlier in the startup sequence to prevent mutex errors in the recursive mutex surrounding the read/write thread registration lock, this patch re-initializes the logmsgs mutex. Part of the start up sequence before forking the process into the background includes reading asterisk.conf; this has to occur prior to the call to daemon in order to read startup parameters. When reading in a conf file, log statements can be generated. Since this can't be avoided, the mutex instead is re-initialized to ensure a reset of any thread tracking information. This patch also includes some additional debugging to catch errors when locking or unlocking the recursive mutex that surrounds locks when the DEBUG_THREADS build option is enabled. DO_CRASH or THREAD_CRASH will cause an abort() if a mutex error is detected. (issue ASTERISK-19463) Reported by: mjordan Tesetd by: mjordan ........ Merged revisions 376586 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 376587 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@376588 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
David M. Lee	4c924bdce1	Added missing newlines to websocket ast_logs. Without these newlines, log messages just continue tacking onto the same line, and do not flush immediately. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@376561 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Mark Michelson	0cc3b6cd9b	Add "Require: timer" to 200 OK responses when appropriate. The method by which the Require header is added to 200 responses is inspired by the method that Olle Johansson uses in his darjeeling-prack branch. (closes issue ASTERISK-20570) Reported by Matt Jordan, at the behest of Olle Johansson Review: https://reviewboard.asterisk.org/r/2172 ........ Merged revisions 376521 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 376522 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@376550 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Alec L Davis	4e76aa4920	Reduce CLI spam of "Extension Changed" device state messages. Asterisk 11 follows RFC3265 that states that after every subscribe or resubscribe a notify should be sent. Thus the console if filled continuously with the following after every subscribe; == Extension Changed 8512[phones] new state IDLE for Notify User cisco1 In Asterisk 1.8 only changes would be sent. Thus only when a device state changed was anything emitted to the console. fix: Only print to console when device state isn't forced. (closes issue ASTERISK-20706) Reported by: alecdavis Tested by: alecdavis alecdavis (license 585) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@376540 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Walter Doekes	65c8d16d79	Fix most leftover non-opaque ast_str uses. Instead of calling str->str, one should use ast_str_buffer(str). Same goes for str->used as ast_str_strlen(str) and str->len as ast_str_size(str). Review: https://reviewboard.asterisk.org/r/2198 ........ Merged revisions 376469 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 376470 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@376471 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	2bb01eb0ff	Reorder startup sequence to prevent lockups when process is sent to background Although it is very rare and timing dependent, the potential exists for the call to 'daemon' to cause what appears to be a deadlock in Asterisk during startup. This can occur when a recursive mutex is obtained prior to the daemon call executing. Since daemon uses fork to send the process into the background, any threading primitives are unsafe to re-use after the call. Implementations of pthread recursive mutexes are highly likely to store the thread identifier of the thread that previously obtained the mutex. If the mutex was locked prior to the fork, a subsequent unlock operation will potentially fail as the thread identifier is no longer valid. Since the mutex is still locked, all subsequent attempts to grab the mutex by other threads will block. This behavior exhibited itself most often when DEBUG_THREADS was enabled, as this compile time option surrounds the mutexes in Asterisk with another recursive mutex that protects the storage of thread related information. This made it much more likely that a recursive mutex would be obtained prior to daemon and unlocked after the call. This patch does the following: a) It backports a patch from Asterisk 11 that prevents the spawning of the localtime monitoring thread. This thread is now spawned after Asterisk has fully booted. b) It re-orders the startup sequence to call daemon earlier during Asterisk startup. This limits the potential of threading primitives being accessed by initialization calls before daemon is called. c) It removes calls to ast_verbose/ast_log/etc. prior to daemon being called. Developers should send error messages directly to stderr prior to daemon, as calls to ast_log may access recursive mutexes that store thread related information. d) It reorganizes when thread local storage is created for storing lock information during the creation of threads. Prior to this patch, the read/write lock protecting the list of threads in ast_register_thread would utilize the lock in the thread local storage prior to it being initialized; this patch prevents that. On a very related note, this patch will greatly improve the stability of the Asterisk Test Suite. Review: https://reviewboard.asterisk.org/r/2197 (closes issue ASTERISK-19463) Reported by: mjordan Tested by: mjordan ........ Merged revisions 376428 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 376431 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@376441 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	3d2e80d93b	Add a test event that reports changes in ConfBridge state This patch adds a test event to ConfBridge that reports transitions between states in ConfBridge. This is used by tests in the Asterisk Test Suite that verify state changes based on the entering/leaving of conference participants. ........ Merged revisions 376414 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@376415 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Jonathan Rose	c0c552de2c	monitor: prevent attempts to move/remove recordings skipped with 'i' and 'o'. The i and o options for monitor skip the input and output sides of a recording respectively. This patch addresses a problem in those options when monitor is called without specifying a specific filename where monitor will try to move the recording that was skipped. Since this usually doesn't exist when these options are used, it would produce a warning when it does this in most cases, but it is conceivable that there are use cases where this could result in moving/removing a file unintentionally. (closes issue ASTERISK-20641) Reported by: Jonathan Rose Review: https://reviewboard.asterisk.org/r/2190/ ........ Merged revisions 376389 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 376390 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@376391 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
David M. Lee	c40b60f970	Fixed extconf.c breakage introduced in r376306. To quote wdoekes: > Note that I'm not confirming legitimacy of having that file in tree at > all. Is anyone using aelparse/conf2ael? ........ Merged revisions 376340 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 376342 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@376343 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
David M. Lee	bbf3526314	Migrate hashtest/hashtest2 to be unit tests. Both hashtest and hashtest2 are manual testing apps that thrash hash tables (hashtab and ao2 containers, respectively), by spinning up several threads that randomly insert, delete, lookup and iterate over the hash table. If the app doesn't crash, the hash table probably passes the test. Those utils are not a part of the typical Asterisk build, so they do not usually get compiled. This all makes them less that useful. This patch removes those manual test programs and replaces them with Asterisk unit test modules (test_{hashtab,astobj2}_thrash.so). It also attempts to make the tests more deterministic. * Rather than spinning up some number of threads that operate on the hash table randomly, spin up four threads that concurrenly add, remove, lookup and iterate over the hash table. * Each thread checks the state of the hash table both during and after execution, and indicates a test failure if things are not as expected. * Each thread times out after 60 seconds to prevent deadlocking the unit test run. (closes issue ASTERISK-20505) Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/2189/ ........ Merged revisions 376306 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 376315 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@376339 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Jonathan Rose	0c77685b91	app_meetme: Fix channels lingering when hung up under certain conditions Channels would get stuck and MeetMe would repeatedly display an Unable to write frame to channel error in the conf_run function if hung up during certain sound prompts such as during user count announcements. This patch fixes that by reintroducing a hangup check in the meetme's main loop (also in conf_run). (closes issue ASTERISK-20486) Reported by: Michael Cargile Review: https://reviewboard.asterisk.org/r/2187/ Patches: meetme_hangup_patch_ASTERISK-20486_v3.diff uploaded by Jonathan Rose (license 6182) ........ Merged revisions 376307 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 376308 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@376310 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Rusty Newton	5d858e92ac	Patch to play correct sound file when a voicemail's urgent status is removed We were attempting to play "vm-urgent-removed", which didn't exist. Now we play "vm-marked-nonurgent" which exists and is the correct sound file. Previous behavior was silence and a warning on the CLI. (issue ASTERISK-20280) (closes issue ASTERISK-20280) Reported by: Tomo Takebe Tested by: Rusty Newton Patches: asterisk20280.patch uploaded by Rusty Newton (license 5829) ........ Merged revisions 376262 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 376263 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@376264 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	52d8a7dee4	Fix call files when astspooldir is relative. Future dated call files are ignored when astspooldir is relative to the current directory. The queue_file() assumed that the qdir needed to be prepended if the given filename did not start with a '/'. If astspooldir is relative it is not going to start from the root directory obviously so it will not start with a '/'. The filename used in queue_file() ultimately results in qdir prepended multiple times. * Made queue_file() not prepend qdir if the filename contains a '/'. (closes issue ASTERISK-20593) Reported by: James Le Cuirot Patches: 0004-Fix-future-call-files-from-relative-directories.patch (license #6439) patch uploaded by James Le Cuirot ........ Merged revisions 376232 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 376233 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@376234 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Brent Eagles	852bc686c9	Patch to prevent stopping the active generator when it is not the silence generator. This patch introduces an internal helper function to safely check whether the current generator is the one that is expected before deactivating it. The current externally accessible ast_channel_stop_generator() function has been modified to be implemented in terms of the new function. (closes issue ASTERISK-19918) Reported by: Eduardo Abad ........ Merged revisions 376199 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 376208 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@376217 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Joshua Colp	4e04ea29c7	Properly check if the "Context" and "Extension" headers are empty in a ShowDialPlan action. The code which handles the ShowDialPlan action wrongly assumed that a non-NULL return value from the function which retrieves headers from an action indicates that the header has a value. This is incorrect and the contents must be checked to see if they are blank. (closes issue ASTERISK-20628) Reported by: jkroon Patches: asterisk-showdialplan-incorrect-error.patch uploaded by jkroon ........ Merged revisions 376166 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 376167 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@376168 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Michael L. Young	1255a271d4	Fix Dynamic Hints Variable Substition - Underscore Problem When adding a dynamic hint, if an extension contains an underscore no variable subsitution is being performed. This patch changes from checking if the extension contains an underscore to checking if the extension begins with an underscore. (closes issue ASTERISK-20639) Reported by: Steven T. Wheeler Tested by: Steven T. Wheeler, Michael L. Young Patches: asterisk-20639-dynamic-hint-underscore.diff uploaded by Michael L. Young (license 5026) Review: https://reviewboard.asterisk.org/r/2188/ ........ Merged revisions 376142 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 376143 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@376144 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Joshua Colp	fb74294b92	Remove a fixed size limitation for producing SDP and change how ICE support is disabled by default. With ICE support enabled in chan_sip and a large number of interfaces on the system it was possible for the produced SDP to be truncated due to some fixed size buffers. These buffers have now been changed so they will dynamically grow as needed. ICE support is now also enabled by default in res_rtp_asterisk to provide a smoother experience for chan_motif users where it is required. To maintain the previous behavior in chan_sip it is no longer enabled by default there. (closes issue ASTERISK-20643) Reported by: coopvr git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@376130 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Mark Michelson	0e7a64e55e	Fix a "set but not used" warning on newer gccs. Turns out the "helpful" setting of ms and res in this macro is completely useless after the timeout antipattern fix. If you're a new guy looking to write code, don't write a macro like this one. ........ Merged revisions 376087 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 376088 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@376089 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	c7e2bf3187	chan_dahdi/SS7: Made reject incoming call for an in-alarm or blocked channel. If a SS7 call comes in requesting a CIC that is in-alarm, the call is accepted and connects if the extension exists in the dialplan. The call does not have any audio. * Made release the call immediately with circuit congestion cause. (closes issue ASTERISK-20204) Reported by: Tuan Le Patches: jira_asterisk_20204_v1.8.patch (license #5621) patch uploaded by rmudgett ........ Merged revisions 376058 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 376059 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@376060 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	1bba4b00b2	Add MALLOC_DEBUG enhancements. * Makes malloc() behave like calloc(). It will return a memory block filled with 0x55. A nonzero value. * Makes free() fill the released memory block and boundary fence's with 0xdeaddead. Any pointer use after free is going to have a pointer pointing to 0xdeaddead. The 0xdeaddead pointer is usually an invalid memory address so a crash is expected. * Puts the freed memory block into a circular array so it is not reused immediately. * When the circular array rotates out a memory block to the heap it checks that the memory has not been altered from 0xdeaddead. * Made the astmm_log message wording better. * Made crash if the DO_CRASH menuselect option is enabled and something is found. * Fixed a potential alignment issue on 64 bit systems. struct ast_region.data[] should now be aligned correctly for all platforms. * Extracted region_check_fences() from __ast_free_region() and handle_memory_show(). * Updated handle_memory_show() CLI usage help. Review: https://reviewboard.asterisk.org/r/2182/ ........ Merged revisions 376029 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 376030 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@376048 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Mark Michelson	a65fbf8012	Multiple revisions 375993-375994 ........ r375993 \| mmichelson \| 2012-11-07 11:01:13 -0600 (Wed, 07 Nov 2012) \| 30 lines Fix misuses of timeouts throughout the code. Prior to this change, a common method for determining if a timeout was reached was to call a function such as ast_waitfor_n() and inspect the out parameter that told how many milliseconds were left, then use that as the input to ast_waitfor_n() on the next go-around. The problem with this is that in some cases, submillisecond timeouts can occur, resulting in the out parameter not decreasing any. When this happens thousands of times, the result is that the timeout takes much longer than intended to be reached. As an example, I had a situation where a 3 second timeout took multiple days to finally end since most wakeups from ast_waitfor_n() were under a millisecond. This patch seeks to fix this pattern throughout the code. Now we log the time when an operation began and find the difference in wall clock time between now and when the event started. This means that sub-millisecond timeouts now cannot play havoc when trying to determine if something has timed out. Part of this fix also includes changing the function ast_waitfor() so that it is possible for it to return less than zero when a negative timeout is given to it. This makes it actually possible to detect errors in ast_waitfor() when there is no timeout. (closes issue ASTERISK-20414) reported by David M. Lee Review: https://reviewboard.asterisk.org/r/2135/ ........ r375994 \| mmichelson \| 2012-11-07 11:08:44 -0600 (Wed, 07 Nov 2012) \| 3 lines Remove some debugging that accidentally made it in the last commit. ........ Merged revisions 375993-375994 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 375995 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@376014 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	647810f1d0	Fix stuck DTMF when bridge is broken. When a bridge is broken by an AMI Redirect action or the ChannelRedirect application, an in progress DTMF digit could be stuck sending forever. * Made simulate a DTMF end event when a bridge is broken and a DTMF digit was in progress. (closes issue ASTERISK-20492) Reported by: Jeremiah Gowdy Patches: bridge_end_dtmf-v3.patch.txt (license #6358) patch uploaded by Jeremiah Gowdy Modified to jira_asterisk_20492_v1.8.patch jira_asterisk_20492_v1.8.patch (license #5621) patch uploaded by rmudgett Tested by: rmudgett Review: https://reviewboard.asterisk.org/r/2169/ ........ Merged revisions 375964 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 375965 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@375966 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Joshua Colp	5f28931a1f	Fix a bug where our Motif ICE candidates were not quite proper, and make us more forgiving. An issue was reported on the mailing list where calling would result in an "Incomplete ICE-UDP candidate received on session" error message. This is the result of the ICE-UDP candidate code not placing a "network" attribute within the candidates. This is now done. To increase compatibility though I have removed the requirement for the "network" attribute to exist within ICE-UDP candidates that are received since we don't actually require the value. Reported on the mailing list by Jean-Denis Girard. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@375925 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	f0cd27e027	Refactor ast_timer_ack to return an error and handle the error in timer users Currently, if an acknowledgement of a timer fails Asterisk will not realize that a serious error occurred and will continue attempting to use the timer's file descriptor. This can lead to situations where errors stream to the CLI/log file. This consumes significant resources, masks the actual problem that occurred (whatever caused the timer to fail in the first place), and can leave channels in odd states. This patch propagates the errors in the timing resource modules up through the timer core, and makes users of these timers handle acknowledgement failures. It also adds some defensive coding around the use of timers to prevent using bad file descriptors in off nominal code paths. Note that the patch created by the issue reporter was modified slightly for this commit and backported to 1.8, as it was originally written for Asterisk 10. Review: https://reviewboard.asterisk.org/r/2178/ (issue ASTERISK-20032) Reported by: Jeremiah Gowdy patches: jgowdy-timerfd-6-22-2012.diff uploaded by Jeremiah Gowdy (license 6358) ........ Merged revisions 375893 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 375894 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@375895 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	463cbc55ac	Add safety NULL pointer check in module user references. Made __ast_module_user_remove() check for NULL pointers. ........ Merged revision 375860 from C.3 ........ Merged revisions 375862 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 375863 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@375864 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Jonathan Rose	1d054dbbba	chan_sip: Document a change to user-field encoding introduced with r303509 The change in question was added to improve compliance with RFC3261, but at the time of commit, it wasn't adequately documented in the UPGRADE notes. (closes issue ASTERISK-20561) Reported by: Deniz Review: https://reviewboard.asterisk.org/r/2177/ ........ Merged revisions 375846 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@375847 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	769d9aa259	Don't attempt to purge sessions when no sessions exist Manager's tcp/tls objects have a periodic function that purge old manager sessions periodically. During shutdown, the underlying container holding those sessions can be disposed of and set to NULL before the tcp/tls periodic function is stopped. If the periodic function fires, it will attempt to iterate over a NULL container. This patch checks for whether or not the sessions container exists before attempting to purge sessions out of it. If the sessions container is NULL, we simply return. Note that this error was also caught by the Asterisk Test Suite. ........ Merged revisions 375800 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 375801 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@375802 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	e4afa2c0e2	Only deref a reserved gateway session if we actually reserved one Its perfectly acceptable to have a gateway session unreserved when we go to first allocate one. Unreffing the reserved gateway session - when its NULL - will result in an assertion error. This problem was caught by the Asterisk Test Suite (once we had enough of the debugging flags enabled) ........ Merged revisions 375797 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@375798 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	d38c42aab8	Properly clean up manager resources on exit This patch does two things: 1) It properly unregisters the manager CLI commands 2) It cleans up AMI users on exit. Prior to this patch, the AMI users were not being disposed of properly, resulting in a memory leak. (closes issue ASTERISK-20646) Reported by: Corey Farrell patches: manager_shutdown.patch uploaded by Corey Farrell (license 5909) ........ Merged revisions 375793 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 375794 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@375795 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	260da253be	Properly finalize prepared SQLite3 statements to prevent memory leak The AstDB uses prepared SQLite3 statements to retrieve data from the SQLite3 database. These statements should be finalized during Asterisk shutdown so that the SQLite3 database can be properly closed. Failure to finalize the statements results in a memory leak and a failure when closing the database. This patch fixes those issues by ensuring that all prepared statements are properly finalized at shutdown. (closes issue ASTERISK-20647) Reported by: Corey Farrell patches: astdb-sqlite3_close.patch uploaded by Corey Farrell (license 5909) ........ Merged revisions 375761 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@375763 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	4483067638	Blocked revisions 375759 ........ Fix memory leak when unloading XML documentation This patch is a modified version of a patch originally committed for the Asterisk 11 branch in r375756. A portion of that patch, that fixed the memory leak during unloading XML documentation, applies to branches 1.8 and 10 as well. The patch for this issue was modified for these two branches. (issue ASTERISK-20648) Reported by: Corey Farrell Tested by: mjordan patches: xmldoc-memory_leak.patch uploaded by Corey Farrell (license 5909) ........ Merged revisions 375758 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@375760 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	717068e362	Fix memory leaks in XML documentation This patch fixes two memory leaks: 1) When building XML documentation items, the 'name' attribute was extracted from XML elements but not properly freed after being copied into the item being built. 2) When unloading XML documentation, the doctree container objects were not properly freed. This patch corrects these memory leaks. Note that this patch was modified slightly for this commmit, as the case where the 'name' attribute doesn't exist also wasn't handled in the item construction. This patch also checks for that attribute not existing. (closes issue ASTERISK-20648) Reported by: Corey Farrell Tested by: mjordan patches: xmldoc-memory_leak.patch uploaded by Corey Farrell (license 5909) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@375756 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	69777a9376	Prevent multiple CDR batches from conflicting when scheduling the CDR write The Asterisk Test Suite caught an error condition where a scheduled CDR batch write can be deleted twice if two channels attempt to post their CDRs at the same time. The batch CDR mutex is locked while the CDRs are appended to the current batch list; however, it is unlocked prior to actually scheduling the CDR write. As such, two threads can attempt to remove the currently scheduled batch write at the same time, resulting in an assertion error. This patch extends the time that the mutex is locked to encompass actually scheduling the write. This prevents two threads from unscheduling the currently scheduled write at the same time. ........ Merged revisions 375727 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 375728 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@375729 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Andrew Latham	794eb78090	Doxygen Updates Replace links to missing text files removed in the 1.6.x series with links to the wiki. Doxygen can handle URLs fine, don't atempt to quote them. Also update the wiki link in the Readme to get everyone on the same page. (issue ASTERISK-20259) ........ Merged revisions 375698 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 375699 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@375702 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago

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