asterisk

Commit Graph

Author	SHA1	Message	Date
Mark Michelson	ec68a15619	Test that shutdown callback is called when expected. git-svn-id: https://origsvn.digium.com/svn/asterisk/team/mmichelson/threadpool@376383 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Mark Michelson	2b36cbe2d5	Change the write-up on taskprocessors to reflect the new design. git-svn-id: https://origsvn.digium.com/svn/asterisk/team/mmichelson/threadpool@376382 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Mark Michelson	12de4198b8	Add a shutdown callback to taskprocessor listeners. This helps account for the fact that it is unknown just how many references may exist for a given taskprocessor listener, so simply unreffing it from the taskprocessor shutdown function is not enough to convey the gravity of the situation. By putting in a shutdown callback, it now becomes clear to the listener not to try to do any further operations on the taskprocessor. git-svn-id: https://origsvn.digium.com/svn/asterisk/team/mmichelson/threadpool@376381 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Automerge script	e8898ec8ba	Merged revisions 376341,376344-376345 via svnmerge from file:///srv/subversion/repos/asterisk/trunk ................ r376341 \| dlee \| 2012-11-15 18:08:00 -0600 (Thu, 15 Nov 2012) \| 34 lines 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 ........ Merged revisions 376339 from http://svn.asterisk.org/svn/asterisk/branches/11 ................ r376344 \| dlee \| 2012-11-15 18:14:00 -0600 (Thu, 15 Nov 2012) \| 1 line Somehow I put in svn-1.6 merge information. Oops. ................ r376345 \| dlee \| 2012-11-15 18:15:30 -0600 (Thu, 15 Nov 2012) \| 15 lines 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 ........ Merged revisions 376343 from http://svn.asterisk.org/svn/asterisk/branches/11 ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/team/mmichelson/threadpool@376352 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Automerge script	4b4be549cb	Merged revisions 376312 via svnmerge from file:///srv/subversion/repos/asterisk/trunk ................ r376312 \| jrose \| 2012-11-15 17:10:13 -0600 (Thu, 15 Nov 2012) \| 23 lines 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 ........ Merged revisions 376310 from http://svn.asterisk.org/svn/asterisk/branches/11 ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/team/mmichelson/threadpool@376320 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Automerge script	748b374ac8	Merged revisions 376291 via svnmerge from file:///srv/subversion/repos/asterisk/trunk ................ r376291 \| beagles \| 2012-11-15 08:35:01 -0600 (Thu, 15 Nov 2012) \| 14 lines 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 376217 from http://svn.asterisk.org/svn/asterisk/branches/11 ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/team/mmichelson/threadpool@376296 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Automerge script	4db8931969	Merged revisions 376282 via svnmerge from file:///srv/subversion/repos/asterisk/trunk ................ r376282 \| newtonr \| 2012-11-14 20:29:40 -0600 (Wed, 14 Nov 2012) \| 16 lines (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 ........ Merged revisions 376264 from http://svn.asterisk.org/svn/asterisk/branches/11 ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/team/mmichelson/threadpool@376287 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Automerge script	c178c4d337	Merged revisions 376235 via svnmerge from file:///srv/subversion/repos/asterisk/trunk ................ r376235 \| rmudgett \| 2012-11-14 13:55:39 -0600 (Wed, 14 Nov 2012) \| 25 lines 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 ........ Merged revisions 376234 from http://svn.asterisk.org/svn/asterisk/branches/11 ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/team/mmichelson/threadpool@376242 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Automerge script	096540c2c0	Merged revisions 376219 via svnmerge from file:///srv/subversion/repos/asterisk/trunk ........ r376219 \| jrose \| 2012-11-13 13:42:13 -0600 (Tue, 13 Nov 2012) \| 12 lines chan_sip: Add SubscribeContext field to SIPshowpeer AMI response The new field is will show up within the response if the requested peer has a subscribe context set. (closes issue ASTERISK-20626) Reported by: Jaco Kroon Patches: asterisk-sip-ami-SubscrContext.patch uploaded by jkroon (license 5671) -with modifications by jrose to conform to style guidelines Review: https://reviewboard.asterisk.org/r/2195/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/team/mmichelson/threadpool@376224 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Automerge script	69ae9ff20d	Merged revisions 376148,376169 via svnmerge from file:///srv/subversion/repos/asterisk/trunk ................ r376148 \| elguero \| 2012-11-12 14:18:47 -0600 (Mon, 12 Nov 2012) \| 26 lines 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 ........ Merged revisions 376144 from http://svn.asterisk.org/svn/asterisk/branches/11 ................ r376169 \| file \| 2012-11-12 14:46:51 -0600 (Mon, 12 Nov 2012) \| 20 lines 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 ........ Merged revisions 376168 from http://svn.asterisk.org/svn/asterisk/branches/11 ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/team/mmichelson/threadpool@376176 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Mark Michelson	426e8d1454	Refine the taskprocessor listener test a bit more. This makes it easier to follow and tests more thoroughly. git-svn-id: https://origsvn.digium.com/svn/asterisk/team/mmichelson/threadpool@376140 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Automerge script	0874e3c825	Merged revisions 376131 via svnmerge from file:///srv/subversion/repos/asterisk/trunk ................ r376131 \| file \| 2012-11-11 11:15:47 -0600 (Sun, 11 Nov 2012) \| 16 lines 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 ........ Merged revisions 376130 from http://svn.asterisk.org/svn/asterisk/branches/11 ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/team/mmichelson/threadpool@376136 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Mark Michelson	7d84963667	Add a test that ensures that taskprocessor listener callbacks are properly called. The new test is a bit raw in that it uses magic numbers, but that can be improved upon easily. git-svn-id: https://origsvn.digium.com/svn/asterisk/team/mmichelson/threadpool@376124 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Mark Michelson	a4a48d9274	Add doxygen and constify some things. git-svn-id: https://origsvn.digium.com/svn/asterisk/team/mmichelson/threadpool@376123 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Mark Michelson	d5716ecae2	Genericize the allocation and destruction of taskprocessor listeners. The goal of this is to take the responsibility away from individual listeners to be sure to properly unref the taskprocessor. git-svn-id: https://origsvn.digium.com/svn/asterisk/team/mmichelson/threadpool@376121 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Mark Michelson	6fc8d830f8	Add a basic taskprocessor test. This just queues a task and ensures that the task gets executed by the default taskprocessor listener. git-svn-id: https://origsvn.digium.com/svn/asterisk/team/mmichelson/threadpool@376120 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Mark Michelson	902a2d8919	Get rid of taskprocessor fields no longer necessary. This includes changing the taskprocessor to use its builtin ao2_lock instead of having a separate mutex. It can do this now since there is no longer an ast_cond_t associated with the taskprocessor. git-svn-id: https://origsvn.digium.com/svn/asterisk/team/mmichelson/threadpool@376119 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Mark Michelson	77725bf293	Move taskprocessors to use a listener model. Taskprocessors are now divided into two units: the task queue and their listeners. When a task is added to the queue, the listener is notified and can take whatever action is desired. This means that taskprocessors are no longer confined to having their tasks executed within a single thread. A default taskprocessor listener has been added that mirrors the old taskprocessor behavior. I've tested it by running Asterisk and placing calls. It appears to work as expected. I'm going to do some cleaning up first and then write some unit tests to be sure everything works as expected. git-svn-id: https://origsvn.digium.com/svn/asterisk/team/mmichelson/threadpool@376118 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Automerge script	7c6c20bfc6	Merged revisions 376092 via svnmerge from file:///srv/subversion/repos/asterisk/trunk ................ r376092 \| mmichelson \| 2012-11-08 16:10:29 -0600 (Thu, 08 Nov 2012) \| 18 lines 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 ........ Merged revisions 376089 from http://svn.asterisk.org/svn/asterisk/branches/11 ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/team/mmichelson/threadpool@376099 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Automerge script	bf369f2018	Merged revisions 376061 via svnmerge from file:///srv/subversion/repos/asterisk/trunk ................ r376061 \| rmudgett \| 2012-11-08 15:12:35 -0600 (Thu, 08 Nov 2012) \| 22 lines 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 ........ Merged revisions 376060 from http://svn.asterisk.org/svn/asterisk/branches/11 ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/team/mmichelson/threadpool@376069 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Automerge script	f69513b85b	Merged revisions 376049 via svnmerge from file:///srv/subversion/repos/asterisk/trunk ................ r376049 \| rmudgett \| 2012-11-08 11:38:31 -0600 (Thu, 08 Nov 2012) \| 41 lines 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 ........ Merged revisions 376048 from http://svn.asterisk.org/svn/asterisk/branches/11 ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/team/mmichelson/threadpool@376054 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Mark Michelson	f28061145f	Get that automerge rollin' git-svn-id: https://origsvn.digium.com/svn/asterisk/team/mmichelson/threadpool@376025 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Mark Michelson	0e9f2aac74	Create a threadpool API for Asterisk. A common theme for a lot of feature development has been that a thread pool would be useful. In this branch, I plan to implement a generic thread pool whose policies can be controlled by listeners who react to activity in the pool. This will more-or-less be a C implementation of the thread pool implemented in Asterisk SCF. The first victim^W entity to use the thread pool will be the PBX. It currently allocates a new thread for each new call and tears it down when the call is completed. Using a thread pool could be useful for systems that have a steady amount of traffic on them. This may, just may, also find use in the new chan_sip that is being developed. git-svn-id: https://origsvn.digium.com/svn/asterisk/team/mmichelson/threadpool@376024 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Mark Michelson	f2bb9afe17	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 ........ Merged revisions 376014 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@376015 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	6ad0126425	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 ........ Merged revisions 375966 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@375967 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Joshua Colp	82dc21e0e1	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. ........ Merged revisions 375925 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@375926 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	a0c363e227	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 ........ Merged revisions 375895 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@375896 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	5336a97f25	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 ........ Merged revisions 375864 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@375865 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Jonathan Rose	ff09fa5ac5	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 ........ Merged revisions 375847 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@375848 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	b6bac916f0	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 ........ Merged revisions 375802 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@375803 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	069f5f8b93	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 ........ Merged revisions 375798 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@375799 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	4bd66cb96b	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 ........ Merged revisions 375795 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@375796 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	a243f4f153	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 ........ Merged revisions 375763 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@375770 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	ff469e9d5d	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) ........ Merged revisions 375756 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@375757 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	19282f682e	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 ........ Merged revisions 375729 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@375730 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Andrew Latham	27a9bab706	Blocked revisions 375702 ........ 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/trunk@375717 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Damien Wedhorn	732767f230	Fix for chan_skinny leaving RTP ports open Skinny wasn't closing RTP sockets. This patch includes ast_rtp_instance_stop before ast_rtp_instance_destroy which fixes the problem. Also add destroy for VRTP (which I believe is unused, but exists). Review: https://reviewboard.asterisk.org/r/2176/ ........ Merged revisions 375660 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@375663 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	f85db0e34d	Things don't need to be that const. ........ Merged revisions 375658 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 375659 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged revisions 375661 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@375662 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	e950086daf	Multiple revisions 375519-375524 ........ r375519 \| rmudgett \| 2012-10-30 16:06:15 -0500 (Tue, 30 Oct 2012) \| 11 lines chan_misdn: Timer primitives must be handled first. The frm->addr is a different "address space" than the stack/instance address of other Lx primitives. The test for B channel instance address could fail. Patches: patch01_timers.diff (license #6372) patch uploaded by Guenther Kelleter JIRA ABE-2888 ........ r375520 \| rmudgett \| 2012-10-30 16:14:58 -0500 (Tue, 30 Oct 2012) \| 10 lines chan_misdn: Free memory in error paths and other memory leaks. The one line commented with BUG is not easily fixable because there is no de-init function one can call. Patches: patch02_memory.diff (license #6372) patch uploaded by Guenther Kelleter JIRA ABE-2888 ........ r375521 \| rmudgett \| 2012-10-30 16:38:41 -0500 (Tue, 30 Oct 2012) \| 14 lines chan_misdn: ISDN NT L2 de-establish/establish * An NT-PTMP cannot de/establish L2 since it doesn't know the TEIs. * On NT-PTP L2 is started when L1 is finally active in handle_l1. * L2 deactivation logging cleanup. * L2 aggregate link status is unknown for NT-PTMP, show as "UNKN". * Removed unused functions and code for L2 handling. Patches: patch03_L2estab.diff (license #6372) patch uploaded by Guenther Kelleter Modified JIRA ABE-2888 ........ r375522 \| rmudgett \| 2012-10-30 16:56:14 -0500 (Tue, 30 Oct 2012) \| 22 lines chan_misdn: Fix broken upper_id/lower_id usage. Sending PH prim via lower_id layer (3 or 1) simply does not work. For TE (3) it returns an error (len=-6) which is not evaluated by handle_l1(), so the L1 layer status ends up wrong. Instead PH must be sent via L4, only then does it reach L1 without an error message. And NT PH prims only reach L1 when they are sent to layer 2 id. --> use upper_id to send PH primitives. * Check for errors in PH_(DE)ACTIVATE \| CONFIRM. * Debug messages are improved. * The lower_id is now not used for anything, except: Why is lower_id layer deleted when it wasn't created? I removed this code since it looks very wrong. Patches: patch04_l1activation.diff (license #6372) patch uploaded by Guenther Kelleter JIRA ABE-2888 ........ r375523 \| rmudgett \| 2012-10-30 17:29:15 -0500 (Tue, 30 Oct 2012) \| 31 lines chan_misdn: Fix loss of B channels if L1 is down. If you make 2 calls out an NT PTMP port which is not connected to any phone, the B channel associated with that call becomes unusable until Asterisk is restarted. The problem is the EVENT_SETUP is queued when L1 is not up in misdn_lib_send_event(). If L1 cannot be activated the event won't be dequeued. It gets even worse when the call is hung up. The queued EVENT_SETUP will be overwritten by an EVENT_DISCONNECT. The reserved B channel then will never be freed. If later someone connects a phone to the port, L1 will eventually activate and the queued EVENT_DISCONNECT is sent down the stack. However, it is ignored because it is the wrong call state. The real fix would be that activation and queueing for a new SETUP is done by the NT stack. But since it doesn't, the workaround must be removed because it doesn't always work. Fix: The event is no longer queued but immediately sent to the stack. If L1 cannot be activated, the L3 state machine that was started by the EVENT_SETUP will do its work, i.e. a timeout will release the B channel properly. The SETUP possibly cannot be sent the first time but is resent by T303 in case L1 could be activated. Patches: patch05_bchan-loss.diff (license #6372) patch uploaded by Guenther Kelleter Modified JIRA ABE-2888 ........ r375524 \| rmudgett \| 2012-10-30 18:26:05 -0500 (Tue, 30 Oct 2012) \| 13 lines chan_misdn: Remove some calls to exit(). Try proper cleanup when something goes wrong in misdn_lib_init(). Especially do not call exit()! * Fix memory leak because stack_destroy() does not free the stack struct. Patches: patch06_cleanup-init.diff (license #6372) patch uploaded by Guenther Kelleter Modified JIRA ABE-2888 ........ Merged revisions 375519-375524 from https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier ........ Merged revisions 375625 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 375626 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged revisions 375627 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@375628 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Michael L. Young	01526b2c3c	Fix Wrong Result In Debug Message For SDP Origin Processing While looking at some debug logs, I noticed that it was being reported that the SDP origin line was unsupported or failed. Upon looking into this on my local machine, I found that I too was getting this debug message yet everything seemed to be getting processed properly. What was discovered is, that, the variable to determine what is displayed in the debug message for the SDP line that was processed, was not being set for the origin line when the result was successful. This patch fixes this and was tested on local machine. ........ Merged revisions 375594 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 375601 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged revisions 375613 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@375614 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Jonathan Rose	d4a357b82f	chan_sip: Fix a bug causing SIP reloads to remove all entries from the registry A regression was introduced in chan_sip by changes to sip reload introduced by r349097. That patch moved peer purging from the beginning of the reload to after the general configuration was finished. This patch fixes that by undoing the repositioning of the original peer purging code and using a similar function after performing general configuration that purges only autocreated peers that were created when persist mode isn't enabled. (closes issue ASTERISK-20611) Reported by: Alisher Review: https://reviewboard.asterisk.org/r/2171/ ........ Merged revisions 375575 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@375576 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Joshua Colp	6de0b18b3b	Fix an issue with res_http_websocket where the chan_sip WebSocket handler could not be registered. On some systems the optional API support uses the GCC compiler attribute "weakref" to provide its functionality. This code changes the function names and prefixes "__" to the front. The res_http_websocket exports file did not take this into account, thereby not allowing those functions to be global and ultimately found. (closes issue ASTERISK-20631) Reported by: danjenkins ........ Merged revisions 375559 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@375560 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	05cee7b717	Properly extract the Body information of an EWS calendar item Unlike all other calendar modules, res_calendar_ews fails to extract the Body information for a calendar item. This is due, in part, to a quirk in the schema in the XML - not only does a CalendarItem contain a Body element, but the CalendarItem exists as a descendant of a different Body element. The neon parser was erroneously skipping all Body elements. This patch fixes that by bypassing Body elements that are not a child of CalendarItem, and parsing the Body element out if it is a child. Note that the original patch by Terry Wilson only needed slight modifications to make it properly pull the Body information out; as such, while I've linked to the patch that I uploaded for Dmitry, I've attributed the patch to Terry. (closes issue ASTERISK-19738) Reported by: Dmitry Burilov Tested by: Dmitry Burilov patches: calendar_ews_body_2012_10_29.diff uploaded by Terry Wilson (license 6283) ........ Merged revisions 375528 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 375531 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged revisions 375532 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@375533 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	9240971cd4	Fix ConfBridge crash if no timing module loaded. (closes issue ASTERISK-19448) Reported by: feyfre Patches: smfix.patch (license #6099) patch uploaded by feyfre Modified for coding guidelines. ........ Merged revisions 375496 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged revisions 375506 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@375511 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Jonathan Rose	1e59b210af	mixmonitor: Add a test event This test event is being used to fix the mixmonitor_audiohook_inherit test. ........ Merged revisions 375484 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 375485 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged revisions 375486 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@375498 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Jonathan Rose	9a80b5da22	confbridge: Fix a bug which made conferences not record with AMI/CLI commands When confbridge was changed to handle conference status with a state machine in r374658. The function responsible for starting recording for a conference was refactored with the function actually responsible for launching the recording thread being split into a function with another name. The old function name was still used for manually started recordings through AMI or CLI. This patch fixes that by switching which function is used to start recording the conference. (closes issue ASTERISK-20601) Reported by: Vilius Patches: confbridge_mixmonitor.diff uploaded by Jonathan Rose (license 6182) ........ Merged revisions 375470 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged revisions 375471 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@375472 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Mark Michelson	5f3f32c494	Prevent resetting of NATted realtime peer address on reload. If a "sip reload" is issued for a SIP peer, then his IP address will be cleared, thus resulting in forgetting the public IP address. Asterisk will then attempt to route SIP traffic to the private IP address. The fix here is to make "sip reload" ignore realtime peers when "host = dynamic" is spotted. Realtime peers can now only have their IP address reset if they have gone from being not dynamic to being dynamic. (closes issue ASTERISK-18203) reported by daren ferreira (closes issue ASTERISK-20572) reported by JoshE Patches: fix_nat_realtime.diff uploaded by JoshE (license #6075) ........ Merged revisions 375415 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 375417 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged revisions 375437 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@375443 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Mark Michelson	da85f8489f	Make evaluation of channel variables consistently case-sensitive. Due to inconsistencies in how variable names were evaluated, the decision was made to make all evaluations case-sensitive. See the UPGRADE.txt file or https://wiki.asterisk.org/wiki/display/AST/Case+Sensitivity for more details. (closes issue ASTERISK-20163) reported by Matt Jordan Review: https://reviewboard.asterisk.org/r/2160 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@375442 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	1eb14dbff8	Ensure that CDRs for a caller in a Queue that is not answered is NO ANSWER. When a caller enters a queue and no queue member answers the call, the current behaviour can be a little odd depending on the paused status of the queue members. If any queue member is paused, but not all, the CDR disposition will be BUSY. If all queue members are paused, then the CDR disposition is based instead on the disposition of the call prior to entering the Queue. This patch modifies the behaviour in the following ways: * If no queue members are paused, the CDR disposition is whatever the disposition was prior to going into Queue. If the call was answered this will be ANSWERED; otherwise, it is NO ANSWER. * If some queue members are pused, the CDR result is NO ANSWER. (This is a change in behaviour, as the result would previously have been BUSY) * If all queue members are paused, the CDR result is whatever the result was prior to going into Queue. This is the same as the behaviour prior to this patch. * If the caller hangs up, times out, or presses '*' with the 'h' option, the CDR disposition is again not set and is dependent on whether or not the caller was Answered prior to entering Queue. This patch was based on one provided by Thomas Arimont, but has been modified to accomodate findings by the reviewers. Review: https://reviewboard.asterisk.org/r/2064/ (closes issue AST-906) Reported by: Thomas Arimont (closes issue ASTERISK-17776) Reported by: Attila Megyeri git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@375416 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	8d65c777c8	Fix the Park 'r' option when a channel parks itself. When a channel uses the Park appliation to park itself with the 'r' option, the channel hears music-on-hold instead of the requested ringing. * Added a missing check for the 'r' option when a channel parks itself. (closes issue ASTERISK-19382) Reported by: James Stocks Patches by: dsessions Review: https://reviewboard.asterisk.org/r/2148/ ........ Merged revisions 375388 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 375389 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged revisions 375390 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@375391 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago

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