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2012-09-24func_audiohookinherit: Document some missed sources.Jonathan Rose
This patch also mentions that AUDIOHOOK_INHERIT can be used to transfer MixMonitor audiohooks. There is also wiki that addresses audiohooks and the use of AUDIOHOOK_INHERIT at the following link: https://wiki.asterisk.org/wiki/display/AST/Audiohooks (closes issue ASTERISK-18220) Reported by: Ishfaq Malik ........ Merged revisions 373467 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 373468 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged revisions 373470 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@373479 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-09-22Doxygen Updates Janitor WorkAndrew Latham
* Whitespace, doc-blocks, spelling, case, missing and incorrect tags. * Add cleanup to Makefile for the Doxygen configuration update * Start updating Doxygen configuration for cleaner output * Enable inclusion of configuration files into documentation * remove mantisworkflow... * update documentation README * Add markup to Tilghman's email and talk with him about updating his email, he knows... * no code changes on this commit other than the mentioned Makefile change (issue ASTERISK-20259) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@373384 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-09-21Doxygen Updates - janitor workAndrew Latham
Doxygen updates including mistakes, misspellings, missing parameters, updates for Doxygen style. Some missing txt file links are removed but their content or essense will be included in some later updates. A majority of the txt files were removed in the 1.6 era but never noted. The HR and EXTREF are simple changes that make the documentation more compatable with more versions of Doxygen. Further updates coming. (issue ASTERISK-20259) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@373330 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-09-21app_queue: Make queue reload members and variants of that workJonathan Rose
Prior to this patch, 'queue reload members' cli command did not work at all. This also affects the manager function 'QueueReload' when supplied with the 'members: yes' field. (closes issue AST-956) Reported by: John Bigelow ........ Merged revisions 373298 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 373300 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged revisions 373318 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@373319 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-09-20Fix incorrect MeetME conference bridge reference count decrementing and ↵Joshua Colp
sometimes premature destruction. When using the 'e' or 'E' option to MeetMe the configured conference bridges are loaded and examined to see if any are empty. If no conference bridges are empty the caller is prompted to enter the number of one. This operation left around a pointer to the last created conference bridge still containing participants. When the caller that was not able to find any empty conference bridge hung up this pointer was disposed of and the reference count of the conference bridge decremented. If there was only a single participant in the conference bridge it was ultimately destroyed prematurely. (closes issue AST-994) Reported by: John Bigelow ........ Merged revisions 373242 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 373245 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged revisions 373246 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@373247 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-09-20Add queue monitoring hintsMatthew Jordan
This patch adds support for hints on a queue. Hints can be added using the nomenclature 'Queue:name', where name is the name of the queue being monitored. This nifty feature was done by Alec Davis. Review: https://reviewboard.asterisk.org/r/1619 Reported by: Alec Davis Tested by: alecdavis patches: review1619.diff2 by alecdavis (license 585) ........ Merged revisions 373235 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@373239 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-09-20Support all ways a member can be available for 'agent available' hintsMatthew Jordan
Alec's patch in r373188 added the ability to subscribe to a hint for when Queue members are available. This patch modifies the check that determines when a Queue member is available by refactoring the availability checks in num_available_members into a shared function is_member_available. This should now handle the ringinuse option, as well as device state values other than AST_DEVICE_NOT_INUSE. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@373222 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-09-20Named call pickup groups. Fixes, missing functionality, and improvements.Richard Mudgett
* ASTERISK-20383 Missing named call pickup group features: CHANNEL(callgroup) - Need CHANNEL(namedcallgroup) CHANNEL(pickupgroup) - Need CHANNEL(namedpickupgroup) Pickup() - Needs to also select from named pickup groups. * ASTERISK-20384 Using the pickupexten, the pickup channel selection could fail even though there was a call it could have picked up. In a call pickup race when there are multiple calls to pickup and two extensions try to pickup a call, it is conceivable that the loser will not pick up any call even though it could have picked up the next oldest matching call. Regression because of the named call pickup group feature. * See ASTERISK-20386 for the implementation improvements. These are the changes in channel.c and channel.h. * Fixed some locking issues in CHANNEL(). (closes issue ASTERISK-20383) Reported by: rmudgett (closes issue ASTERISK-20384) Reported by: rmudgett (closes issue ASTERISK-20386) Reported by: rmudgett Tested by: rmudgett Review: https://reviewboard.asterisk.org/r/2112/ ........ Merged revisions 373220 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@373221 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-09-19app_queue: Support an 'agent available' hintAlec L Davis
Sets INUSE when no free agents, NOT_INUSE when an agent is free. modifes handle_statechange() scan members loop to scan for a free agent and updates the Queue:queuename_avial devstate. Previously exited early if the member was found in the queue. Now Exits later when both a member was found, and a free agent was found. alecdavis (license 585) Reported by: Alec Davis Tested by: alecdavis Review: https://reviewboard.asterisk.org/r/2121/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@373188 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-09-10app_meetme: Document that 'p' option will continue in dialplan.Jonathan Rose
(closes issue AST-991) Reported by John Bigelow ........ Merged revisions 372765 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 372767 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged revisions 372768 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@372769 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-09-07Fix exception path typo in app_queue.c try_calling().Richard Mudgett
(closes issue ASTERISK-20380) Reported by: Jeremy Pepper Patches: fix-local-channel-locking.patch (license #6350) patch uploaded by Jeremy Pepper ........ Merged revisions 372624 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 372625 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged revisions 372626 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@372627 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-09-07Fix VoicemailUserEntry event headers ServerEmail and MailCommand reported ↵Richard Mudgett
values. The AMI action VoicemailUsersList VoicemailUserEntry event headers ServerEmail and MailCommand did not report the global values if they were not overridden. The VoicemailUserEntry event header ServerEmail was not populated with the global value if the voicemail user did not override it. The VoicemailUserEntry event header MailCommand was never populated with a value. * Removed unused struct ast_vm_user member mailcmd[]. (closes issue AST-973) Reported by: John Bigelow Tested by: rmudgett ........ Merged revisions 372620 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 372621 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged revisions 372622 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@372623 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-09-07Free ast_str objects when temp file fails to be created in MiniVMMatthew Jordan
The previous commit (r372554) was from a patch that was written before r366880, which ensured that ast_str objects allocated in the sendmail routine were free'd in off nominal paths. This commit frees the string objects in the off nominal path introduced in r372554. (issue ASTERISK-17133) Reported by: Tzafrir Cohen ........ Merged revisions 372581 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 372582 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged revisions 372583 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@372584 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-09-07Fix file descriptor leak and pointer scope issue in MiniVM when sending mailMatthew Jordan
When MiniVM sends an e-mail and it has the volgain option set, it will spawn sox in a separate process to handle the manipulation of the sound file. In doing so, it creates a temporary file. There are two problems here: 1) The file descriptor returned from mkstemp is leaked 2) The finalfilename character pointer points to a buffer that loses scope once volgain processing is finished. Note that in r316265, Russell fixed some gcc warnings by using the return value of the mkstemp call. A warning was placed in minivm that the file descriptor was going to be leaked. This patch reverts that change, as it handles the leak and 'uses' the file descriptor returned from mkstemp. (closes issue ASTERISK-17133) Reported by: Tzafrir Cohen patches: minivm_18501_demo.diff uploaded by Tzafrir Cohen (license #5035) ........ Merged revisions 372554 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 372555 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged revisions 372556 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@372557 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-09-06Update QueueMemberStatus event documentation to include member status valuesMatthew Jordan
The Status: header in a QueueMemberStatus event (and other QueueMember* events) is the numeric value of the device state corresponding to that Queue Member. As those values are not exactly obvious, listing them in the documentation is useful. Matt Riddell reported this indirectly through the wiki page. (closes issue ASTERISK-20243) Reported by: Matt Riddell ........ Merged revisions 372531 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@372538 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-09-06Ensure listed queues are not offered for completionKinsey Moore
When using tab-completion for the list of queues on "queue reset stats" or "queue reload {all|members|parameters|rules}", the tab-completion listing for further queues erroneously listed queues that had already been added to the list. The tab-completion listing now only displays queues that are not already in the list. (closes issue AST-963) Reported-by: John Bigelow ........ Merged revisions 372517 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 372518 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged revisions 372519 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@372520 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-09-06Ensure "rules" is tab-completable for "queue show"Kinsey Moore
Previously, tabbing at the end of "queue show" produced a list of available queues about which information could be shown, but did not include an alternative command, "rules", to access information about queue rules. The "rules" item should now be shown in the list of tab-completable items. (closes issue AST-958) Reported-by: John Bigelow ........ Merged revisions 372444 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 372445 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged revisions 372446 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@372447 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-09-06Allow configured numbers for FollowMe to be greater than 90 charactersMatthew Jordan
When parsing a 'number' defined in followme.conf, FollowMe previously parsed the number in the configuration file into a buffer with a length of 90 characters. This can artificially limit some parallel dial scenarios. This patch allows for numbers of any length to be defined in the configuration file. Note that Clod Patry originally wrote a patch to fix this problem and received a Ship It! on the JIRA issue. The patch originally expanded the buffer to 256 characters. Instead, the patch being committed duplicates the string in the config file on the stack before parsing it for consumption by the application. (closes issue ASTERISK-16879) Reported by: Clod Patry Tested by: mjordan patches: followme_no_limit.diff uploaded by Clod Patry (license #5138) Slightly modified for this commit. ........ Merged revisions 372390 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 372391 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged revisions 372392 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@372393 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-09-05Add fixes and cleanup to app_alarmreceiver.Mark Michelson
This work comes courtesy of Pedro Kiefer (License #6407) The work was posted to review board by Kaloyan Kovachev (License #5506) (closes issue ASTERISK-16668) Reported by Grant Crawshay (closes issue ASTERISK-16694) Reported by Fred van Lieshout (closes issue ASTERISK-18417) Reported by Kostas Liakakis (closes issue ASTERISK-19435) Reported by Deon George (closes issue ASTERISK-20157) Reported by Pedro Kiefer (closes issue ASTERISK-20158) Reported by Pedro Kiefer (closes issue ASTERISK-20224) Reported by Pedro Kiefer Review: https://reviewboard.asterisk.org/r/2075 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@372310 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-09-05Fix memory leaks in app_voicemail when using IMAP storage or realtime configMatthew Jordan
This patch fixes two memory leaks: 1. When find_user is called with NULL as its first parameter, the voicemail user returned is allocated on the heap. The inboxcount2 function uses find_user in such a fashion when counting new messages, and fails to free the resulting voicemail user object. 2. When populate_defaults is called on a voicemail user, it wipes whatever flags have been set on the object by copying over the global flags object. If the VM_ALLOCED flag was ste on the voicemail user prior to doing so, that flag is removed. This leaks the voicemail user when free_user is later called. (closes issue ASTERISK-19155) Reported by: Filip Jenicek patches: asterisk.patch2 uploaded by Filip Jenicek (license 6277) Patch slightly modified for this commit. Review: https://reviewboard.asterisk.org/r/2096 ........ Merged revisions 372268 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 372288 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged revisions 372289 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@372302 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-09-04app_queue: Only log PAUSEALL/UNPAUSEALL when 1+ memebers changed.Jonathan Rose
Prior to this patch, if pause or unpause was issued on an interface without specifying a specific queue, a PAUSEALL or UNPAUSEALL event would be logged in the queue log even if that interface wasn't a member of any queues. This patch changes it so that these events are only logged when at least one member of any queue exists for that interface. (closes issue AST-946) Reported by: John Bigelow Review: https://reviewboard.asterisk.org/r/2079/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@372148 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-09-04Prevent crash from using app_page with no confbridge.conf file provided.Mark Michelson
Also prevents other potential crashes when using aco API with uninitialized aco_info structs. (closes issue ASTERISK-20305) reported by Noah Engelberth Tested by Noah Engelberth Review: https://reviewboard.asterisk.org/r/2086 ........ Merged revisions 372135 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@372136 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-08-30Prevent crash on shutdown due to refcount error on queues container.Mark Michelson
When app_queue is unloaded, the queues container has its refcount decremented, potentially to 0. Then the taskprocessor responsible for handling device state changes is unreferenced. If the taskprocessor happens to be just about to run its task, then it will create and destroy an iterator on the queues container. This can cause the refcount on the queues container to increase to 1 and then back to 0. Going back to 0 a second time results in double frees. This failure was seen periodically in the testsuite when Asterisk would shut down. ........ Merged revisions 372089 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 372090 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged revisions 372091 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@372092 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-08-30Help prevent ringing queue members from being rung when ringinuse set to no.Mark Michelson
Queue member status would not always get updated properly when the member was called, thus resulting in the member getting multiple calls. With this change, we update the member's status at the time of calling, and we also check to make sure the member is still available to take the call before placing an outbound call. (closes issue ASTERISK-16115) reported by nik600 Patches: app_queue.c-svn-r370418.patch uploaded by Italo Rossi (license #6409) ........ Merged revisions 372048 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 372049 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged revisions 372050 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@372051 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-08-30Clean up doxygen warningsMatthew Jordan
This patch fixes numerous doxygen warnings across Asterisk. It also updates the makefile to regenerate the doxygen configuration on the local system before running doxygen to help prevent warnings/errors on the local system. Much thanks to Andrew for tackling one of the Asterisk janitor projects! (issue ASTERISK-20259) Reported by: Andrew Latham Patches: doxygen_partial.diff uploaded by Andrew Latham (license 5985) make_progdocs.diff uploaded by Andrew Latham (license 5985) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@371989 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-08-29Fix compile errors.Richard Mudgett
........ Merged revisions 371950 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@371951 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-08-29app_meetme: Adding test events for following activity in MeetMe.Jonathan Rose
........ Merged revisions 371919 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 371920 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged revisions 371921 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@371922 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-08-29Fix hangup cause passthrough regression.Richard Mudgett
The v1.8 -r369258 change to fix the F and F(x) action logic introduced a regression in passing the hangup cause from the called channel to the caller channel. (closes issue ASTERISK-20287) Reported by: Konstantin Suvorov Patches: app_dial_hangupcause.patch (license #6421) patch uploaded by Konstantin Suvorov (modified) Tested by: rmudgett ........ Merged revisions 371860 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 371861 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged revisions 371862 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@371863 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-08-21Fix misuses of asprintf throughout the code.Mark Michelson
This fixes three main issues * Change asprintf() uses to ast_asprintf() so that it pairs properly with ast_free() and no longer causes MALLOC_DEBUG to freak out. * When ast_asprintf() fails, set the pointer NULL if it will be referenced later. * Fix some memory leaks that were spotted while taking care of the first two points. (Closes issue ASTERISK-20135) reported by Richard Mudgett Review: https://reviewboard.asterisk.org/r/2071 ........ Merged revisions 371590 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 371591 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged revisions 371592 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@371593 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-08-15Fix bug where final queue member would not be removed from memory.Mark Michelson
If a static queue had realtime members, then there could be a potential for those realtime members not to be properly deleted from memory. If the queue's members were loaded from realtime and then all the members were deleted from the backend, then the queue would still think these members existed. The reason was that there was a short- circuit in code such that if there were no members found in the backend, then the queue would not be updated to reflect this. Note that this only affected static queues with realtime members. Realtime queues with realtime members were unaffected by this issue. (closes issue ASTERISK-19793) reported by Marcus Haas ........ Merged revisions 371306 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 371313 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged revisions 371324 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@371325 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-08-13Add test instrumentationKinsey Moore
This adds test instrumentation for loading and unloading of modules and for certain actions in MeetMe to be used in the testsuite or any other consumer of AMI events. These will only be generated when Asterisk is built with TEST_FRAMEWORK enabled. (issue PQ-1131) (issue PQ-1133) ........ Merged revisions 371201 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 371203 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged revisions 371227 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@371228 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-08-10Fix a couple of documentation problems in app_queue.cMark Michelson
* The RemoveQueueMember app made mention of options that could be passed in, but no options are supported. I have removed the listing of options from the documentation. * The RQMSTATUS variable did not list "NOTDYNAMIC" as a possible value that could be set. (closes issue AST-949) reported by Steve Pitts (closes issue AST-954) reported by Steve Pitts ........ Merged revisions 371141 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 371142 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged revisions 371143 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@371144 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-08-09Correct documentation for the MeetMe x flagKinsey Moore
The documentation for the x flag for MeetMe incorrectly described its function as closing down the conference when the last marked user left. It actually causes the users with that flag to leave the conference when the last marked user exits. The functionality of this flag is not changing. ........ Merged revisions 370985 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 370986 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370987 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-08-08Fix Not Unreferencing A Spied ChannelMichael L. Young
When a channel hangs up while being spied upon and the option to exit the ChanSpy application when the spied on channel hangs up is set, ast_autochan_destroy is not being called and therefore a reference to the spied upon channel is not removed. The symptom being reported was that when using func_group in the dialplan and calling "group show channels" at the cli, the spied upon channel was still being shown while "core show channels" showed that the channel was not up. This patch calls ast_autochan_destroy when a spied upon channel hangs up and the option to exit the ChanSpy application is set, removing the reference to the channel allowing the count for the group that the spied channel was part of to be decremented. (closes issue ASTERISK-17515) Reported by: Arkadiusz Malka Tested by: Alexandr Gordeev, Michael L. Young Patches: asterisk-17515-destroy-autochan.diff uploaded by Michael L. Young (license 5026) ........ Merged revisions 370952 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 370954 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370955 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-08-08Allow support for early media on AMI originates and call files.Mark Michelson
This is based on the work done by Olle Johansson on review board. The idea is that the channel specified in an AMI originate or call file is typically not connected to the outgoing extension until the channel has been answered. With this change, an EarlyMedia header can be specified for AMI originates and an early_media option can be specified in call files. With this option set, once early media is received on a channel, it will be connected with the outgoing extension. (closes issue ASTERISK-18644) Reported by Olle Johansson Review: https://reviewboard.asterisk.org/r/1472 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370951 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-07-31Clean up and ensure proper usage of alloca()Kinsey Moore
This replaces all calls to alloca() with ast_alloca() which calls gcc's __builtin_alloca() to avoid BSD semantics and removes all NULL checks on memory allocated via ast_alloca() and ast_strdupa(). (closes issue ASTERISK-20125) Review: https://reviewboard.asterisk.org/r/2032/ Patch-by: Walter Doekes (wdoekes) ........ Merged revisions 370642 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 370643 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370655 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-07-30DECLINE to load confbridge if the config fails to load.Richard Mudgett
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370589 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-07-30app_meetme: Change app_meetme support level to extended from deprecatedJonathan Rose
(closes issue ASTERISK-20134) Reported by: Leif Madsen ........ Merged revisions 370547 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370548 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-07-21Fix segfault introduced by conversion to ACO APITerry Wilson
The value "none" is specified in the config file as a valid value for the "video_mode" option. The code prior to the ACO conversion did not check for "none", but just ignored it and relied on the default zero value. The parsing with ACO is more strict, so without handling "none" specifically, parsing would fail. When parsing failed, but the module loaded anyway, the config info would never be stored, and one place in the code did not check for this case and would segfault. It was also possible that the aco_info struct's internals would be destroyed and used as well. This patch keeps the module from loading after parse failures, adds the "none" option to "video_mode", registers CLI functions only after parsing has completed, checks the config data for NULL before accessing it, and returns -1 on some allocation failures when initializing. (closes issue ASTERISK-20159) Reported by: Birger "WIMPy" Harzenetter Tested by: Birger "WIMPy" Harzenetter Patches: confbridge_fix3.txt uploaded by Terry Wilson git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370341 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-07-20Clean up ManagerEvent Dial documentationMatthew Jordan
The paragraph describing the SubEvent belongs with the SubEvent parameter itself, and not with its enum values. The order of parsing was placing the description after the last enum, which isn't correct. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370329 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-07-19Convert app_confbridge to use the config options frameworkTerry Wilson
Review: https://reviewboard.asterisk.org/r/2024/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370303 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-07-18Ensure that all ast_datastore_info structures are 'const'.Kevin P. Fleming
While addressing a bug, I came across a instance of 'struct ast_datastore_info' that was not declared 'const'. Since the API already expects them to be 'const', this patch changes the declarations of all existing instances that were not already declared that way. ........ Merged revisions 370183 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 370184 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370187 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-07-10Improve Goto and GotoIf related documentationKinsey Moore
Correct documentation on labeliftrue and labeliffalse parameters of GotoIf() and update several other locations that use the same syntax. (closes issue ASTERISK-20007) Patch-by: Leif Madsen Reported-by: WIMPy ........ Merged revisions 369869 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 369871 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369872 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-07-05AST-2012-011: Resolve heap corruption issue with voicemailKinsey Moore
The heard and deleted arrays in the voicemail state structure were not handled properly following the memory leak fix in r354890 and a fix for an invalid free in r356797. This could result in accessing and writing into freed memory. The allocation for these arrays has been reworked to avoid the possibility of invalid frees, access of freed memory, and crashes that were occurring as a result of this. Locking around accesses and modifications of the voicemail state structure members dh_arraysize, heard, and deleted has been added to prevent simultaneous modification and access when IMAP storage is in use. If IMAP storage is not in use, this locking is not compiled in. Review: https://reviewboard.asterisk.org/r/1994/ (closes issue ASTERISK-19923) Reported by: Dan Delaney Tested by: Dan Delaney, Julian Yap Patches: vm_alloc_fix.diff uploaded by kmoore (license 6273) ........ Merged revisions 369652 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 369653 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369676 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-07-05app_mixmonitor: Fix a reference leak in manager_mixmonitor functionJonathan Rose
Manager_mixmonitor included an early return on failed executions of mixmonitor that would result in a leaked channel reference. (closes issue ASTERISK-19943) Reported by: Mark Murawski Patches: mixmonitor-trunk-368394.patch uploaded by Mark Murawski (license 5791) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369644 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-06-29Hangup handlers - Dialplan subroutines that run when the channel hangs up.Richard Mudgett
Hangup handlers are an alternative to the h extension. They can be used in addition to the h extension. The idea is to attach a Gosub routine to a channel that will execute when the call hangs up. Whereas which h extension gets executed depends on the location of dialplan execution when the call hangs up, hangup handlers are attached to the call channel. You can attach multiple handlers that will execute in the order of most recently added first. (closes issue ASTERISK-19549) Reported by: Mark Murawski Tested by: rmudgett Review: https://reviewboard.asterisk.org/r/2002/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369493 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-06-25Add AMI event documentationMatthew Jordan
This patch adds the core changes necessary to support AMI event documentation in the source files of Asterisk, and adds documentation to those AMI events defined in the core application modules. Event documentation is built from the source by two new python scripts, located in build_tools: get_documentation.py and post_process_documentation.py. The get_documentation.py script mirrors the actions of the existing AWK get_documentation scripts, except that it will scan the entirety of a source file for Asterisk documentation. Upon encountering it, if the documentation happens to be an AMI event, it will attempt to extract information about the event directly from the manager event macro calls that raise the event. The post_process_documentation.py script combines manager event instances that are the same event but documented in multiple source files. It generates the final core-[lang].xml file. As this process can take longer to complete than a typical 'make all', it is only performed if a new make target, 'full', is chosen. Review: https://reviewboard.asterisk.org/r/1967/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369346 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-06-22Explicitly check caller hangup in app Queue rather than a polluted res2 value.Richard Mudgett
........ Merged revisions 369262 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 369263 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369264 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-06-22Fix F and F(x) action logic in Queue application.Richard Mudgett
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369261 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-06-22Check if PBX was started and fix F and F(x) action logic in Dial application.Richard Mudgett
........ Merged revisions 369258 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 369259 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369260 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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