asterisk

Commit Graph

Author	SHA1	Message	Date
Russell Bryant	7124314fb7	Make SLA reload more paranoid. Reload support was originally not included for SLA. It was added later, but in a fairly non-traditional way. It basically sets a flag indicating that a reload is pending, and then waits for a time where it thinks everything SLA related is idle and unused, and then executes the reload. It does this because the reload process is destructive. It starts by throwing everything away and starting over. There are a number of problems with this approach. One of them is that the check to see if anything in use was incomplete. This patch makes it more complete and thus less likely for a crash to occur during reload processing. However, this approach still has problems so some much more significant reworking of this code will need to come in as a next step. Patch credit and testing by CoreDial, LLC. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@387688 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Olle Johansson	fa0b6a2fb5	Play periodic prompst for first call in a call queue Review: https://reviewboard.asterisk.org/r/2263/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@386792 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Michael L. Young	7ff1e29675	Fix Manager Segfault When app_queue Is Unloaded When app_queue is unloaded, some manager commands are not being unregistered which result in a segfault. This patch corrects this. (closes issue ASTERISK-21397) Reported by: Peter Katzmann, Corey Farrell Tested by: Corey Farrell Patches: asterisk-21397-missing-unreg-manager-cmd_1.8.diff Michael L. Young (license 5026) asterisk-21397-missing-unreg-manager-cmd_11.diff Michael L. Young (license 5026) Review: https://reviewboard.asterisk.org/r/2444/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@385593 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Michael L. Young	1f2145b7b4	Fix app_voicemail Segfault And A Few Memory Leaks The original report was that app_voicemail would crash. This was caused by ast_config_load() returning CONFIG_STATUS_FILEINVALID but no checks being performed for that return status. After adding the initial patch to fix this issue, Jaco Kroon (jkroon) added some fixes to memory leaks he had discovered. During review, Walter Doekes (wdoekes) suggested adding a helper function in order to determine if we had a valid configuration or not. This patch does the following: * Creates a helper function to check if the configuration is valid * Adds calls to the new helper function where appropiate * Fixes memory leaks where the code returned without running ast_config_destroy() on the configuration that was loaded (closes issue ASTERISK-21302) Reported by: Jaco Kroon Tested by: Jaco Kroon, Michael L. Young Patches: asterisk-11.3.0-app_voicemail-ast_config-fixes.patch Jaco Kroon (license 5671) asterisk-21302-valid_cfg_and_mem_leaks_v3-1.8.diff Michael L. Young (license 5026) Review: https://reviewboard.asterisk.org/r/2443/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@385551 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Jonathan Rose	86f663d7f4	app_voicemail: Add blank argument to externnotify if no context argument At least one call to run_externnotify provides a NULL context parameter and because the snprintf statement doesn't account for a NULL context parameter, it simply writes '(null)' to the arguments string instead. This patch makes it write two quotes back to back for that argument instead in the event of a NULL context. (closes issue ASTERISK-18207) Reported by: Barry L. Kline Patches: modified from patch-20130306 uploaded by Karsten Wemheuer (License 5930) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@384325 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Russell Bryant	37b7df75d4	Fix multi-station answer race condition. When an SLA trunk is ringing (inbound call on the trunk) Asterisk will make outbound calls to the stations that have that trunk. If more than one station answers the call at the same time, all channels other than the first one to answer are left in a bad state. The channel gets leaked, is not connected to anything, and there's no way to get rid of it. We now properly clean up these losing channels by hanging up on them. Since they lost the race, as we process their answer, there is no ringing trunk for them to answer. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@383835 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Kinsey Moore	143ebd5665	Correct app_page documentation The 'A' and 'n' options for Page() mention that the announcement will be played simultaneously. This is not necessarily the case. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@382513 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	fc5e76b96a	Let channels joining a MeetMe conference opt out of the denoiser For some channel drivers, specifically those that have a varying rate in the number of audio samples, the audio quality for a MeetMe conference can be exceedingly poor. This is due to a unilateral application of the DENOISE function in func_speex to channels joining the conference. The denoiser function in the speex library is initialized with the number of audio samples in each sample that will be provided to it. If the number of audio samples changes, the denoiser has to be thrown away and re-initialized. While this could be worked around by removing func_speex, that doesn't help if you actually use the denoiser with other channels on the system. This patches does the following: * Checks for the presence of func_speex as opposed to codec_speex when determining if the DENOISE function is present (which is where the function is actually implemented) * Adds an option to MeetMe 'n' that causes the denoiser to not be applied to a channel when it joins. This keeps the current behavior the default, but let's users disable the denoiser if it causes problems on their system. Review: https://reviewboard.asterisk.org/r/2358 (closes issue AST-1062) Reported by: Thomas Arimont git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@382227 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	726b60ef30	Make ParkAndAnnounce return to priority + 1 when return context is not defined The ParkAndAnnounce application documentation for the optional return_context parameter states the following: return_context The goto-style label to jump the call back into after timeout. Default 'priority+1'. Unfortunately, the application was sending the channel back into the dialplan at 'priority', which is the ParkAndAnnounce application call. This causes an infinite loop of the channel constantly being parked, announced, timed out, parked, announced, timed out... while fun, especially for those callers you wish to drive to the end of madness, this was not the intent of the application. (closes issue ASTERISK-20113) Reported by: serginuez patches: app_parkandannounce.diff uploaded by serginuez (License 6405) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@381916 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	775e80226e	Don't throw a spurious error when using DBdeltree The function call ast_db_deltree returns the number of row deleted, or a negative number if it failed. DBdeltree was treating any non-zero return as an error, causing a spurious verbose error message to be displayed. This patch handles the return code of ast_db_deltree correctly. (closes issue ASTERISK-21070) Reported by: ianc patches: dbdeltree.diff uploaded by ianc (License #5955) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@381364 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Kevin Harwell	42b6f880f4	Properly load say.conf upon reload of module app_playback. If say.conf did not exists prior to originally loading module app_playback it would not load on subsequent reloads of the module once it had been created. This occurred because upon reload of the app_playback module it would only load a new configuration if an old one had previously existed. This fix simply removed the association between checking if an old configuration existed and the loading of the new one. (closes issue ASTERISK-20800) Reported by: pgoergler git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@381216 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Jonathan Rose	79b60d9d4b	app_meetme: Use new prompts for administrator menu The old prompts for the administrator menu were inadequate. They didn't mention that the menu had additional options through the 8 key and pressing the 8 key wouldn't reveal what those options were. This patch fixes all of that while also organizing code pertaining to each individual menu type which was previously all stored in one gigantic function along with many of the basic conference functions. (closes issue AST-996) Reported by: John Bigelow Review: http://reviewboard.digium.internal/r/360/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@379885 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	c5523284d1	Fix station ringback; trunk hangup issues in SLA This patch fixes two bugs: * If an outbound call is made from a SLA phone using SLAStation, then there is no ringtone audible to the phone that originates the call. The indication of the ringing was not being passed to the SLA station; this patch fixes that by passing through the progress indications. * If an SLA station hangs up before the called party answers, then the channel to the called party continues to ring until a timeout occurs. If the called party manages to answer, Asterisk attempts to connect the called party to a non-existant MeetMe room. This patch corrects the behavior by abandoning the call attempt if it detects that the SLA station is no longer in use while attempting to call the called party. Review: https://reviewboard.asterisk.org/r/2275/ (closes issue ASTERISK-20462) Reported by: dkerr patches: asterisk-11-bugid20440+20462.patch uploaded by dkerr (license 5558) asterisk-11-bugid20462.patch uploaded by dkerr (license 5558) (closes issue ASTERISK-20440) Reported by: dkerr patches: asterisk-11-bugid20440.patch uploaded by dkerr (license 5558) asterisk-11-bugid20440+20462.patch uploaded by dkerr (license 5558) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@379825 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	fcb9c5ff50	Fix crash in app_minivm when mime encoding string An incorrect string initializations was left in ast_str_encode_mime from the patch that converted string manipulations to use ast_str strings (r191140). The string initialization causes a crash when ast_str_set is called on the string later on in the function. (closes issue ASTERISK-18697) Reported by: Chris Boot patches: minivm-null-pointer-dereference-fix.patch uploaded by bootc (license 6309) (issue ASTERISK-20854) Reported by: Chris Warr Tested by: Chris Warr git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@379608 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	7ba4fe48ca	app_queue: Fix multiple calls to a queue member that is in only one queue. When ringinuse=no queue members can receive more than one call if these calls happen at nearly the same time. * Fix so a queue member does not receive more than one call from a queue. NOTE: This fix does not prevent multiple calls to a member if the member is in more than one queue. * Did some refactoring to eliminate some code redundancy. (issue ASTERISK-16115) Reported by: nik600 Patches: jira_asterisk_16115_single_q_v1.8.patch (license #5621) patch uploaded by rmudgett Modified git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@378663 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Michael L. Young	afd0961a3c	Fix Queue Log Reporting Every Call COMPLETECALLER With "h" Extension Present When the "h" extension is present within the context of the queue, all calls are being reported COMPLETECALLER even when the agent is hanging up the call. This patch checks to see if the agent hung-up or not instead of only relying on checking if the queue (caller) channel hung-up or not. It would appear that having the h extension in the mix, the pbx goes to the h extension, "hanging-up" the queue channel and triggering the reporting of COMPLETECALLER. (closes issue ASTERISK-20743) Reported by: call Tested by: call, Michael L. Young Patches: asterisk-20743-q-cmplt-caller.diff uploaded by Michael L. Young (license 5026) Review: https://reviewboard.asterisk.org/r/2256/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@378514 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	5996dd6a23	Prevent exhaustion of system resources through exploitation of event cache Asterisk maintains an internal cache for devices in the event subsystem. The device state cache holds the state of each device known to Asterisk, such that consumers of device state information can query for the last known state for a particular device, even if it is not part of an active call. The concept of a device in Asterisk can include entities that do not have a physical representation. One way that this occurred was when anonymous calls are allowed in Asterisk. A device was automatically created and stored in the cache for each anonymous call that occurred; this was possible in the SIP and IAX2 channel drivers and through channel drivers that utilized the res_jabber/res_xmpp resource modules (Gtalk, Jingle, and Motif). These devices are never removed from the system, allowing anonymous calls to potentially exhaust a system's resources. This patch changes the event cache subsystem and device state management to no longer cache devices that are not associated with a physical entity. (issue ASTERISK-20175) Reported by: Russell Bryant, Leif Madsen, Joshua Colp Tested by: kmoore patches: event-cachability-3.diff uploaded by jcolp (license 5000) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@378303 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	c75a2efd1b	app_queue: Revert bad ringinuse=no patch. With the option ringinuse=no set, the patch committed for ASTERISK-16115 causes non-SIP queue members to never be called because the device state is checked after a channel is created to determine if the member is busy. These queue members always get the "Member %s is busy, cannot dial" message. Most channel drivers other than chan_sip use the default device state handling. The default device-state state is considered in use or unknown if the channel exists or not respectively. (closes issue ASTERISK-20801) Reported by: rmudgett Patches: jira_asterisk_16115_revert_r370418_v1.8.patch (license #5621) patch uploaded by rmudgett git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@378036 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	1e8a45c170	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. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@376657 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Jonathan Rose	aaa5514742	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) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@376307 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Rusty Newton	c50c68592a	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) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@376262 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Mark Michelson	e95efa6c50	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/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@375993 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Jonathan Rose	a72df39f77	mixmonitor: Add a test event This test event is being used to fix the mixmonitor_audiohook_inherit test. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@375484 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Matthew Jordan	7a8ba7aa08	Ensure that the Queue application tracks busy members in off nominal situations There are a few code paths where the Queue application fails to count a paused or in use queue member as being 'busy'. This can cause callers to get stuck in the Queue until a paused agent unpauses themselves. (closes issue ASTERISK-20623) Reported by: Bryan Walters patches: app_queue.patch uploaded by Bryan Walters (license 5851) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@375450 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jonathan Rose	40c6c96065	app_queue: Make ordering of rrmemory/rrordered persist over add/remove members Prior to this patch, adding, removing or reloading members to rrmemory would cause the order to become completely jumbled. Now it behaves more or less like rrordered other than the fact that it stores the members on a hash table rather than a linked list. This patch also prevents removal of members and member reloads from jumbling rrordered queues. (issue AST-989) Reported by: Thomas Arimont Review: https://reviewboard.asterisk.org/r/2164/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@375216 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Mark Michelson	30287adef2	Fix some potential misuses of ast_str in the code. Passing an ast_str pointer by value that then calls ast_str_set(), ast_str_set_va(), ast_str_append(), or ast_str_append_va() can result in the pointer originally passed by value being invalidated if the ast_str had to be reallocated. This fixes places in the code that do this. Only the example in ccss.c could result in pointer invalidation though since the other cases use a stack-allocated ast_str and cannot be reallocated. I've also updated the doxygen in strings.h to include notes about potential misuse of the functions mentioned previously. Review: https://reviewboard.asterisk.org/r/2161 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@375025 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	d925564657	app_queue: Made pass connected line updates from the caller to ringing queue members. Party A calls Party B Party B puts Party A on hold. Party B calls a queue. Ringing queue member D sees Party B identification. Party B transfers Party A to the queue. Queue member D does not get a connected line update for Party A. Queue member D answers the call and still sees Party B information. However, if Party A later transfers the call to Party C then queue member D gets a connected line update for Party C. * Made pass connected line updates from the caller to queue members while the queue members are ringing. (closes issue AST-1017) Reported by: Thomas Arimont (closes issue ABE-2886) Reported by: Thomas Arimont Tested by: rmudgett ........ Merged revisions 374801 from https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@374802 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Sean Bright	509c3ebaf7	app_queue: Support persisting and loading of long member lists. Greenlight in #asterisk brought up that he was receiving an error message "Could not create persistent member string, out of space" when running app_queue in Asterisk 10. dump_queue_members() made an assumption that 8K would be enough to store the generated string, but with queues that have large member lists this is not always the case. This patch removes the limitation and uses ast_str instead of a fixed sized buffer. The complicating factor comes from the fact that ast_db_get requires a buffer and buffer size argument, which doesn't let us pull back more than what we pass in, so I introduced a new ast_db_get_allocated() which returns an ast_strdup()'d copy of the value from astdb. As an aside, I did some testing on the maximum size of data that we can store in the BDB library we distribute and was able to store a 10MB string and retrieve it with no problems, so I feel this is a safe patch. Review: https://reviewboard.asterisk.org/r/2136/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@374108 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	727c378636	Fix SendDTMF crash and channel reference leak using channel name parameter. The SendDTMF channel name parameter has two issues. 1) Crashes if the channel name does not exist. 2) Leaks a channel reference if the channel is the current channel. Problem introduced by ASTERISK-15956. * Updated SendDTMF documentation. * Renamed app to senddtmf_name and tweaked the type. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@373945 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	277e9a8358	Fixed meetme tab completion and command documentation. * Removed unnecessary case sensitivity in meetme list, lock, unlock, mute, unmute, and kick commands. * Separated meetme lock/unlock, mute/unmute, and kick commands into their own registered commands to simplify tab completion and parameter checking. meetme_lock_cmd(), meetme_mute_cmd(), and meetme_kick_cmd() * Simplified meetme_show_cmd() (closes issue AST-1006) Reported by: John Bigelow Tested by: rmudgett git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@373815 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Mark Michelson	eb6ad1d9a9	Fix error where improper IMAP greetings would be deleted. (closes issue ASTERISK-20435) Reported by: fhackenberger Patches: asterisk-20435-imap-del-greeting.diff uploaded by Michael L. Young (License #5026) (with suggested modification made by me) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@373735 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Kinsey Moore	bc716c7acd	"show" completion option for "queue" shouldn't appear twice When tab-completing CLI commands starting with "queue", "show" appeared twice in the list due to the way that Asterisk's tab completion functions and the order in which the commands were registered. The registration order has been altered to resolve this issue. (closes issue AST-940) Reported-by: Steve Pitts git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@373666 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jonathan Rose	1c200ea1b2	func_audiohookinherit: Document some missed sources. 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 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@373467 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jonathan Rose	2faa39b458	app_queue: Make queue reload members and variants of that work 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 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@373298 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Joshua Colp	dde552054a	Fix incorrect MeetME conference bridge reference count decrementing and 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 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@373242 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jonathan Rose	ade82766ec	app_meetme: Document that 'p' option will continue in dialplan. (closes issue AST-991) Reported by John Bigelow git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@372765 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	dc28841aad	Fix exception path typo in app_queue.c try_calling(). (closes issue ASTERISK-20380) Reported by: Jeremy Pepper Patches: fix-local-channel-locking.patch (license #6350) patch uploaded by Jeremy Pepper git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@372624 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	4e5b787aa4	Fix VoicemailUserEntry event headers ServerEmail and MailCommand reported 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 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@372620 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Matthew Jordan	2557db0321	Free ast_str objects when temp file fails to be created in MiniVM 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 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@372581 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Matthew Jordan	ca74dccd72	Fix file descriptor leak and pointer scope issue in MiniVM when sending mail 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) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@372554 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Kinsey Moore	3c13a80d5a	Ensure listed queues are not offered for completion 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 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@372517 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Kinsey Moore	daf2b4fb9e	Ensure "rules" is tab-completable for "queue show" 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 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@372444 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Matthew Jordan	1ae95fa41f	Allow configured numbers for FollowMe to be greater than 90 characters 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. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@372390 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Matthew Jordan	e6f3d29864	Fix memory leaks in app_voicemail when using IMAP storage or realtime config 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 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@372268 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Mark Michelson	5d843ed6fc	Prevent crash on shutdown due to refcount error on queues container. 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. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@372089 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Mark Michelson	90d111988e	Help prevent ringing queue members from being rung when ringinuse set to no. 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) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@372048 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jonathan Rose	466e22fafc	app_meetme: Adding test events for following activity in MeetMe. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@371919 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	2ed8fb7637	Fix hangup cause passthrough regression. 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 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@371860 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Mark Michelson	3b476dfd1e	Fix misuses of asprintf throughout the code. 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 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@371590 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Mark Michelson	b0d337d2cb	Fix bug where final queue member would not be removed from memory. 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 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@371306 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago

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4100 Commits (b7da14ecf40da45153e5fe3424fbd593990d8b8d)