asterisk

Commit Graph

Author	SHA1	Message	Date
Kinsey Moore	1fac2fba4b	Deprecated macro usage for connected line, redirecting, and CCSS This commit adds GoSub alternatives to connected line, redirecting, and CCSS macro hooks so that macro can finally be deprecated. This also adds deprecation warnings for those features when used and in documentation. Review: https://reviewboard.asterisk.org/r/1760/ (closes issue SWP-4256) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@357013 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Matthew Jordan	5e40f2cd98	Fix crash in app_voicemail during close_mailbox In r354890, a memory leak in app_voicemail was fixed by properly disposing of the allocated heard/deleted pointers. However, there are situations, particularly when no messages are found in a folder, where these pointers are not allocated and not NULL. In that case, an invalid free would be attempted, which could crash app_voicemail. As there are a number of code paths where this could occur, this patch uses the number of messages detected in the folder before it attempts to free the pointers. This resolves the crash detected in the Asterisk Test Suite's check_voicemail_nominal test. ........ Merged revisions 356797 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 356798 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@356799 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Terry Wilson	ebaf59a656	Opaquification for ast_format structs in struct ast_channel Review: https://reviewboard.asterisk.org/r/1770/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@356573 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Paul Belanger	26865092e6	Multiple revisions 356290,356335,356337 ........ r356290 \| pabelanger \| 2012-02-22 15:20:29 -0500 (Wed, 22 Feb 2012) \| 4 lines Fix -Werror=unused-but-set-variable compiler error (gcc 4.6.2) Review: https://reviewboard.asterisk.org/r/1763/ ........ r356335 \| pabelanger \| 2012-02-22 16:29:25 -0500 (Wed, 22 Feb 2012) \| 2 lines Add back strsep() function for previous commit ........ r356337 \| pabelanger \| 2012-02-22 16:36:37 -0500 (Wed, 22 Feb 2012) \| 2 lines Missed one strsep() function ........ Merged revisions 356290,356335,356337 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 356428 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@356429 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Terry Wilson	57f42bd74f	ast_channel opaquification of pointers and integral types Review: https://reviewboard.asterisk.org/r/1753/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@356042 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Paul Belanger	73b5346e79	Fix channel opaquification for app_rpt git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@355667 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Terry Wilson	34c55e8e7c	Opaquify char * and char[] in ast_channel Review: https://reviewboard.asterisk.org/r/1733/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@354968 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Joshua Colp	fd100261f3	Don't try to play sound files that do not exist. (closes issue ASTERISK-19188) Reported by: slesru ........ Merged revisions 354938 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@354939 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jason Parker	6749b6e2be	Fix a voicemail memory leak with heard/deleted messages. open_mailbox() was changed quite a long time ago to allocate this memory. close_mailbox() should have been changed to be responsible for freeing it. ........ Merged revisions 354889 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 354890 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@354903 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Matthew Jordan	a8cf4dc2b5	Fix IMAP app_voicemail compilation issue introduced in r354429 This simply fixes the compilation issue introduced in r354429 by re-adding the 'quote' variable. (closes issue ASTERISK-19337) Reported by: John Taylor git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@354799 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	5816c60654	Fix crash in ParkAndAnnounce. Well, thats embarrasing. I forgot to initialize the caller_id storage. (closes issue ASTERISK-19311) Reported by: tootai Tested by: rmudgett ........ Merged revisions 354495 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 354496 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@354497 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Walter Doekes	db24fc2523	Avoid cppcheck warnings; removing unused vars and a bit of cleanup. Patch by: Clod Patry Review: https://reviewboard.asterisk.org/r/1651 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@354429 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Joshua Colp	afdd96712c	Make the 'c' option to MeetMe work even if the 'q' option is used. (closes issue ASTERISK-17053) Reported by: justdave git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@354084 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	23bc964e1c	Constify some more channel driver technology callback parameters. Review: https://reviewboard.asterisk.org/r/1707/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@353685 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	2e04182efc	Audit of ao2_iterator_init() usage for v10. Missed one. ........ Merged revisions 353039 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@353040 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	27b69e7d29	Audit of ao2_iterator_init() usage for v1.8. Fixes numerous reference leaks and missing ao2_iterator_destroy() calls as a result. Review: https://reviewboard.asterisk.org/r/1697/ ........ Merged revisions 352955 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 352956 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@352957 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Paul Belanger	5be89b07e2	Fix -Werror=unused-but-set-variable compiler error (gcc 4.6.2) ........ Merged revisions 352643 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 352651 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@352659 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Terry Wilson	99cae5b750	Opaquify channel stringfields Continue channel opaque-ification by wrapping all of the stringfields. Eventually, we will restrict what can actually set these variables, but the purpose for now is to hide the implementation and keep people from adding code that directly accesses the channel structure. Semantic changes will follow afterward. Review: https://reviewboard.asterisk.org/r/1661/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@352348 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jonathan Rose	a1bef6041d	Adds the ability to stop specific mixmonitors by using unique IDs set at monitor launch. MixMonitor receives a new option i(channel_variable) which stores the unique id at said variable. StopMixMonitor now accepts ID as an optional argument, which if included will make StopMixMonitor specifically target the mixmonitor on that particular channel. CLI commands and AMI actions have been ammended to work with the IDs as well. In addition, monitors across a channel can now be listed be listed via CLI command "mixmonitor list <channel>" which will display all of the mixmonitors active on that channel along with the files they each have open. Created by Sergio González Martín. (closes issue ASTERISK-19096) Reported by: Sergio González Martín Review: https://reviewboard.asterisk.org/r/1643/ Review: https://reviewboard.asterisk.org/r/1682/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@352093 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Mark Michelson	b98a25ef93	Prevent potential buffer overflow on AMI MixMonitor command. Don't be alarmed. This only affected trunk, and it would have required manager access to your system. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@351900 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Matthew Jordan	9c4821f468	Realtime queues failed to load queue information without queue member table Previously, realtime queues could be loaded without defining the queue member table. This allowed for queue members to be dynamic, while the realtime queue definitions could exist in some backing storage. Revision 342223 broke this when it changed the return value for realtime_multientry to return NULL when no results are returned. Previously, an empty ast_config object was expected. (closes issue ASTERISK-19170) Reported by: Rene Mendoza Tested by: Rene Mendoza Patches: rt_queue_member_patch.diff uploaded by Matt Jordan (license 6283) ........ Merged revisions 350552 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 350553 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@350554 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Matthew Jordan	a8276fe8ef	Fix crash from bridge channel hangup race condition in ConfBridge This patch addresses two issues in ConfBridge and the channel bridge layer: 1. It fixes a race condition wherein the bridge channel could be hung up 2. It removes the deadlock avoidance from the bridging layer and makes the bridge_pvt an ao2 ref counted object Patch by David Vossel (mjordan was merely the commit monkey) (issue ASTERISK-18988) (closes issue ASTERISK-18885) Reported by: Dmitry Melekhov Tested by: Matt Jordan Patches: chan_bridge_cleanup_v.diff uploaded by David Vossel (license 5628) (closes issue ASTERISK-19100) Reported by: Matt Jordan Tested by: Matt Jordan Review: https://reviewboard.asterisk.org/r/1654/ ........ Merged revisions 350550 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@350551 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	edf466012f	Make FollowMe optionally update connected line information when the accepting endpoint is bridged. Like Dial and Queue, FollowMe needs to deal with AST_CONTROL_CONNECTED_LINE information so when the parties are initially bridged, the connected line information will be correct. * Added the 'I' option just like the app_dial and app_queue 'I' option. * Made 'N' option ignored if the call is already answered. (closes issue ASTERISK-18969) Reported by: rmudgett Tested by: rmudgett Review: https://reviewboard.asterisk.org/r/1656/ ........ Merged revisions 350364 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 350415 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@350416 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Terry Wilson	04da92c379	Replace direct access to channel name with accessor functions There are many benefits to making the ast_channel an opaque handle, from increasing maintainability to presenting ways to kill masquerades. This patch kicks things off by taking things a field at a time, renaming the field to '__do_not_use_${fieldname}' and then writing setters/getters and converting the existing code to using them. When all fields are done, we can move ast_channel to a C file from channel.h and lop off the '__do_not_use_'. This patch sets up main/channel_interal_api.c to be the only file that actually accesses the ast_channel's fields directly. The intent would be for any API functions in channel.c to use the accessor functions. No more monkeying around with channel internals. We should use our own APIs. The interesting changes in this patch are the addition of channel_internal_api.c, the moving of the AST_DATA stuff from channel.c to channel_internal_api.c (note: the AST_DATA stuff will have to be reworked to use accessor functions when ast_channel is really opaque), and some re-working of the way channel iterators/callbacks are handled so as to avoid creating fake ast_channels on the stack to pass in matching data by directly accessing fields (since "name" is a stringfield and the fake channel doesn't init the stringfields, you can't use the ast_channel_name_set() function). I went with ast_channel_name(chan) for a getter, and ast_channel_name_set(chan, name) for a setter. The majority of the grunt-work for this change was done by writing a semantic patch using Coccinelle ( http://coccinelle.lip6.fr/ ). Review: https://reviewboard.asterisk.org/r/1655/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@350223 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Kinsey Moore	c04f4d72fd	Prevent SLA settings from getting wiped out on reload If SLA was reloaded without the config file being changed, current settings got wiped out before the SLA reload code decided it wasn't going to reload the file since nothing was changed. Moving the settings reset later in the reload process fixes this. (closes issue AST-744) ........ Merged revisions 350023 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 350024 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@350025 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	d7005bf8ad	Fix memory leaks in app_followme find_realtime(). (closes issue ASTERISK-19055) Reported by: Matt Jordan ........ Merged revisions 349872 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 349873 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@349874 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Matthew Jordan	baa7f14aab	Fix for ConfBridge config parser unlocking channel mutex too many times When looking up a ConfBridge profile, the config parser would, if it found a channel datastore on the channel requesting the bridge profile, unlock the channel mutex twice. Since that's a little aggressive, it now only unlocks it once. (closes issue ASTERISK-19042) Reported by: Matt Jordan Tested by: Matt Jordan Patches: 19042 uploaded by David Vossel (license 5628) ........ Merged revisions 349619 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@349634 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Matthew Jordan	b0243fb57c	Allow overriding of IMAP server settings on a user by user basis This patch allows the imapserver, imapport, and imapflags settings to be overridden for any voicemail user. It also documents the settings in the sample voicemail.conf file, and updates the voicemail schema to allow storage of those columns. (closes issue ASTERISK-16489) Reporter: Hubert Mickael Tested by: Matt Jordan Review: https://reviewboard.asterisk.org/r/1614/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@349106 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Sean Bright	35a64c2e61	Merged revisions 349045 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/10 ................ r349045 \| seanbright \| 2011-12-23 12:32:33 -0500 (Fri, 23 Dec 2011) \| 25 lines Merged revisions 349044 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r349044 \| seanbright \| 2011-12-23 12:25:01 -0500 (Fri, 23 Dec 2011) \| 18 lines In ChanSpy, don't create audiohooks that will never be used. When ChanSpy is initialized it creates and attaches 3 audiohooks: 1) Read audio off of the channel that we are spying on 2) Write audio to the channel that we are spying on 3) Write audio to the channel that is bridged to the channel that we are spying on. The first is always necessary, but the others are used only when specific options are passed to the ChanSpy application (B, d, w, and W to be specific). When those flags are not passed, neither of those audiohooks are ever sent frames, but we still try to process the hooks for each voice frame that we recieve on the channel. So in short - only create and attach audiohooks that we actually need. ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@349046 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Kinsey Moore	011843e36c	Fix missing doc tags found while fixing ASTERISK-18689 Add missing <variable></variable> tags in app_dial documentation. ........ Merged revisions 348992 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 348993 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@348994 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Matthew Jordan	cf0c9830bf	Add Asterisk TestSuite event hooks to support ConfBridge testing This patch adds initial testsuite event hooks so that ConfBridge tests can be executed in the Asterisk TestSuite. (issue ASTERISK-19059) ........ Merged revisions 348846 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@348848 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jonathan Rose	1b0741c7db	Voicemail with the saycid option will now play a caller's name based on cid if available. In order to check the availability of the caller's name, app_voicemail will check for an audio file in <astspooldir>/recordings/callerids/ This change sets a precedent for where to put recordings of names. Currently the idea is that recordings here could also be used for applications like confbridge and meetme to find recorded names in this folder from callerid (when another recording isn't available) (closes issue ASTERISK-18565) Reporter: Russell Brown Patches: r uploaded by Russel Brown (license 6182) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@348416 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	b05d4603c4	Fix crash during CDR update. The ast_cdr_setcid() and ast_cdr_update() were shown in ASTERISK-18836 to be called by different threads for the same channel. The channel driver thread and the PBX thread running dialplan. * Add lock protection around CDR API calls that access an ast_channel pointer. (closes issue ASTERISK-18836) Reported by: gpluser Review: https://reviewboard.asterisk.org/r/1628/ ........ Merged revisions 348362 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 348363 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@348364 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	8baea2b35e	Fix ParkAndAnnounce to pass the CallerID to the announcing channel. ParkAndAnnounce tried to pass the CallerID to the announcing channel but the ID was wiped out by the channel masquerade done when parking the call. * Save the CallerID before parking the channel to pass it to the announcing channel. * Fixed a minor memory leak in ParkAndAnnounce. * Updated some ParkAndAnnounce log messages. ........ Merged revisions 348310 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 348311 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@348312 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Matthew Jordan	7a3bda0ce3	Added support for all slin formats to app_originate Previously, app_originate could not originate a call into a non-8kHz conference bridge as the formats for non-8kHz slin codecs were not applied to the created channel. This patch adds all of the formats by default, such that if a created channel has a codec that supports a higher sampling rate, a translation path can be built between it and other channels. ........ Merged revisions 348265 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@348266 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Matthew Jordan	aaa715bfae	Fixed Asterisk crash when function QUEUE_MEMBER receives invalid input The function QUEUE_MEMBER has two required parameters (queuename, option). It was only checking for the presence of queuename. The patch checks for the existence of the option parameter and provides better error logging when invalid values are provided for the option parameter as well. ........ Merged revisions 348211 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@348215 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Matthew Jordan	2556729983	Improve error message in CONFBRIDGE_INFO Provided a more descriptive error message when a value supplied for the parameter type is not one of the acceptable values. (closes issue ASTERISK-18717) Reported by: Paul Belanger Patches: __20111103-better-confbridge_info-error-msg.txt (License #4999) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@348160 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	090f9d83a5	Fix FollowMe CallerID on outgoing calls. The addition of the Connected Line support changed how CallerID is passed to outgoing calls. The FollowMe application was not updated to pass CallerID to the outgoing calls. * Fix FollowMe CallerID on outgoing calls. * Restructured findmeexec() to fix several memory leaks and eliminate some duplicated code. * Made check the return value of create_followme_number(). Putting a NULL into the numbers list is bad if create_followme_number() fails. * Fixed a couple uses of ast_strdupa() inside loops. * The changes to bridge_builtin_features.c fix a similar CallerID issue with the bridging API attended and blind transfers. (Not used at this time.) (closes issue ASTERISK-17557) Reported by: hamlet505a Tested by: rmudgett Review: https://reviewboard.asterisk.org/r/1612/ ........ Merged revisions 348101 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 348102 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@348103 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jonathan Rose	e8181c22cd	Adds MixMonitor and StopMixMonitor AMI commands to the manager These commands work much like the dialplan applications that would otherwise invoke them. A nice benefit of these is that they can be invoked on a call remotely and at any time during a call. They work much like the Monitor and StopMonitor ami commands. (closes issue ASTERISK-17726) Reported by: Sergio González Martín Patches: mixmonitor_actions.diff uploaded by Sergio González Martín (license 5644) Review: https://reviewboard.asterisk.org/r/1193/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@347903 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jonathan Rose	518ccb6706	Remove autojump extensions from SayUnixTime, make an option to perform automatic jumps. When a caller sends DTMF while the SayUnixTime application is saying the time, The call would jump to the next extension much like it does during Background(). This patch adds option 'j' to SayUnixTime which when used employs the old behavior. Also, this patch allows arguments to sayunixtime to not be used as empty strings in the case of something like 'sayunixtime(,,,j)' or 'sayunixtime(,,pattern). (closes issue ASTERISK-16675) Reported by: jlpedrosa Patches: patch_SayUnixTime_noJump.patch uploaded by jlpedrosa (license 5959) Review: https://reviewboard.asterisk.org/r/956/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@347866 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jonathan Rose	e1884139c4	Fix regressed behavior of queue set penalty to work without specifying 'in <queuename>' r325483 caused a regression in Asterisk 10+ that would make Asterisk segfault when attempting to set penalty on an interface without specifying a queue in the queue set penalty CLI command. In addition, no attempt would be made whatsoever to perform the penalty setting on all the queues in the core list with either the cli command or the non-segfaulting ami equivalent. This patch fixes that and also makes an attempt to document and rename some functions required by this command to better represent what they actually do. Oh yeah, and the use of this command without specifying a specific queue actually works now. Review: https://reviewboard.asterisk.org/r/1609/ ........ Merged revisions 347656 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@347658 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jonathan Rose	8e94432d9a	Fix: Meetme recording variables from realtime DB use null entries over channel variables Meetme would attempt to substitute the realtime values of RECORDING_FILE and RECORDING_FORMAT from the meetme db entry instead of using the channel variable set for those variables in spite of those database entries being NULL or even lacking a column to represent them. (closes issue ASTERISK-18873) Reported by: Byron Clark Patches: ASTERISK-18873-1.patch uploaded by Byron Clark (license 6157) ........ Merged revisions 347369 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 347383 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@347395 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Walter Doekes	fd64bb66f9	Add VM_INFO() dialplan function to gather information about a mailbox. Deprecates MAILBOX_EXISTS. Provides count, email, exists, fullname, language, locale, pager, password, tz. (closes issue ASTERISK-18634) Patch by: Kris Shaw Review: https://reviewboard.asterisk.org/r/1568 Reviewed by: Walter Doekes git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@347192 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Walter Doekes	7bdaa31d25	Add regression tests for issue ASTERISK-18838. Review: https://reviewboard.asterisk.org/r/1572 Reviewed by: Matt Jordan ........ Merged revisions 347131 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 347146 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@347163 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Walter Doekes	03fd2c0c94	The voicemail [general] zonetag and locale variables weren't loaded until after the mailboxes were initialized. This caused the settings to be unset for those mailboxes until a reload was performed. (closes issue ASTERISK-18838) Review: https://reviewboard.asterisk.org/r/1570 Reviewed by: Matt Jordan ........ Merged revisions 347111 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 347124 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@347157 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Tilghman Lesher	77b670c4ab	Allow each logging destination and console to have its own notion of the verbosity level. Review: https://reviewboard.asterisk.org/r/1599 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@346391 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Paul Belanger	51ce2669af	Add missing sound_only_one config variable (closes issue ASTERISK-18895) Reported by: zvision Patches: conf_config_parser.diff (license #5755) patch uploaded by zvision ........ Merged revisions 345882 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@345883 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Matthew Jordan	279873e8eb	Add admin toggle mute all and participant count menu options to app_confbridge This patch adds two new menu features to app_confbridge, admin_toggle_menu_ participants and participant_count. The admin action will globally mute / unmute all non-admin participants on a converence, while the participant count simply exposes the existing participant count function to the conference bridge menu. This also adds configuration options to change the sound played when the conference is globally muted / unmuted, as well as the necessary config hooks to place these functions in the DTMF menus. (closes issue ASTERISK-18204) Reported by: Kevin Reeves Tested by: Matt Jordan Patches: app_confbridge.c.patch.txt, conf_config_parser.c.patch.txt, confbridge.h.patch.txt uploaded by Kevin Reeves (license 6281) Review: https://reviewboard.asterisk.org/r/1518/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@345560 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jonathan Rose	2d67b1b378	Guarantee messages go into the right folders with multiple recipients Before, using the U flag in Voicemail with multiple recipients would put urgent messages in the INBOX folder for all users past the first thanks to a bug with the message copying function. This would also cause messages to fail to be sent if the INBOX directory hadn't been created for that mailbox yet. (closes issue ASTERISK-18245) Reported by: Matt Jordan (closes issue ASTERISK-18246) Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/1589/ ........ Merged revisions 345487 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 345488 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@345489 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	9e726d9cb4	Make queue log indicate if ADDMEMBER is paused for AMI and realtime. * Add parameter to queue log ADDMEMBER to indicate if the member is paused. (closes issue ASTERISK-18645) Reported by: garlew Patches: paused.diff (License #5337) patch uploaded by garlew Tested by: rmudgett, garlew Review: https://reviewboard.asterisk.org/r/1469/ ........ Merged revisions 345285 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 345290 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@345317 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago

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4057 Commits (890717f3052c9715e1ac6be08d9e5d77b55892be)