4040 Commits (973aeabf2db75e29b11abed188c1db11b110c70d)

Author	SHA1	Message	Date
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
Jonathan Rose	ec237d2e4a	Moves voicemail setup password entry to the end of the setup process. ... This change was made because forcegreeting and forcename settings in voicemail could be circumvented by hanging up after entering a password, because the only way voicemail currently observes whether a mailbox is new or not is by checking to see if the password is the same as the mailbox number or not. (closes issue ASTERISK-18282) Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/1581/ ........ Merged revisions 345062 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 345117 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@345120 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
TransNexus OSP Development	f436a6f27c	Increased max number of destinations. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@345023 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	751488b84c	Fix app_macro.c MODULEINFO section termination. ... (closes issue ASTERISK-18848) Reported by: Tony Mountifield ........ Merged revisions 344557 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@344560 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	46089f6b51	Fix potential deadlock calling ast_call() with channel locks held. ... Fixed app_queue.c:ring_entry() calling ast_call() with the channel locks held. Chan_local attempts to do deadlock avoidance in its ast_call() callback and could deadlock if a channel lock is already held. ........ Merged revisions 344539 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 344540 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@344541 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	464b337b3c	Make AMI event AgentCalled get CallerID/ConnectedLine info from the incoming channel. ... It was strange that the AgentCalled AMI event would get most of its information from the incoming channel but then get the CallerID information from the outgoing channel. Before connected line support was added, this information was always the same at this point. (closes issue ASTERISK-18152) Reported by: Thomas Farnham Tested by: rmudgett ........ Merged revisions 344536 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 344537 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@344538 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Kinsey Moore	dc05ce5e4f	Fix another incorrect case with meetme's PIN logic and add documentation ... This fixes an issue where a user of a dynamic conference was asked for a PIN twice. This also adds documentation to assist in future modifications to the piece of code responsible for PIN checking. (closes issue AST-670) ........ Merged revisions 344439 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 344440 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@344441 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Kinsey Moore	c1647ab33a	Fix pin parameter behavior regression in MeetMe ... The last time this code was touched (by me), a subtlety was missed based on the difference between needing to check a pin's validity and the need to prompt for a pin. (closes issue ASTERISK-18488) ........ Merged revisions 344102 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 344103 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@344104 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Leif Madsen	55ffab4cd9	Add note about how Authenticate() application with option 'd' works. ... (closes issue ASTERISK-17422) Reported by: Leif Madsen ........ Merged revisions 343102 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 343103 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@343104 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Kevin P. Fleming	784bbf70d7	Modify comments in MeetMe application documentation about DAHDI. ... The MeetMe application documentation has some comments about usage of DAHDI, and they were a bit outdated relative to modern DAHDI releases. This patch changes the comment to just tell the user that a functional DAHDI timing source is required, and no longer mention 'dahdi_dummy', since that module does not exist in current DAHDI releases. ........ Merged revisions 342990 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 342991 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@342992 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Terry Wilson	6e730a6806	Use int for storing ao2_container_count instad of size_t ... AST-676 ........ Merged revisions 342435 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 342436 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@342437 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Terry Wilson	f8351a8342	Simplify queue membercount code ... Despite an ominous sounding comment stating that membercount was for "logged in" members only and thus we couldn't use ao2_container_count(), I could not find a single place in the code where that seemed to be accurate. The only time we decremented membercount was when we were marking something dead or actually removing it. The only places we incremented it were either after ao2_link(), or trying to correct for having set it to 0 during a reload. In every case where we were correcting the value, it seemed that we were trying to make the count actually match what ao2_container_count() would return. The only place I could find where we made a determination about something being "logged in" or not, we didn't trust the membercount, but instead looked at devicestate, paused, etc. This patch removes membercount, replaces its use with ao2_container_count, and manually adds the results of ao2_container_count to a "membercount" field for ast_data queue query results. This patch also would fix AST-676, but as it is slightly riskier than the previously committed fix, the two commits have been made separately. Reivew: https://reviewboard.asterisk.org/r/1541/ ........ Merged revisions 342383 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 342384 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@342385 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Terry Wilson	5749ef5be8	Properly update membercount for reloaded members ... Since q->membercount is set to 0 before reloading, it is important to increment it again for reloaded members as well as added. (closes issue AST-676) Review: https://reviewboard.asterisk.org/r/1541/ ........ Merged revisions 342380 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 342381 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@342382 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	3e9f1ee3e0	Fix use of OBJ_KEY in Queue application. ... To use the new OBJ_KEY flag, the container hash and compare callback functions must be updated to support OBJ_KEY. Otherwise, bad things happen. (issue ASTERISK-14769) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@342112 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Gregory Nietsky	7ac53e57b3	queues container needs locking when using the OBJ_NOLOCK flag ... ........ Merged revisions 342017 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@342018 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Gregory Nietsky	3d55a05019	Remove some ref leaks and a return without unlock. ... There some resource leaks introduced in asterisk 10 make sure that locks are not held on return and we release ref's held. ........ Merged revisions 341972 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@341973 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Gregory Nietsky	d36c70e021	Whitespace Fixups / Add Braces ... This janitorial patch is related to work on RB1538 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@341923 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Gregory Nietsky	71b7df16bf	Merged revisions 341580 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/10 ........ r341580 | irroot | 2011-10-20 19:13:23 +0200 (Thu, 20 Oct 2011) | 15 lines Add option to check state when state is unknown r341486 reverts r325483 this is a rework of the patch. optimize to minimize load. add option check_state_unknown to control whether a member with unknown device state is checked there is a small % chance that calls will be sent to the member when they on a call. app_queue will see a device with unknown state as available and does not try verify the state without this option enabled. Review: https://reviewboard.asterisk.org/r/1535/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@341583 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago