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2014-06-13stasis: Reduce creation of channel snapshots to improve performanceMatthew Jordan
During some performance testing of Asterisk with AGI, ARI, and lots of Local channels, we noticed that there's quite a hit in performance during channel creation and releasing to the dialplan (ARI continue). After investigating the performance spike that occurs during channel creation, we discovered that we create a lot of channel snapshots that are technically unnecessary. This includes creating snapshots during: * AGI execution * Returning objects for ARI commands * During some Local channel operations * During some dialling operations * During variable setting * During some bridging operations And more. This patch does the following: - It removes a number of fields from channel snapshots. These fields were rarely used, were expensive to have on the snapshot, and hurt performance. This included formats, translation paths, Log Call ID, callgroup, pickup group, and all channel variables. As a result, AMI Status, "core show channel", "core show channelvar", and "pjsip show channel" were modified to either hit the live channel or not show certain pieces of data. While this is unfortunate, the performance gain from this patch is worth the loss in behaviour. - It adds a mechanism to publish a cached snapshot + blob. A large number of publications were changed to use this, including: - During Dial begin - During Variable assignment (if no AMI variables are emitted - if AMI variables are set, we have to make snapshots when a variable is changed) - During channel pickup - When a channel is put on hold/unhold - When a DTMF digit is begun/ended - When creating a bridge snapshot - When an AOC event is raised - During Local channel optimization/Local bridging - When endpoint snapshots are generated - All AGI events - All ARI responses that return a channel - Events in the AgentPool, MeetMe, and some in Queue - Additionally, some extraneous channel snapshots were being made that were unnecessary. These were removed. - The result of ast_hashtab_hash_string is now cached in stasis_cache. This reduces a large number of calls to ast_hashtab_hash_string, which reduced the amount of time spent in this function in gprof by around 50%. #ASTERISK-23811 #close Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/3568/ ........ Merged revisions 416211 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@416216 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-06-12app_queue: delayed state can cause early leavewhenempty ringingScott Griepentrog
In app_queue, device state changes arrive in event messages and update the queue member status value. That value is checked in get_member_status() to decide that the caller should leave when there are no available members. Although event messages can be delayed by other activity, there is no adverse affect by lagged status except in one specific case: there is only one available member, it was just rung, and leavewhenempty is enabled set for ringing members. This change adds a direct check of the device state only under this condition where the caller may be dropped incorrectly, resolving this issue without affecting performance of app_queue normally. AST-1248 #close Review: https://reviewboard.asterisk.org/r/3595/ Reported by: Thomas Arimont ........ Merged revisions 415833 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 415835 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 415836 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@415839 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-06-12MixMontior: Add class authorization requirements to MixMonitor AMI commandsJonathan Rose
MixMonitor AMI commands StartMixMonitor and StopMixMonitor lacked class authorization. StopMixMonitor now requires that the manager user either have the call or system class authorization. StartMixMonitor is a slightly larger issue since it can execute shell commands if the right arguments are passed into it, and we consider this a permission escalation. A security release will be issued for problem this shortly. ASTERISK-23609 #close Reported by: Corey Farrell ........ Merged revisions 415825 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 415832 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@415834 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-06-05app_confbridge: Allow muting of users waiting to enter a ConfBridgeMatthew Jordan
Prior to this patch, users waiting to enter a ConfBridge were not considered when muted via the CLI or via AMI. Instead, a confusing message would be emitted stating that the channel did not exist. This patch allows a user to be muted when waiting to enter a ConfBridge conference. This is equivalent to start when muted, only toggled via the CLI or AMI. Review: https://reviewboard.asterisk.org/r/3582 #ASTERISK-23824 #close patches: rb3582.patch uploaded by tm1000 (License 6524) ........ Merged revisions 415206 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 415207 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@415208 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-06-04app_confbridge: Correct verification of conference name lengthCorey Farrell
Conference names were not checked for maximum length, allowing unexpected behaviour. This change adds checking to ensure the maximum length is not exceeded. The maximum length is also changed from 32 to AST_MAX_EXTENSION. ASTERISK-23035 #close Reported by: Iñaki Cívico Tested by: Iñaki Cívico Patches: confbridge-enforce_max-1.8.patch uploaded by coreyfarrell (license 5909) confbridge-enforce_max-11up.patch uploaded by coreyfarrell (license 5909) ........ Merged revisions 415060 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 415066 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 415078 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@415080 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-05-28Logger/CLI/etc.: Fix some aesthetic issues; reduce chatty verbose messagesMatthew Jordan
This patch addresses some aesthetic issues in Asterisk. These are all just minor tweaks to improve the look of the CLI when used in a variety of settings. Specifically: * A number of chatty verbose messages were removed or demoted to DEBUG messages. Verbose messages with a verbosity level of 5 or higher were - if kept as verbose messages - demoted to level 4. Several messages that were emitted at verbose level 3 were demoted to 4, as announcement of dialplan applications being executed occur at level 3 (and so the effects of those applications should generally be less). * Some verbose messages that only appear when their respective 'debug' options are enabled were bumped up to always be displayed. * Prefix/timestamping of verbose messages were moved to the verboser handlers. This was done to prevent duplication of prefixes when the timestamp option (-T) is used with the CLI. * Verbose magic is removed from messages before being emitted to non-verboser handlers. This prevents the magic in multi-line verbose messages (such as SIP debug traces or the output of DumpChan) from being written to files. * _Slightly_ better support for the "light background" option (-W) was added. This includes using ast_term_quit in the output of XML documentation help, as well as changing the "Asterisk Ready" prompt to bright green on the default background (which stands a better chance of being displayed properly than bright white). Review: https://reviewboard.asterisk.org/r/3547/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@414798 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-05-28app_agent_pool: Return to dialplan if the agent fails to ack the call.Richard Mudgett
Improvements to the agent pool functionality. * AgentRequest no longer hangs up the caller if the agent fails to connect with the caller. It now continues in the dialplan. * AgentRequest returns AGENT_STATUS set to NOT_CONNECTED if the agent failed to connect with the call. Most likely because the agent did not acknowledge the call in time or got disconnected. * The agent alerting play file configured by the agent.conf custom_beep option can now be disabled by setting the option to an empty string. The agent is effectively alerted to a call presence when MOH stops. * Fixed bridge reference leak when the agent connects with a caller. ASTERISK-23499 #close Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/3551/ ........ Merged revisions 414747 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@414748 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-05-22app_meetme: Don't interrupt MOH for waitmarked users.Richard Mudgett
Occasionally, when the last marked user leaves the conference, waitmarked users don't get MOH if MOH is supposed to be played while a waitmarked user is waiting for another marked user. * Made not interrupt MOH when the user is a waitmarked user. The waitmarked user doesn't need to hear any leave announcements from the conference as the user would have already heard different leave announcements if they were enabled. Apparently DAHDI occasionally sends unending non-silent streams to these users or a normal user still in the conference has continuous high background noise. These non-silent streams cause MOH to be suspended while the never ending "announcement" is played. Issue caused by ASTERISK-13680. AST-1349 #close Reported by: Tyler Stewart Review: https://reviewboard.asterisk.org/r/3543/ ........ Merged revisions 414401 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 414402 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 414404 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@414417 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-05-22ARI: Add ability to raise arbitrary User EventsScott Griepentrog
User events can now be generated from ARI. Events can be signalled with arbitrary json variables, and include one or more of channel, bridge, or endpoint snapshots. An application must be specified which will receive the event message (other applications can subscribe to it). The message will also be delivered via AMI provided a channel is attached. Dialplan generated user event messages are still transmitted via the channel, and will only be received by a stasis application they are attached to or if the channel is subscribed to. This change also introduces the multi object blob mechanism used to send multiple snapshot types in a single message. The dialplan app UserEvent was also changed to use multi object blob, and a new stasis message type created to handle them. ASTERISK-22697 #close Review: https://reviewboard.asterisk.org/r/3494/ ........ Merged revisions 414405 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@414406 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-05-15app_meetme: Fix overwrite of DAHDI conference data structure.Richard Mudgett
Starting a conference recording using the admin menu overwrites the DAHDI conference data structure used to modify the admin user's conference mute mode. * Made no longer pass the user's DAHDI conference data structure into the menu functions. The menu now uses its own DAHDI conference data structure to start the recording channel. * Moved the unlock conf->playlock to before playing the conf-full message. No sense keeping the lock while that prompt is playing. The user is never going to get into the conference at that point. ........ Merged revisions 413991 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 413992 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 413993 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@413994 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-05-12app_chanspy: Fix a test that was failing on account of r413551Jonathan Rose
ASTERISK-23381 #close ASTERISK-23381 #comment Reported by: Robert Moss Review: https://reviewboard.asterisk.org/r/3505/ ........ Merged revisions 413710 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 413712 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@413713 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-05-09Allow Asterisk to compile under GCC 4.10Kinsey Moore
This resolves a large number of compiler warnings from GCC 4.10 which cause the build to fail under dev mode. The vast majority are signed/unsigned mismatches in printf-style format strings. ........ Merged revisions 413586 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 413587 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 413588 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@413589 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-05-09app_chanspy: Fix a bug where Barge mode could failJonathan Rose
If the barge audiohook was attached prior to the spyee and its peer actually being bridged, the audiohook would not be applied and the connected peer would not be able to hear audio from the spy when the spy is in barge mode. (closes issue ASTERISK-23381) Reported by: Robert Moss Review: https://reviewboard.asterisk.org/r/3505/ ........ Merged revisions 413551 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 413556 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@413557 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-05-08app_queue: Extend documentation for various Manager actions and events.Joshua Colp
........ Merged revisions 413485 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 413486 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 413487 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@413488 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-05-07app_confbridge: Fixed "CBAnn" channels not going away.Richard Mudgett
Fixed a ref leak in conf_handle_talker_cb() everytime the conference bridge was found to report a channel's talker status change. The resulting leak caused the "CBAnn" channels and the conference bridge to never be destroyed. Thanks to Richard Kenner on the asterisk-user's list for locating the problem. Reported by: Richard Kenner ........ Merged revisions 413454 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@413455 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-05-07app_confbridge: Fix ref leak in CLI "confbridge kick" command.Richard Mudgett
Fixed ref leak in the CLI "confbridge kick" command when the channel to be kicked was not in the conference. ........ Merged revisions 413451 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 413452 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@413453 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-04-21Confbridge: Fix ConfbridgeKick AMI documentationKinsey Moore
This adds documentation for the "all" channel option for the ConfbridgeKick AMI action and adjusts AMI responses accordingly. (issue ASTERISK-23282) Reported by: Dorian Logan ........ Merged revisions 412730 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@412731 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-04-21Confbridge: Add references for kick all optionKinsey Moore
After the ability to kick all attendees from a conference was added, a rework removed the comment about that feature from the CLI documentation. This adds that documentation and adds "all" to the participant tab completion list for the confbridge kick command. (closes issue ASTERISK-23282) Reported by: Dorian Logan ........ Merged revisions 412728 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@412729 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-04-19app_sms: Fix uninitialized values; hangup channel when REL is sent successfullyMatthew Jordan
This patch fixes two issues in app_sms: (1) Firstly, the 'flags' field on the stack in sms_exec() is uninitialised, causing it to use the wrong protocol in some cases. This patch correctly initializes the flags fields. (2) Secondly, when disconnect supervision is not working or inbanddisconnect=yes is set in chan_dahdi.conf, app_sms was failing to terminate the call after it sent the REL(ease) message and the peer stopped talking to it. This patch fixes the code to handle the 'bad stop bit' message more gracefully in that case, and hang up the call. Review: https://reviewboard.asterisk.org/r/1392/ ASTERISK-18331 #close Reported by: David Woodhouse patches: asterisk-fix-sms.patch uploaded by David Woodhouse (License 5754) ........ Merged revisions 412655 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 412656 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 412657 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@412697 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-04-18Originated calls: Fix several originate call problems.Richard Mudgett
* Restore the reason value set by pbx_outgoing_attempt() to use AST_CONTROL_xxx values as all the consumers were expecting rather than cause codes. * Fixed the dial routines to set cause codes for more than just ast_request() so pbx_outgoing_attempt() reason codes will function. * Fix inconsistent locked_channel return status in pbx_outgoing_attempt(). The chanel may not have been locked or the channel may have been a stale pointer. * Fixed the OutgoingSpoolFailed channel to run dialplan whenever the dialing fails for an originate exten and 1 < synchronous. * Fix incorrect ast_cond_wait() usage in pbx_outgoing_attempt(). Indroduced by issue ASTERISK-22212 patch. * Made struct pbx_outgoing use the ao2 lock instead of its own lock for the cond wait mutex. No sense in having two locks associated with the same struct when only one is needed. Review: https://reviewboard.asterisk.org/r/3421/ ........ Merged revisions 412581 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@412583 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-04-18app_dial and app_queue: Make lock the forwarding channel while taking the ↵Richard Mudgett
channel snapshot. * Fixed ast_channel_publish_dial_forward() not locking the forwarded channel when taking the channel snapshot. * Fixed app_dial.c:do_forward() using the wrong channel to get the original call forwarding string. * Removed unnecessary locking when calling ast_channel_publish_dial() and ast_channel_publish_dial_forward() in app_dial and app_queue. Holding channel locks when calling ast_channel_publish_dial_forward() with a forwarded channel could result in pausing the system while the stasis bus completes processsing a forwarded channel subscription. Review: https://reviewboard.asterisk.org/r/3451/ ........ Merged revisions 412579 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@412580 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-04-15(mix)monitor: Add options to enable a periodic beepRussell Bryant
Add an option to enable a periodic beep to be played into a call if it is being recorded. If enabled, it uses the PERIODIC_HOOK() function internally to play the 'beep' prompt into the call at a specified interval. This option is provided for both Monitor() and MixMonitor(). Review: https://reviewboard.asterisk.org/r/3424/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@412427 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-04-11app_stack: Add missing unlock in off-nominal path of STACK_PEEK function.Richard Mudgett
ASTERISK-23620 #close Reported by: Bradley Watkins Patches: ASTERISK-23620_unlock_oldlist.patch (license #5021) patch uploaded by Bradley Watkins ........ Merged revisions 412225 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 412226 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 412227 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@412228 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-04-11bridging: Ensure locking during snapshot creationKinsey Moore
While the vast majority of bridge snapshot creation is locked properly, there are currently some instances that are not. This adds the missing locking to ensure bridge state is not malleable during snapshot creation. (closes issue ASTERISK-22904) Review: https://reviewboard.asterisk.org/r/3415/ Reported by: Matt Jordan ........ Merged revisions 412193 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@412194 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-04-09Add a Command header to the AMI Mixmonitor action.Mark Michelson
This fixes a parsing error that occurred during the processing of the AMI action. The error did not result in MixMonitor itself misbehaving, but it could result in the AMI response not giving correct information back. The new header allows for one to specify a post-process command to run when recording finishes. Previously, in order to do this, the post-process command would have to be placed at the end of the Options: header. Patches: mixmonitor_command_2.patch by jhardin (License #6512) ........ Merged revisions 412048 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@412050 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-04-08app_confbridge: Fix confbridge.conf dsp_talking_threshold option setting ↵Richard Mudgett
wrong parameter. Fixed copy pasta error. ASTERISK-23545 #close Reported by: John Knott ........ Merged revisions 411944 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 411945 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@411946 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-04-07app_queue: Re-add HoldTime to QueueCallerAbandon event (simple typo during ↵Walter Doekes
ast12 refactor). Reported by: Ibrahim22 (on IRC) Tested by: Ibrahim22 ........ Merged revisions 411811 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@411812 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-04-01app_queue: Fix a bug where realtime members would be deleted during reload ↵Joshua Colp
causing waiting callers to get ejected. This patch causes realtime queue members to remain in queues during the reload process. Previously these members would be removed causing any waiting callers to be ejected from the queue with a reason of "EXITEMPTY". ASTERISK-23547 #close ASTERISK-23547 #comment Patch app_queue_fix_realtime_reload_1.8_trunk.patch submitted by Italo Rossi (license 6409) Review: https://reviewboard.asterisk.org/r/3404/ ........ Merged revisions 411584 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 411585 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 411586 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@411587 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-03-27Fix dialplan function NULL channel safety issuesCorey Farrell
(closes issue ASTERISK-23391) Reported by: Corey Farrell Review: https://reviewboard.asterisk.org/r/3386/ ........ Merged revisions 411313 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 411314 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 411315 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@411328 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-03-20app_confbridge: Fix bug - users with startmuted set don't start mutedJonathan Rose
(closes issue ASTERISK-23461) Reported by: Chico Manobela Review: https://reviewboard.asterisk.org/r/3373/ ........ Merged revisions 410965 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 410966 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@410967 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-03-17app_confbridge: Add missing destructor call to announcer channel destructor.Richard Mudgett
........ Merged revisions 410671 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@410672 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-03-12app_confbridge: Make explicitly stop MOH if a user is kicked or hangs up ↵Richard Mudgett
while MOH is playing. When MOH is playing to a user in a conference and the user is kicked or hangs up from the conference then the AMI MusicOnHoldStop events didn't happen. (Asterisk v11 AMI event: MusicOnHold, state:Stop) (closes issue ASTERISK-23311) Reported by: Benjamin Keith Ford Review: https://reviewboard.asterisk.org/r/3306/ ........ Merged revisions 410490 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 410491 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@410492 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-03-07uniqueid: channel linkedid, ami, ari object creation with id'sScott Griepentrog
Much needed was a way to assign id to objects on creation, and much change was necessary to accomplish it. Channel uniqueids and linkedids are split into separate string and creation time components without breaking linkedid propgation. This allowed the uniqueid to be specified by the user interface - and those values are now carried through to channel creation, adding the assignedids value to every function in the chain including the channel drivers. For local channels, the second channel can be specified or left to default to a ;2 suffix of first. In ARI, bridge, playback, and snoop objects can also be created with a specified uniqueid. Along the way, the args order to allocating channels was fixed in chan_mgcp and chan_gtalk, and linkedid is no longer lost as masquerade occurs. (closes issue ASTERISK-23120) Review: https://reviewboard.asterisk.org/r/3191/ ........ Merged revisions 410157 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@410158 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-03-06app_confbridge: Remove some noop code.Richard Mudgett
........ Merged revisions 409976 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@409977 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-02-28app_queue: Fix documented AMI event nameKinsey Moore
During the rewrite of AMI events to use the Stasis bus, the name of the QueueMemberPaused event was changed to QueueMemberPause. This corrects documentation to reflect that. ........ Merged revisions 409234 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@409237 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-02-21json: Fix off-nominal json ref counting issues.Richard Mudgett
* Fixed off-nominal json ref counting issue with using the following API calls: ast_json_object_set() and ast_json_array_append(). * Fixed off-nominal error reporting in ast_ari_endpoints_list(). * Fixed some miscellaneous off-nominal json ref counting issues in report_receive_fax_status() and dial_to_json(). ........ Merged revisions 408713 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@408714 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-02-21app_chanspy: Documentation Update To Clarify "x" OptionMichael L. Young
When using the "x" option (specify a DTMF digit to exit the application), it is not obvious in the documentation that this only works when spying on a channel. If a channel being used to spy on other channels is waiting to connect to a channel or is no longer attached to a channel, the DTMF is ignored. As noted on the issue tracker, since there are workarounds available and this is a rarely used option we are opting for a documentation change here. (closes issue ASTERISK-22661) Reported by: Chris Hillman Patches: asterisk-22661-doc-clarify-chan_spy.diff uploaded by Michael L. Young (license 5026) Review: https://reviewboard.asterisk.org/r/2990/ ........ Merged revisions 408536 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 408537 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 408538 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@408539 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-02-20apps/app_queue - Fix incorrect Macro parameter documentationRusty Newton
Macro is executed on the called channel, not the calling channel. (closes issue ASTERISK-23069) Reported By: Bryan Anderson ........ Merged revisions 408447 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 408448 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 408449 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@408450 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-02-10ConfBridge: Correct prompt playback targetKinsey Moore
Currently, when the first marked user enters the conference that contains waitmarked users, a prompt is played indicating that the user is being placed into the conference. Unfortunately, this prompt is played to the marked user and not the waitmarked users which is not very helpful. This patch changes that behavior to play a prompt stating "The conference will now begin" to the entire conference after adding and unmuting the waitmarked users since the design of confbridge is not conducive to playing a prompt to a subset of users in a conference in an asynchronous manner. (closes issue PQ-1396) Review: https://reviewboard.asterisk.org/r/3155/ Reported by: Steve Pitts ........ Merged revisions 407857 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 407858 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@407859 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-02-01app_stack: protect against missing parameters to STACK_PEEK and LOCAL_PEEKCorey Farrell
STACK_PEEK requires 2 parameters and LOCAL_PEEK requires 1 parameter. This protects against situations where those parameters are blank or missing by logging an error and returning. (closes issue ASTERISK-23220) Reported by: James Sharp ........ Merged revisions 407100 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 407103 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 407104 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@407105 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-01-31CDRs: fix a variety of dial status problems, h/hangup handler creating CDRsMatthew Jordan
This patch fixes a number of small-ish problems that were noticed when witnessing the records that the FreePBX dialplan produces: (1) Mid-call events (as well as privacy options) have the ability to change the overall state of the Dial operation after the called party answers. This means that publishing the DialEnd event when the called party is premature; we have to wait for the execution of these subroutines to complete before we can signal the overall status of the DialEnd. This patch moves that publication and adds handlers for the mid-call events. (2) The AST_FLAG_OUTGOING channel flag is cleared if an after bridge goto datastore is detected. This flag was preventing CDRs from being recorded for all outbound channels that had a 'continue' option enabled on them by the Dial application. (3) The CDR engine now locks the 'Dial' application as being the CDR application if it detects that the current CDR has entered that app. This is similar to the logic that is done for Parking. In general, if we entered into Dial, then we want that CDR to record the application as such - this prevents pre-dial handlers, mid-call handlers, and other shenaniganry from changing the application value. (4) The CDR engine now checks for the AST_SOFTHANGUP_HANGUP_EXEC in more places to determine if the channel is in hangup logic or dead. In either case, we don't want to record changes in the channel. (5) The default option for "endbeforehexten" has been changed to "yes". In general, you don't want to see CDRs in the 'h' exten or in hangup logic. Since the semantics of that option changed in 12, it made sense to update the default value as well. (6) Finally, because we now have the ability to synchronize on the messages published to the CDR topic, on shutdown the CDR engine will now synchronize to the messages currently in flight. This helps to ensure that all in-flight CDRs are written before shutting down. (closes issue ASTERISK-23164) Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/3154 ........ Merged revisions 407084 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@407085 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-01-31app_dial: Allow macro/gosub pre-bridge execution to occur on prioritiesMatthew Jordan
The parsing for the destination of the macro/gosub uses the '^' character to separate out context, extension, and priority. However, the logic for the macro/gosub execution was written such that it would only do the actual macro/gosub jump if a '^' character existed. This doesn't apply when the macro/gosub jump occurs in a priority/priority label. This patch changes the logic so that the parsing still occurs, but the jump will occur even for priorities/priority labels. (issue ASTERISK-23164) Review: https://reviewboard.asterisk.org/r/3154 ........ Merged revisions 407041 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 407074 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 407082 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@407083 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-01-31ChanSpy: Add ability to specify channel uniqueids as well as channel names.Richard Mudgett
* Made ChanSpy accept a channel uniqueid or a fully specified channel name as the chanprefix parameter if the 'u' option is specified. (closes issue AFS-42) Review: https://reviewboard.asterisk.org/r/3160/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@407033 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-01-22ConfBridge: Fix channel parameter documentationKinsey Moore
Confbridge AMI and CLI commands for mute, unmute, and setting the single video source can accept channel prefixes in lieu of a full channel name, but documentation states only that it is required and is a channel name. This corrects the documentation. (closes issue PQ-1397) Reported by: Steve Pitts ........ Merged revisions 406217 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 406223 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@406224 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-01-17Documentation: doc fixes across various parts of the code for ASTERISK ↵Rusty Newton
issues 23061,23028,23046,23027 Fixes typos of "transfered" instead of "transferred" in various code. Fixes incorrect gosub param help text for app_queue. Fixes Asterisk man pages containing unquoted minus signs. Adds note about the "textsupport" option in sip.conf.sample. (issue ASTERISK-23061) (issue ASTERISK-23028) (issue ASTERISK-23046) (issue ASTERISK-23027) (closes issue ASTERISK-23061) (closes issue ASTERISK-23028) (closes issue ASTERISK-23046) (closes issue ASTERISK-23027) Reported by: Eugene, Jeremy Laine, Denis Pantsyrev Patches: transferred.patch uploaded by Jeremy Laine (license 6561) hyphen.patch uploaded by Jeremy Laine (license 6561) sip.conf.sample.patch uploaded by Eugene (license 6360) ........ Merged revisions 405791 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 405792 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 405829 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@405830 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-01-14verbosity: Fix performance of console verbose messages.Richard Mudgett
The per console verbose level feature as previously implemented caused a large performance penalty. The fix required some minor incompatibilities if the new rasterisk is used to connect to an earlier version. If the new rasterisk connects to an older Asterisk version then the root console verbose level is always affected by the "core set verbose" command of the remote console even though it may appear to only affect the current console. If an older version of rasterisk connects to the new version then the "core set verbose" command will have no effect. * Fixed the verbose performance by not generating a verbose message if nothing is going to use it and then filtered any generated verbose messages before actually sending them to the remote consoles. * Split the "core set debug" and "core set verbose" CLI commands to remove the per module verbose support that cannot work with the per console verbose level. * Added a silent option to the "core set verbose" command. * Fixed "core set debug off" tab completion. * Made "core show settings" list the current console verbosity in addition to the root console verbosity. * Changed the default verbose level of the 'verbose' setting in the logger.conf [logfiles] section. The default is now to once again follow the current root console level. As a result, using the AMI Command action with "core set verbose" could again set the root console verbose level and affect the verbose level logged. (closes issue AST-1252) Reported by: Guenther Kelleter Review: https://reviewboard.asterisk.org/r/3114/ ........ Merged revisions 405431 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 405432 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@405436 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-01-12CDRs: Synchronize dialplan applications that manipulate CDRs with the engineMatthew Jordan
In https://reviewboard.asterisk.org/r/3057/, applications and functions that manipulate CDRs were made to interact over Stasis. This was done to synchronize manipulations of CDRs from the dialplan with the updates the engine itself receives over the message bus. This change rested on a faulty premise: that messages published to the CDR topic or to a topic that forwards to the CDR topic are synchronized with the messages handled by the CDR topic subscription in the CDR engine. This is not the case. There is no ordering guaranteed for two messages published to the same topic; ordering is only guaranteed if a message is published to the same subscriber. Stasis was modified in r405311 to allow a publisher to synchronize on the subscriber. This patch uses that API to synchronize the CDR publishers with the CDR engine message router, which maintains the overall topic subscription. (closes issue ASTERISK-22884) Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/3099/ ........ Merged revisions 405312 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@405314 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-01-09app_confbridge: Fix crash caused when waitmarked/marked users leave togetherMatthew Jordan
When waitmarked users join a ConfBridge, the conference state is transitioned from EMPTY -> INACTIVE. In this state, the users are maintined in a waiting users list. When a marked user joins, the ConfBridge conference transitions from INACTIVE -> MULTI_MARKED, and all users are put onto the active list of users. This process works correctly. When the marked user leaves, if they are the last marked user, the MULTI_MARKED state does the following: (1) It plays back a message to the bridge stating that the leader has left the conference. This requires an unlocking of the bridge. (2) It moves waitmarked users back to the waiting list (3) It transitions to the appropriate state: in this case, INACTIVE However, because it plays the prompt back to the bridge before moving the users and before finishing the state transition, this creates a race condition: with the bridge unlocked, waitmarked users who leave the conference (or are kicked from it) can cause a state transition of the bridge to another state before the conference is transitioned to the INACTIVE state. This causes the state machine to get a bit wonky, often leading to a crash when the MULTI_MARKED state attempts to conclude its processing. This patch fixes this problem: (1) It prevents kicked users from being kicked again. That's just a nicety. (2) More importantly, it fixes the race condition by only playing the prompt once the state has transitioned correctly to INACTIVE. If waitmarked users sneak out during the prompt being played, no harm no foul. Review: https://reviewboard.asterisk.org/r/3108/ Note that the patch committed here is essentially the same as uploaded by Simon Moxon on ASTERISK-22740, with the addition of the double kick prevention. (closes issue AST-1258) Reported by: Steve Pitts (closes issue ASTERISK-22740) Reported by: Simon Moxon patches: ASTERISK-22740.diff uploaded by Simon Moxon (license 6546) ........ Merged revisions 405215 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 405216 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@405217 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-01-09"Minimun" typo.Walter Doekes
........ Merged revisions 405160 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 405161 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 405162 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@405163 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2014-01-06app_voicemail: Explicitly set defaultenabled=yesRichard Mudgett
........ Merged revisions 405006 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@405007 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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