4571 Commits (e23f07beb80a76160edec656d6509db5f1a2885a)

Author	SHA1	Message	Date
Walter Doekes	e23f07beb8	Fix typo's (retrieve, specified, address). ... ........ Merged revisions 430996 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@430998 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
Matthew Jordan	beb20440e0	apps/app_voicemail: Trigger MWI notification with MixMonitor m() option ... The MixMonitor m() option allows a recording to be pushed to a specific voicemail mailbox. If the message is delivered to the mailbox's INBOX, however, no MWI notification is currently raised. This patch corrects the issue by properly calling notify_new_state from the msg_create_from_file function. This will cause MWI to be triggered if the message was placed in the mailbox's INBOX. ASTERISK-24709 #close Reported by: Gareth Palmer patches: app_voicemail-430919.patch uploaded by Gareth Palmer (License 5169) ........ Merged revisions 430920 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@430921 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
Matthew Jordan	894d4d781c	apps/app_dial: Don't publish DialEnd twice on unexpected GoSub/Macro values ... The Dial application has some interesting options with the mid-call Macro (M) and GoSub (U) options. If the MACRO_RESULT/GOSUB_RESULT returns specific values, the Dial application will take some action upon the channels involved in the dial operation (such as hanging up a particular party, etc.) The Dial application ensures that a Stasis message is published in the event that MACRO_RESULT/GOSUB_RESULT returns a value that kills the dial operation, so that there is a corresponding DialEnd event published in AMI/ARI for the DialBegin event that preceeded it. A bug exists where that same DialEnd event will be published on Stasis even if the value returned in MACRO_RESULT/GOSUB_RESULT is not one that the Dial application cares about. This causes two DialEnd events to be published - one with the MACRO_RESULT/GOSUB_RESULT and another with "ANSWERED" - which is all sorts of wrong. This patch fixes the bug by ensuring that we only publish a DialEnd message to Stasis if the Dial application's mid-call Macro/GoSub returns something that Dial cares about. Review: https://reviewboard.asterisk.org/r/4336 ASTERISK-24682 #close Reported by: Matt Jordan git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@430842 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
Matthew Jordan	e659b3e53d	app_voicemail: Temp message left after review/hangup with ODBC/IMAP backend ... When using ODBC or IMAP storage, temporary files created on the file system must be disposed of using the DISPOSE macro. The DELETE macro will map to a deletion function for the backend storage, but does not clean up any local files created as a result of the operation. When using voicemail with the operator and review options enabled, pressing 0 to enter the menu, followed by 1 to save the message, followed by any other DTMF press to delete the message, will result in the temporary file lingering on the file system. This patch properly calls DISPOSE after the DELETE. This causes the local file to be disposed of. ASTERISK-24288 #close Reported by: LEI FU patches: voicemail_odbc_review_fix.diff uploaded by LEI FU (License 6640) ........ Merged revisions 430795 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@430796 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
Richard Mudgett	1f94b96749	app_macro: Don't restore the calling location on a channel redirect. ... v11: If a channel redirect to a macro exten of a macro that is active happens, the redirect location doesn't get executed. Instead the original macro location is restored and gets reexecuted. v13: An additional effect happens if a parked call times out to an extension in the macro that parked the call then the macro is reexecuted instead of the expected park return location. * Made not restore the macro calling location on an AST_SOFTHANGUP_ASYNCGOTO. * Increased the locked channel range when setting up the macro execution environment to cover things that should be done while the channel is locked. * Removed unnecessary NULL tests before calling ast_free() in _macro_exec(). ASTERISK-23850 #close Reported by: Andrew Nagy Review: https://reviewboard.asterisk.org/r/4292/ ........ Merged revisions 430564 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@430565 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
Richard Mudgett	4b363688d4	AMI: Make AMI actions that generate event lists consistent. ... * Made the following AMI actions use list API calls for consistency: Agents BridgeInfo BridgeList BridgeTechnologyList ConfbridgeLIst ConfbridgeLIstRooms CoreShowChannels DAHDIShowChannels DBGet DeviceStateList ExtensionStateList FAXSessions Hangup IAXpeerlist IAXpeers IAXregistry MeetmeList MeetmeListRooms MWIGet ParkedCalls Parkinglots PJSIPShowEndpoint PJSIPShowEndpoints PJSIPShowRegistrationsInbound PJSIPShowRegistrationsOutbound PJSIPShowResourceLists PJSIPShowSubscriptionsInbound PJSIPShowSubscriptionsOutbound PresenceStateList PRIShowSpans QueueStatus QueueSummary ShowDialPlan SIPpeers SIPpeerstatus SIPshowregistry SKINNYdevices SKINNYlines Status VoicemailUsersList * Incremented the AMI version to 2.7.0. * Changed astman_send_listack() to not use the listflag parameter and always set the value to "Start" so the start capitalization is consistent. i.e., The FAXSessions used "Start" while the rest of the system used "start". The corresponding complete event always used "Complete". * Fixed ami_show_resource_lists() "PJSIPShowResourceLists" to output the AMI ActionID for all of its list events. * Fixed off-nominal AMI protocol error in manager_bridge_info(), manager_parking_status_single_lot(), and manager_parking_status_all_lots(). Use of astman_send_error() after responding to the original AMI action request violates the action response pattern by sending two responses. * Fixed minor protocol error in action_getconfig() when no requested categories are found. Each line needs to be formatted as "Header: text". * Fixed off-nominal memory leak in manager_build_parked_call_string(). * Eliminated unnecessary use of RAII_VAR() in ami_subscription_detail(). ASTERISK-24049 #close Reported by: Jonathan Rose Review: https://reviewboard.asterisk.org/r/4315/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@430434 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
Walter Doekes	9ae57e0dd6	Fix printf problems with high ascii characters after r413586 (1.8). ... In r413586 (1.8) various casts were added to silence gcc 4.10 warnings. Those fixes included things like: -out += sprintf(out, "%%%02X", (unsigned char) *ptr); +out += sprintf(out, "%%%02X", (unsigned) *ptr); That works for low ascii characters, but for the high range that yields e.g. FFFFFFC3 when C3 is expected. This changeset: - fixes those casts to use the 'hh' unsigned char modifier instead - consistently uses %02x instead of %2.2x (or other non-standard usage) - adds a few 'h' modifiers in various places - fixes a 'replcaes' typo - dev/urandon typo (in 13+ patch) Review: https://reviewboard.asterisk.org/r/4263/ ASTERISK-24619 #close Reported by: Stefan27 (on IRC) ........ Merged revisions 429673 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 429674 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@429675 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	0321d00413	apps/app_meetme: Apply default values on initial load with no config file ... When the app_meetme module is loaded without its configuration file, the module settings aren't initialized. In particular, this impacts the use of logging realtime members. This patch guarantees that we always set the default module settings on initial load. Review: https://reviewboard.asterisk.org/r/4242/ ASTERISK-24572 #close Reported by: Nuno Borges patches: 24572.patch uploaded by Nuno Borges (License 6116) ........ Merged revisions 429027 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 429028 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@429029 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	bda506c346	apps/app_voicemail: Fix crash with IMAP when streams are opened simultaneously ... The UW IMAP library is instrinsically not thread-safe, and relies upon higher level applications to guarantee thread safety. For the most part, this is provided by the vms object, which provides locking for individual streams. Unfortunately, this is not sufficient for calls to mail_open which create the IMAP stream. mail_open can, on some systems, call into a UW IMAP specific function for determining the address of a system based on a hostname, ip_nametoaddr. In the ip6_unix implementation of this function, static variables are used to hold parsing buffers. This can cause a crash if multiple threads attempt to convert a hostname to an address at the same time. Locking on a single mail stream is not sufficient to prevent simultaneous access to these static variables. In the IMAP library, this function can be called from the mail_open and imap_status functions. As the imap_status function is not used by app_voicemail, locking on access to mail_open is sufficient to prevent any mangling of the buffers. Review: https://reviewboard.asterisk.org/r/4188/ ASTERISK-24516 #close Reported by: David Duncan Ross Palmer Tested by: David Duncan Ross Palmer patches: ASTERISK-24516.diff uploaded by David Duncan Ross Palmer (License 6660) ........ Merged revisions 428863 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 428864 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@428865 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	d79c68d3fb	main/stasis: Allow subscriptions to use a threadpool for message delivery ... Prior to this patch, all Stasis subscriptions would receive a dedicated thread for servicing published messages. In contrast, prior to r400178 (see review https://reviewboard.asterisk.org/r/2881/), the subscriptions shared a thread pool. It was discovered during some initial work on Stasis that, for a low subscription count with high message throughput, the threadpool was not as performant as simply having a dedicated thread per subscriber. For situations where a subscriber receives a substantial number of messages and is always present, the model of having a dedicated thread per subscriber makes sense. While we still have plenty of subscriptions that would follow this model, e.g., AMI, CDRs, CEL, etc., there are plenty that also fall into the following two categories: * Large number of subscriptions, specifically those tied to endpoints/peers. * Low number of messages. Some subscriptions exist specifically to coordinate a single message - the subscription is created, a message is published, the delivery is synchronized, and the subscription is destroyed. In both of the latter two cases, creating a dedicated thread is wasteful (and in the case of a large number of peers/endpoints, harmful). In those cases, having shared delivery threads is far more performant. This patch adds the ability of a subscriber to Stasis to choose whether or not their messages are dispatched on a dedicated thread or on a threadpool. The threadpool is configurable through stasis.conf. Review: https://reviewboard.asterisk.org/r/4193 ASTERISK-24533 #close Reported by: xrobau Tested by: xrobau ........ Merged revisions 428681 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@428687 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Joshua Colp	6e30ccd242	app_record: Fix bug where using the 'k' option and hanging up would trim 1/4 of a second of the recording. ... The Record dialplan function trims 1/4 of a second from the end of recordings in case they are terminated because of DTMF. When hanging up, however, you don't want this to happen. This change makes it so on hangup this does not occur. ASTERISK-24530 #close Reported by: Ben Smithurst patches: app_record_v2.diff submitted by Ben Smithurst (license 6529) Review: https://reviewboard.asterisk.org/r/4201/ ........ Merged revisions 428653 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 428654 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@428655 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Kevin Harwell	19ba0d1363	AST-2014-017 - app_confbridge: permission escalation/ class authorization. ... Confbridge dialplan function permission escalation via AMI and inappropriate class authorization on the ConfbridgeStartRecord action. The CONFBRIDGE dialplan function when executed from an external protocol (for instance AMI), could result in a privilege escalation. Also, the AMI action “ConfbridgeStartRecord” could also be used to execute arbitrary system commands without first checking for system access. Asterisk now inhibits the CONFBRIDGE function from being executed from an external interface if the live_dangerously option is set to no. Also, the “ConfbridgeStartRecord” AMI action is now only allowed to execute under a user with system level access. ASTERISK-24490 Reported by: Gareth Palmer ........ Merged revisions 428332 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 428334 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@428339 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	45f4fe6f1c	apps/app_confbridge: Ensure 'normal' users hear message when last marked leaves ... When r428077 was made for ASTERISK-24522, it failed to take into account users who are neither wait_marked nor end_marked. These users are *also* supposed to hear the 'leader has left the conference' message. Granted, this behaviour is a bit odd; however, that is how it used to work... and behaviour changes are not good. This patch ensures that if there are any 'normal' users present when the last marked user leaves the conference, the message will still be played to them. Note that this regression was caught by the Asterisk Test Suite's confbridge_nominal test, which has a quirky combination of users. ........ Merged revisions 428113 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 428114 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@428115 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	6d59f1546e	app_confbridge: Don't play leader leaving prompt if no one will hear it ... Consider the following: - A marked user in a conference - One or more end_marked only users in the conference When the marked users leaves, we will be in the conf_state_multi_marked state. This currently will traverse the users, kicking out any who have the end_marked flags. When they are kicked, a full ast_bridge_remove is immediately called on the channels. At this time, we also unilaterally set the need_prompt flag. When the need_prompt flag is set, we then playback a sound to the bridge informing everyone that the leader has left; however, no one is left in the bridge. This causes some odd behaviour for the end_marked users - they are stuck waiting for the bridge to be unlocked. This results in them waiting for 5 or 6 seconds of dead air before hearing that they've been kicked. Unfortunately, we do have to keep the bridge locked while we're playing back the 'leader-has-left' prompt. If there are any wait_marked users in the conference, this behaviour can't be easily changed - but we do make the case of the end_marked users better with this patch. Review: https://reviewboard.asterisk.org/r/4184/ ASTERISK-24522 #close Reported by: Matt Jordan ........ Merged revisions 428077 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 428078 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@428079 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Mark Michelson	1536b0ecb6	Fix race condition that could result in ARI transfer messages not being sent. ... From reviewboard: "During blind transfer testing, it was noticed that tests were failing occasionally because the ARI blind transfer event was not being sent. After investigating, I detected a race condition in the blind transfer code. When blind transferring a single channel, the actual transfer operation (i.e. removing the transferee from the bridge and directing them to the proper dialplan location) is queued onto the transferee bridge channel. After queuing the transfer operation, the blind transfer Stasis message is published. At the time of publication, snapshots of the channels and bridge involved are created. The ARI subscriber to the blind transfer Stasis message then attempts to determine if the bridge or any of the involved channels are subscribed to by ARI applications. If so, then the blind transfer message is sent to the applications. The way that the ARI blind transfer message handler works is to first see if the transferer channel is subscribed to. If not, then iterate over all the channel IDs in the bridge snapshot and determine if any of those are subscribed to. In the test we were running, the lone transferee channel was subscribed to, so an ARI event should have been sent to our application. Occasionally, though, the bridge snapshot did not have any channels IDs on it at all. Why? The problem is that since the blind transfer operation is handled by a separate thread, it is possible that the transfer will have completed and the channels removed from the bridge before we publish the blind transfer Stasis message. Since the blind transfer has completed, the bridge on which the transfer occurred no longer has any channels on it, so the resulting bridge snapshot has no channels on it. Through investigation of the code, I found that attended transfers can have this issue too for the case where a transferee is transferred to an application." The fix employed here is to decouple the creation of snapshots for the transfer messages from the publication of the transfer messages. This way, snapshots can be created to reflect what they are at the time of the transfer operation. Review: https://reviewboard.asterisk.org/r/4135 ........ Merged revisions 427848 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@427870 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Joshua Colp	9067e6faa4	app_confbridge: Play "leader has left" sound even when musiconhold is enabled. ... Currently if the leader of a conference bridge leaves any participant that has musiconhold enabled will not hear the "leader has left" sound. This is because musiconhold is started and THEN the sound is played. This change makes it so that the sound is played and THEN musiconhold is started. This provides a better experience for users as they may not have known previously why they went back to musiconhold. Review: https://reviewboard.asterisk.org/r/4177/ ........ Merged revisions 427844 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 427845 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@427846 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	ffe194fd78	app_voicemail: Fix enhancement that allowed multiple recipients in To: header ... An issue existed in r420577, which added multiple recipients to voicemail emails. The patch, when looking at the intended recipients, looked ahead for the '|' character inside a while loop which already had pulled out the appropriate field parsing on the '|' character. This would cause it to skip the recipients. This patch fixes it such that it relies completely on the while loop to parse through the e-mail fields. Note that the original author of the patch looked at this fix and approved it. ASTERISK-24250 #close Reported by: abelbeck patches: voicemail-420577-to-comma-fix.diff uploaded by abelbeck (License 5903) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@427585 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	a2d6e15af8	app_agent_pool: Made agent alert interruptable by DTMF. ... Made agent able to interrupt the alerting beep playback with DTMF. Any digit can interrupt if the call does not need to be acknowledged. Only the first digit of the acknowledgement can interrupt if the call needs to be acknowledged. The agent interrupting the alerting playback builds on the ASTERISK-24447 patch because it knows what digit interrupted the playback and needs to be able to pass that digit to the DTMF hook digit collection code. ASTERISK-24257 #close Reported by: Steve Pitts Review: https://reviewboard.asterisk.org/r/4123/ ........ Merged revisions 427508 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@427512 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Corey Farrell	5bec46e6c8	Fix compile error caused by review 4138 ... There is no procedure called ast_closeframe, fix code to use ast_closestream. Reported By: Matt Jordan ........ Merged revisions 427087 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 427088 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@427089 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Corey Farrell	5f17490f4d	Fix ast_writestream leaks ... Fix cleanup in __ast_play_and_record where others[x] may be leaked. This was caught where prepend != NULL && outmsg != NULL, once realfile[x] == NULL any further others[x] would be leaked. A cleanup block was also added for prepend != NULL && outmsg == NULL. 11+: Fix leak of ast_writestream recording_fs in app_voicemail:leave_voicemail. ASTERISK-24476 #close Reported by: Corey Farrell Review: https://reviewboard.asterisk.org/r/4138/ ........ Merged revisions 427023 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 427024 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 427025 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@427026 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Corey Farrell	7bd256b711	app_queue: fix a couple leaks to struct call_queue in set_member_value ... set_member_value has a couple leaks to references in the variable q found through testsuite tests/queues/set_penalty. Also remove the REF_DEBUG_ONLY_QUEUES compiler declaration, this is no longer possible with the updated REF_DEBUG code. ASTERISK-24466 #close Reported by: Corey Farrell Review: https://reviewboard.asterisk.org/r/4125/ ........ Merged revisions 426805 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 426806 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@426807 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Walter Doekes	3f31b73f54	app_voicemail: Fix unchecked bounds of myArray in IMAP_STORAGE. ... In update_messages_by_imapuser(), messages were appended to a finite array which resulted in a crash when an IMAP mailbox contained more than 256 entries. This memory is now dynamically increased as needed. Observe that this patch adds a bunch of XXX's to questionable code. See the review (url below) for more information. ASTERISK-24190 #close Reported by: Nick Adams Tested by: Nick Adams Review: https://reviewboard.asterisk.org/r/4126/ ........ Merged revisions 426691 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 426692 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 426696 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@426702 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Corey Farrell	688edd55c3	app_queue: Cleanup ao2_iterator ... Clean ao2_iterator, resolving reference leak to queue members. ASTERISK-24454 #close Reported by: Corey Farrell Review: https://reviewboard.asterisk.org/r/4111/ ........ Merged revisions 426255 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 426260 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@426266 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
George Joseph	8bb06d592a	manager/config: Support templates and non-unique category names via AMI ... This patch provides the capability to manipulate templates and categories with non-unique names via AMI. Summary of changes: GetConfig and GetConfigJSON: Added "Filter" parameter: A comma separated list of name_regex=value_regex expressions which will cause only categories whose variables match all expressions to be considered. The special variable name TEMPLATES can be used to control whether templates are included. Passing 'include' as the value will include templates along with normal categories. Passing 'restrict' as the value will restrict the operation to ONLY templates. Not specifying a TEMPLATES expression results in the current default behavior which is to not include templates. UpdateConfig: NewCat now includes options for allowing duplicate category names, indicating if the category should be created as a template, and specifying templates the category should inherit from. The rest of the actions now accept a filter string as defined above. If there are non-unique category names, you can now update specific ones based on variable values. To facilitate the new capabilities in manager, corresponding changes had to be made to config, most notably the addition of filter criteria to many of the APIs. In some cases it was easy to change the references to use the new prototype but others would have required touching too many files for this patch so a wrapper with the original prototype was created. Macros couldn't be used in this case because it would break binary compatibility with modules such as res_digium_phone that are linked to real symbols. Tested-by: George Joseph Review: https://reviewboard.asterisk.org/r/4033/ ........ Merged revisions 425383 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@425384 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	cff192429b	audiohooks: Reevaluate the bridge technology when an audiohook is added or removed. ... Adding a mixmonitor to a channel causes the bridge to change technologies from native to simple_bridge so the call can be recorded. However, when the mixmonitor is stopped the bridge does not switch back to the native technology. * Added unbridge requests to reevaluate the bridge when a channel audiohook is removed. * Moved the unbridge request into ast_audiohook_attach() ensure that the bridge reevaluates whenever an audiohook is attached. This simplified the mixmonitor and chan_spy start code as well. * Added defensive code to stop_mixmonitor_full() in case additional arguments are ever added to the StopMixMonitor application. * Made ast_framehook_detach() not do an unbridge request if the framehook does not exist. * Made ast_framehook_list_fixup() do an unbridge request if there are any framehooks. Also simplified the loop. ASTERISK-24195 #close Reported by: Jonathan Rose Review: https://reviewboard.asterisk.org/r/4046/ ........ Merged revisions 424506 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@424507 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Scott Griepentrog	79b702f308	Voicemail: get correct duration when copying file to vm ... Changes made during format improvements resulted in the recording to voicemail option 'm' of the MixMonitor app writing a zero length duration in the msgXXXX.txt file. This change introduces a new function ast_ratestream(), which provides the sample rate of the format associated with the stream, and updates the app_voicemail function for ast_app_copy_recording_to_vm to calculate the right duration. Review: https://reviewboard.asterisk.org/r/3996/ ASTERISK-24328 #close git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@423192 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	0fbd9947e2	main/cdrs: Preserve context/extension when executing a Macro or GoSub ... The context/extension in a CDR is generally considered the destination of a call. When looking at a 2-party call CDR, users will typically be presented with the following: context exten channel dest_channel app data default 1000 SIP/8675309 SIP/1000 Dial SIP/1000,,20 However, if the Dial actually takes place in a Macro, the current behaviour in 12 will result in the following CDR: context exten channel dest_channel app data macro-dial s SIP/8675309 SIP/1000 Dial SIP/1000,,20 The same is true of a GoSub: context exten channel dest_channel app data subs dial_stuff SIP/8675309 SIP/1000 Dial SIP/1000,,20 This generally makes the context/exten fields less than useful. It isn't hard to preserve these values in the CDR state machine; however, we need to have something that informs us when a channel is executing a subroutine. Prior to this patch, there isn't anything that does this. This patch solves this problem by adding a new channel flag, AST_FLAG_SUBROUTINE_EXEC. This flag is set on a channel when it executes a Macro or a GoSub. The CDR engine looks for this value when updating a Party A snapshot; if the flag is present, we don't override the context/exten on the main CDR object. In a funny quirk, executing a hangup handler must *not* abide by this logic, as the endbeforehexten logic assumes that the user wants to see data that occurs in hangup logic, which includes those subroutines. Since those execute outside of a typical Dial operation (and will typically have their own dedicated CDR anyway), this is unlikely to cause any heartburn. Review: https://reviewboard.asterisk.org/r/3962/ ASTERISK-24254 #close Reported by: tm1000, Tony Lewis Tested by: Tony Lewis ........ Merged revisions 422718 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@422719 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
George Joseph	e0c66f5f14	confbridge: Add Duration to ConfbridgeList event ... The ConfbridgeList event doesn't include how long the user has been a member of the conference. This patch adds Duration (seconds) which is based on user->chan->answertime. Tested by: George Joseph Review: https://reviewboard.asterisk.org/r/3955/ ........ Merged revisions 422444 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@422445 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
George Joseph	372236cc2a	confbridge: Add 'Admin' param to join, leave, mute, unmute and talking events ... Currently there's no way to tell if a user is an admin or not when receiving the join, leave, mute, unmute and talking events. This patch adds that capability. Tested by: George Joseph Review: https://reviewboard.asterisk.org/r/3950/ ........ Merged revisions 422176 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@422177 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
George Joseph	e985cb076d	confbridge: Make kick, mute and unmute handle channel targets consistently. ... Kick, mute and unmute were a little inconsistent in their handling of channel targets. This patch cleans that up by insuring they all handle the 'all' target consistently and adds the 'participants' target which acts on non-admins. Documentation for kick was also cleaned up as it never supported partial channel names. Tested by: George Joseph Review: https://reviewboard.asterisk.org/r/3944/ ........ Merged revisions 422090 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@422091 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Mark Michelson	16c96760e7	Fix a locking inversion in MixMonitor. ... We need to unlock the audiohook before trying to lock the channel, since the correct locking order is channel then audiohook. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@421882 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	b9906d8438	apps/app_meetme: Fix crash when publishing MeetMe messages with no channel ... The same function, meetme_stasis_generate_msg, handles creating and publishing Stasis message both when there are channels in the MeetMe conference and when there are no channels in the conference. When the performance improvement was made to use cached snapshots, this created a situation where Asterisk would crash: obtaining a cached snapshot is not NULL tolerant. This patch restores the previous implementation, which used a NULL safe set of routines to produce a blob containing the channel snapshot (if available) and information about the MeetMe conference. ASTERISK-24234 #close Reported by: Shaun Ruffell Tested by: Shaun Ruffell ........ Merged revisions 421270 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@421273 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	f5d4e581f3	apps/app_dial: Fix Dial 'z' option ... The 'z' option is supposed to disable the dial timeout in the case of a call forward. Unfortunately, the wrong timeout timer was passed to the do_forward function, resulting in the option not working. ASTERISK-24225 #close Reported by: dimitripietro Tested by: dimitripietro patches: jira_asterisk_24225_v1.8.patch uploaded by rmudgett (License 5621) jira_asterisk_24225_v11.patch uploaded by rmudgett (License 5621) ........ Merged revisions 421232 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 421233 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 421234 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@421235 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	544e092b2d	app_voicemail/app: Remove test events that were duplicated by r421059 ... Moving the test event raised when a file is played back (which occurred in r421059) broke the ever loving snot out of the voicemail tests. This caused duplicate test events to get raised, as app_voicemail and main/app were raising events prior to call ast_streamfile. The voicemail tests did not enjoy getting multiple events. Since raising the playback event in ast_streamfile is far more useful to the vast majority of tests, this patch keeps the call there and simply removes the extraneous calls that duplicated the event. ........ Merged revisions 421125 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 421164 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 421165 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@421166 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Jonathan Rose	cd28e5dda2	Bridges: Fix feature interruption/unintended kick caused by external actions ... If a manager or CLI user attached a mixmonitor to a call running a dynamic bridge feature while in a bridge, the feature would be interrupted and the channel would be forcibly kicked out of the bridge (usually ending the call during a simple 1 to 1 call). This would also occur during any similar action that could set the unbridge soft hangup flag, so the fix for this was to remove unbridge from the soft hangup flags and make it a separate thing all together. ASTERISK-24027 #close Reported by: mjordan Review: https://reviewboard.asterisk.org/r/3900/ ........ Merged revisions 420934 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@420940 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Joshua Colp	a2bbe5d360	app_voicemail: Fix the "test_voicemail_vm_info" unit test. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@420856 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Mark Michelson	ef70c08dc7	Improve call forwarding reporting, especially with regards to ARI. ... This patch addresses a few issues: 1) The order of Dial events have been changed when performing a call forward. The order has now been altered to 1) Dial begins dialing channel A. 2) When A forwards the call to B, we issue the dial end event to channel A, indicating the dial is being canceled due to a forward to B. 3) When the call to channel B occurs, we then issue a new dial begin to channel B. 2) Call forwards are now reported on the calling channel, not the peer channel. 3) AMI DialEnd events have been altered to display the extension the call is being forwarded to when relevant. 4) You can now get the values of channel variables for channels that are not currently in the Stasis application. This brings the retrieval of channel variables more in line with the rest of channel read operations since they may be performed on channels not in Stasis. ASTERISK-24134 #close Reported by Matt Jordan ASTERISK-24138 #close Reported by Matt Jordan Patches: forward-shenanigans.diff uploaded by Matt Jordan (License #6283) Review: https://reviewboard.asterisk.org/r/3899 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@420794 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	95871451f6	app_queue: Add RealTime support for queue rules ... This patch gives the optional ability to keep queue rules in RealTime. It is important to note that with this patch: (a) Queue rules in RealTime are only examined on module load/reload (b) Queue rules are loaded both from the queuerules.conf file as well as the RealTime backend To inform app_queue to examine RealTime for queue rules, a new setting has been added to queuerules.conf's general section "realtime_rules". RealTime queue rules will only be used when this setting is set to "yes". The schema for the database table supports a rule_name, time, min_penalty, and max_penalty columns. min_penalty and max_penalty can be relative, if a '-' or '+' literal is provided. Otherwise, the penalties are treated as constants. For example: rule_name, time, min_penalty, max_penalty 'default', '10', '20', '30' 'test2', '20', '30', '55' 'test2', '25', '-11', '+1111' 'test2', '400', '112', '333' 'test3', '0', '4564', '46546' 'test_rule', '40', '15', '50' which would result in : Rule: default - After 10 seconds, adjust QUEUE_MAX_PENALTY to 30 and adjust QUEUE_MIN_PENALTY to 20 Rule: test2 - After 20 seconds, adjust QUEUE_MAX_PENALTY to 55 and adjust QUEUE_MIN_PENALTY to 30 - After 25 seconds, adjust QUEUE_MAX_PENALTY by 1111 and adjust QUEUE_MIN_PENALTY by -11 - After 400 seconds, adjust QUEUE_MAX_PENALTY to 333 and adjust QUEUE_MIN_PENALTY to 112 Rule: test3 - After 0 seconds, adjust QUEUE_MAX_PENALTY to 46546 and adjust QUEUE_MIN_PENALTY to 4564 Rule: test_rule - After 40 seconds, adjust QUEUE_MAX_PENALTY to 50 and adjust QUEUE_MIN_PENALTY to 15 If you use RealTime, the queue rules will be always reloaded on a module reload, even if the underlying file did not change. With the option disabled, the rules will only be reloaded if the file was modified. Review: https://reviewboard.asterisk.org/r/3607/ ASTERISK-23823 #close Reported by: Michael K patches: app_queue.c_realtime_trunk.patch uploaded by Michael K (License 6621) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@420624 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Jason Parker	7438e38f22	Fix build in devmode. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@420592 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Jason Parker	2b804661bc	app_voicemail: Add the ability to specify multiple email addresses. ... ASTERISK-24045 Reported by: Jacob Barber Review: https://reviewboard.asterisk.org/r/3833/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@420577 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Kinsey Moore	0ac7f96057	Stasis: Convey transfer information to applications ... This fixes a class of issues where Stasis applications were not made aware that their channels were being manipulated or replaced by external entitiessuch as transfers, AMI commands, or dialplan applications such as Bridge(). Inconsistent information such as StasisEnd events with unknown channels as a result of masquerades has also been corrected. To accomplish these fixes, several new fields were added to blind and attended transfer messages as well as StasisStart and BridgeAttendedTransfer Stasis events. ASTERISK-23941 #close Review: https://reviewboard.asterisk.org/r/3865/ Review: https://reviewboard.asterisk.org/r/3857/ Review: https://reviewboard.asterisk.org/r/3852/ Review: https://reviewboard.asterisk.org/r/3816/ Review: https://reviewboard.asterisk.org/r/3731/ Review: https://reviewboard.asterisk.org/r/3729/ Review: https://reviewboard.asterisk.org/r/3728/ ........ Merged revisions 420325 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@420338 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Kinsey Moore	f1036f40dc	Stasis: Allow message types to be blocked ... This introduces stasis.conf and a mechanism to prevent certain message types from being published. Internally, this works by preventing the chosen message types from being created which ensures that those message types can never be published. This patch also adjusts message publishers such that message payloads are not created if the related message type is not available. ASTERISK-23943 #close Review: https://reviewboard.asterisk.org/r/3823/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@420124 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	2758cc76e5	datastores: Audit ast_channel_datastore_remove usage. ... Audit of v1.8 usage of ast_channel_datastore_remove() for datastore memory leaks. * Fixed leaks in app_speech_utils and func_frame_trace. * Fixed app_speech_utils not locking the channel when accessing the channel datastore list. Review: https://reviewboard.asterisk.org/r/3859/ Audit of v11 usage of ast_channel_datastore_remove() for datastore memory leaks. * Fixed leak in func_jitterbuffer. (Was not in v12) Review: https://reviewboard.asterisk.org/r/3860/ Audit of v12 usage of ast_channel_datastore_remove() for datastore memory leaks. * Fixed leaks in abstract_jb. * Fixed leak in ast_channel_unsuppress(). Used by ARI mute control and res_mutestream. * Fixed ref leak in ast_channel_suppress(). Used by ARI mute control and res_mutestream. Review: https://reviewboard.asterisk.org/r/3861/ ........ Merged revisions 419684 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 419685 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 419686 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@419688 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Mark Michelson	dcf1ad14da	Add module support level to ast_module_info structure. Print it in CLI "module show" . ... ASTERISK-23919 #close Reported by Malcolm Davenport Review: https://reviewboard.asterisk.org/r/3802 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@419592 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Joshua Colp	41042588b9	app_bridgewait: Remove possibility of race condition between channels leaving/joining. ... Bridges created by app_bridgewait previously had the "dissolve when empty" flag set. This caused the bridge core to destroy them when the last channel had left. This introduced a race condition where we may have a reference to the bridge but it is not actually joinable when we try to join it. This flag has now been removed and the bridge is guaranteed to be joinable at all times. ASTERISK-23987 #close Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/3836/ ........ Merged revisions 419538 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@419539 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	a2ce95d9d2	accountcode: Slightly change accountcode propagation. ... The previous behavior was to simply set the accountcode of an outgoing channel to the accountcode of the channel initiating the call. It was done this way a long time ago to allow the accountcode set on the SIP/100 channel to be propagated to a local channel so the dialplan execution on the Local;2 channel would have the SIP/100 accountcode available. SIP/100 -> Local;1/Local;2 -> SIP/200 Propagating the SIP/100 accountcode to the local channels is very useful. Without any dialplan manipulation, all channels in this call would have the same accountcode. Using dialplan, you can set a different accountcode on the SIP/200 channel either by setting the accountcode on the Local;2 channel or by the Dial application's b(pre-dial), M(macro) or U(gosub) options, or by the FollowMe application's b(pre-dial) option, or by the Queue application's macro or gosub options. Before Asterisk v12, the altered accountcode on SIP/200 will remain until the local channels optimize out and the accountcode would change to the SIP/100 accountcode. Asterisk v1.8 attempted to add peeraccount support but ultimately had to punt on the support. The peeraccount support was rendered useless because of how the CDR code needed to unconditionally force the caller's accountcode onto the peer channel's accountcode. The CEL events were thus intentionally made to always use the channel's accountcode as the peeraccount value. With the arrival of Asterisk v12, the situation has improved somewhat so peeraccount support can be made to work. Using the indicated example, the the accountcode values become as follows when the peeraccount is set on SIP/100 before calling SIP/200: SIP/100 ---> Local;1 ---- Local;2 ---> SIP/200 acct: 100 \/ acct: 200 \/ acct: 100 \/ acct: 200 peer: 200 /\ peer: 100 /\ peer: 200 /\ peer: 100 If a channel already has an accountcode it can only change by the following explicit user actions: 1) A channel originate method that can specify an accountcode to use. 2) The calling channel propagating its non-empty peeraccount or its non-empty accountcode if the peeraccount was empty to the outgoing channel's accountcode before initiating the dial. e.g., Dial and FollowMe. The exception to this propagation method is Queue. Queue will only propagate peeraccounts this way only if the outgoing channel does not have an accountcode. 3) Dialplan using CHANNEL(accountcode). 4) Dialplan using CHANNEL(peeraccount) on the other end of a local channel pair. If a channel does not have an accountcode it can get one from the following places: 1) The channel driver's configuration at channel creation. 2) Explicit user action as already indicated. 3) Entering a basic or stasis-mixing bridge from a peer channel's peeraccount value. You can specify the accountcode for an outgoing channel by setting the CHANNEL(peeraccount) before using the Dial, FollowMe, and Queue applications. Queue adds the wrinkle that it will not overwrite an existing accountcode on the outgoing channel with the calling channels values. Accountcode and peeraccount values propagate to an outgoing channel before dialing. Accountcodes also propagate when channels enter or leave a basic or stasis-mixing bridge. The peeraccount value only makes sense for mixing bridges with two channels; it is meaningless otherwise. * Made peeraccount functional by changing accountcode propagation as described above. * Fixed CEL extracting the wrong ie value for the peeraccount. This was done intentionally in Asterisk v1.8 when that version had to punt on peeraccount. * Fixed a few places dealing with accountcodes that were reading from channels without the lock held. AFS-65 #close Review: https://reviewboard.asterisk.org/r/3601/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@419520 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Scott Griepentrog	b9ac1feed7	app_voicemail: use a consistent generator string ... When updating voicemail.conf when a user changes their pin, change the generator string to be the same as the module name when reading so that the same config_hook will be called. Review: https://reviewboard.asterisk.org/r/3837/ ........ Merged revisions 419284 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 419285 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@419286 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Michael L. Young	20cb961b3e	apps/app_mixmonitor: Add Options To Play Beep At Start Or Stop ... We have a new periodic beep feature but sometimes a user needs some sort of feedback, without the need to have a periodic beep during the recording, to let them know that MixMonitor started recording or ended the recording. The use case where this patch is being used is when using Dynamic Features to start and end MixMonitor. This patch adds an option to play a beep when MixMonitor starts and an option to play a beep when MixMonitor ends. ASTERISK-24051 #close Reported by: Michael L. Young patches: mixmonitor-play-beep-start-stop.diff uploaded by Michael L. Young (license 5026) Review: https://reviewboard.asterisk.org/r/3820/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@419238 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Kinsey Moore	9056c23bbd	Fix more dev-mode build issues ... ........ Merged revisions 419129 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 419162 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 419163 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@419175 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Corey Farrell	e04607f8a3	res_smdi: convert to astobj2 ... Remove functions: ast_smdi_interface_unref ast_smdi_md_message_putback ast_smdi_mwi_message_putback ast_smdi_md_message destructor ast_smdi_mwi_message destructor Includes for astobj.h are removed everywhere it's possible. ASTERISK-24066 #close Review: https://reviewboard.asterisk.org/r/3758/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@419060 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago