4364 Commits (2befe82721e5f19b1dadfd2011cc491c25f15953)

Author	SHA1	Message	Date
George Joseph	a0dc90c5b8	app_voicemail: Fix compile breaking in app_voicemail with IMAP_STORAGE. ... There is a leftover "assert" in app_voicemail/__messagecount that references variables that don't exist. This causes the compile to fail when --enable-dev-mode and IMAP_STORAGE are selected. This patch removes the assert. Tested-by: George Joseph Review: https://reviewboard.asterisk.org/r/4461/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@432484 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
Kevin Harwell	9445a45188	app_chanspy, channel: fix frame leaks ... Fixed a couple of frame leaks that were found during testing. ASTERISK-24828 #close Reported by: John Hardin Review: https://reviewboard.asterisk.org/r/4445/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@432362 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
Matthew Jordan	8bb48e1f50	make: Remove 'res_features' from libraries to link against with cygwin/mingw32 ... Both the apps and channels Makefiles still listed 'res_features' as modules to link against when compiling for cygwin or mingw32. This module hasn't existed for quite some time. ASTERISK-18105 #close Reported by: feyfre git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@432341 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
Matthew Jordan	37df042fd8	apps/app_voicemail: Demote an ERROR message to a WARNING message ... When using IMAP voicemail with FreePBX, you will often get ERROR messages complaining about not being able to find a mailbox. This is due to how FreePBX handles voicemail mailboxes. Unfortunately, app_voicemail has to consider this a configuration error, as in any other system it would be indicative of someone misconfiguring their system. Regardless, a misconfiguration is a WARNING, and not an ERROR. This patch demotes the message so that system administrators can hopefully reduce some of the noise in their log files. Note that in the original patch this was made into a NOTICE, but that's a too forgiving. ASTERISK-24790 #close Reported by: Graham Barnett patches: app_voicemail.c.patch_noise uploaded by Graham Barnett (License 6685) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@432098 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
Matthew Jordan	991f979039	apps/app_voicemail: Fix IMAP header compatibility issue with Microsoft Exchange ... When interfacing with Microsoft Exchange, custom headers will be returned as all lower case. Currently, the IMAP header code will fail to parse the returned custom headers, as it will be performing a case sensitive comparison. This can cause playback of messages to fail, as needed information - such as origtime - will not be present. This patch updates app_voicemail's header parsing code to perform a case insensitive lookup for the requested custom headers. Since the headers are specific to Asterisk, e.g., 'x-asterisk-vm-orig-time', and headers should be unique in an IMAP message, this should cause no issues with other systems. ASTERISK-24787 #close Reported by: Graham Barnett patches: app_voicemail.c.patch_MSExchange uploaded by Graham Barnett (License 6685) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@432012 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
Matthew Jordan	a1ed030c5c	apps/app_mixmonitor: Move Test Event for MIXMONITOR_END to after it finishes ... The Test Event for MIXMONITOR_END - which signals that a MixMonitor has completed - technically fired before the filestream was closed. If a test used this to trigger a condition to verify that the file was written, it could result in a race condition where the file size would not be what the test expected. Luckily, no tests were using this (although they should have been). Since the test event needed to be moved after the point where the MixMonitor autochan has been destroyed, the test event no longer emits the channel name. Luckily, nothing needs it. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@431788 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
Richard Mudgett	c9ce281846	app_confbridge: Repeatedly starting and stopping recording ref leaks the recording channel. ... Starting and stopping conference recording more than once causes the recording channels to be leaked. For v13 the channels also show up in the CLI "core show channels" output. * Reworked and simplified the recording channel code to use ast_bridge_impart() instead of managing the recording thread in the ConfBridge code. The recording channel's ref handling easily falls into place and other off nominal code paths get handled better as a result. ASTERISK-24719 #close Reported by: John Bigelow Review: https://reviewboard.asterisk.org/r/4368/ Review: https://reviewboard.asterisk.org/r/4369/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@431135 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
Richard Mudgett	045557ad1b	app_confbridge: Whitespace ... Because there is sometimes no sence to any whitespace. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@431049 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
Walter Doekes	08efda063a	Fix typo's (retrieve, specified, address). ... git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@430996 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
Matthew Jordan	189bbe46c0	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) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@430920 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
Matthew Jordan	ee3290a70c	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) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@430795 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	2ecff992af	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/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@430564 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Walter Doekes	822abf9e9b	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) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@429673 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	c02beb1097	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) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@429027 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	4a054379de	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) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@428863 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Joshua Colp	d2d6a36bc8	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/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@428653 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Kevin Harwell	060ced4b54	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 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@428332 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	f20ddb1285	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. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@428113 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	10d242b728	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 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@428077 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Joshua Colp	093db340b1	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/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@427844 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Corey Farrell	47ee18acc1	Fix compile error caused by review 4138 ... There is no procedure called ast_closeframe, fix code to use ast_closestream. Reported By: Matt Jordan git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@427087 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Corey Farrell	9dc2f92921	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 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@427024 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Corey Farrell	a3ec9d8f1b	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/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@426805 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Walter Doekes	15f16e3187	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 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@426692 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Corey Farrell	37d9bfdd05	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/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@426255 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	a640d70ae8	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 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@421233 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	252ead3b13	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 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@421164 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	c2e464699f	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. Review: https://reviewboard.asterisk.org/r/3860/ ........ Merged revisions 419684 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@419685 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Scott Griepentrog	59795008d5	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/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@419284 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Kinsey Moore	22b9d0ddff	Fix more dev-mode build issues ... ........ Merged revisions 419129 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@419162 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Corey Farrell	7a914e14d0	Fix minor reference leaks in app_skel and TEST_FRAMEWORK ... * Cleanup games object in app_skel. * Cleanup stasis subscription to TEST_FRAMEWORK in manager.c (12+). Review: https://reviewboard.asterisk.org/r/3757/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@418465 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Scott Griepentrog	a5f39fc2ba	app_queue: delayed state can cause early leavewhenempty ringing ... 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 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@415835 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Jonathan Rose	064bd035e7	MixMonitor: Add class authorization requirements to MixMonitor AMI commands ... 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 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@415825 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	a890f5469b	app_confbridge: Allow muting of users waiting to enter a ConfBridge ... 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) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@415206 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Corey Farrell	b9838a9960	app_confbridge: Correct verification of conference name length ... 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 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@415066 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	c3d1e68377	app_meetme: Don't interrupt MOH for waitmarked users. ... 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 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@414402 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	e5d1800160	app_meetme: Fix overwrite of DAHDI conference data structure. ... 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 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@413992 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Jonathan Rose	c4e0f4361f	app_chanspy: Fix a test that was failing on account of r413551 ... ASTERISK-23381 #close ASTERISK-23381 #comment Reported by: Robert Moss Review: https://reviewboard.asterisk.org/r/3505/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@413710 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Kinsey Moore	abac3330cf	Allow Asterisk to compile under GCC 4.10 ... 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 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@413587 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Jonathan Rose	d55a68a531	app_chanspy: Fix a bug where Barge mode could fail ... 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/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@413551 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Joshua Colp	50925e6c24	app_queue: Extend documentation for various Manager actions and events. ... ........ Merged revisions 413485 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@413486 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	f9e01d04a6	app_confbridge: Fix ref leak in CLI "confbridge kick" command. ... Fixed ref leak in the CLI "confbridge kick" command when the channel to be kicked was not in the conference. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@413451 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	b5b698afbf	app_sms: Fix uninitialized values; hangup channel when REL is sent successfully ... 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 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@412656 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	d4b964c29d	app_stack: Add missing unlock in off-nominal path of STACK_PEEK function. ... 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 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@412226 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	e72594e403	app_confbridge: Fix confbridge.conf dsp_talking_threshold option setting wrong parameter. ... Fixed copy pasta error. ASTERISK-23545 #close Reported by: John Knott git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@411944 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Corey Farrell	6208cfee0d	app_voicemail: fix missing symbol ... ASTERISK-23391 caused a regression where the symbol 'defaultlanguage' was used by app_voicemail but not exported by main/asterisk. This change renames the variable to ast_defaultlanguage. The variable was already renamed in Asterisk 12+. (closes issue ASTERISK-23559) Reported by: Corey Farrell Review: https://reviewboard.asterisk.org/r/3408/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@411633 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Joshua Colp	84a4ba6c68	app_queue: Fix a bug where realtime members would be deleted during reload 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 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@411585 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Corey Farrell	8fe29356ac	Fix dialplan function NULL channel safety issues ... (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 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@411314 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Jonathan Rose	5b0f3c7458	app_confbridge: Fix bug - users with startmuted set don't start muted ... (closes issue ASTERISK-23461) Reported by: Chico Manobela Review: https://reviewboard.asterisk.org/r/3373/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@410965 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	4a3c8065ab	app_confbridge: Make explicitly stop MOH if a user is kicked or hangs up 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/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@410490 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago