4860 Commits (13.24)

Author	SHA1	Message	Date
lvl	91630834f7	app_queue: Revert broken queue channel reference patch ... Revert commit 6409e7b11a, and add NULL checks for all app_queue event handling code. Related issues: ASTERISK~25185, ASTERISK~27006, ASTERISK~25844 ASTERISK-28125 Change-Id: I37334ea184ebb56e54471496b82937d4927815a0	7 years ago
George Joseph	363d19326b	Revert "app_voicemail: Remove need to subscribe to stasis" ... This reverts commit 1843b0e2b5. That commit closed a long standing hole which allowed subscriptions to mailboxes that weren't configured in voicemail.conf. This caused an issue with FreePBX which depdended on that behavior. The commit is being reverted until FreePBX can handle the new behavior. ASTERISK-28151 Reported by: Ronald Raikes Change-Id: I57b7b85e75d7dd97c742b5c69d718a0f61260c15	7 years ago
Corey Farrell	0a9904e1c6	astobj2: Eliminate usage of legacy ao2_container_alloc routine. ... Replace usage of ao2_container_alloc with ao2_container_alloc_hash or ao2_container_alloc_list. ao2_container_alloc is now restricted to modules only and is being removed from Asterisk 17. Change-Id: I0907d78bc66efc775672df37c8faad00f2f6c088	7 years ago
Joshua Colp	d748ed4147	stasis: Add internal filtering of messages. ... This change adds the ability for subscriptions to indicate which message types they are interested in accepting. By doing so the filtering is done before being dispatched to the subscriber, reducing the amount of work that has to be done. This is optional and if a subscriber does not add message types they wish to accept and set the subscription to selective filtering the previous behavior is preserved and they receive all messages. There is also the ability to explicitly force the reception of all messages for cases such as AMI or ARI where a large number of messages are expected that are then generically converted into a different format. ASTERISK-28103 Change-Id: I99bee23895baa0a117985d51683f7963b77aa190	7 years ago
Joshua Colp	dc52719cb9	Merge "app_dial/queue/followme: 'I' options to block initial updates in both directions" into 13	7 years ago
George Joseph	46e3dbf1fd	Merge "app_dial/app_queue: Update application option documentation" into 13	7 years ago
Alexei Gradinari	0e7b91a22b	app_dial/queue/followme: 'I' options to block initial updates in both directions ... The 'I' option currently blocks initial CONNECTEDLINE or REDIRECTING updates from the called parties to the caller. This patch also blocks updates in the other direction before call is answered. ASTERISK-27980 Change-Id: I6ce9e151a2220ce9e95aa66666933cfb9e2a4a01	7 years ago
Richard Mudgett	badf0adf86	app_dial/app_queue: Update application option documentation ... * Update the post-answer documentation and example. The Dial example was incorrect and misleading for the post-answer subroutine useage. * Fix note and warning paragraphs in option descriptions. They don't show up in the wiki. Change-Id: I81019a1fd75d5b9151f76b52c38e2a90da682d14	7 years ago
Corey Farrell	01716f70e6	taskprocessor: Warn on unused result from pushing task. ... Add attribute_warn_unused_result to ast_taskprocessor_push, ast_taskprocessor_push_local and ast_threadpool_push. This will help ensure we perform the necessary cleanup upon failure. Change-Id: I7e4079bd7b21cfe52fb431ea79e41314520c3f6d	7 years ago
Richard Mudgett	ec52409a53	app_queue.c: Fix json ref leak ... Declining the queue_member_status_type stasis message in stasis.conf causes these messages to leak json objects. * Add missing ast_json_unref() if the type is NULL in queue_publish_member_blob(). ASTERISK-28084 Change-Id: I691ecf49bd1f7d9c29182e1eee8c4bb7103be9fc	7 years ago
Cao Minh Hiep	74c5c1cd1b	app_queue: Fix Attended transfer hangup with removing pending member. ... This issue related to setting of holdtime, announcements, member delays. It works well if we set the member delays to "0" and no announcements and no holdtime.This issue will happen if we set member delays to "1", "2"... or announcements or holdtime and hangs up the call during processing it. And here is the reason: (At the step of answering a phone.) It takes care any holdtime, announcements, member delays, or other options after a call has been answered if it exists. Normally, After the call has been aswered, and we wait for the processing one of the cases of the member delays or hold time or announcements finished, "if (ast_check_hangup(peer))" will be not executed, then queue will be updated at update_queue(). Here, pending member will be removed. However, after the call has been aswered, if we hangs up the call during one of the cases of the member delays or hold time or announcements, "if (ast_check_hangup(peer))" will be executed. outgoing = NULL and at hangupcalls, pending members will not be removed. * This fixed patch will remove the pending member from container before hanging up the call with outgoing is NULL. ASTERISK-27920 Reported by: Cao Minh Hiep Tested by: Cao Minh Hiep Change-Id: Ib780fbf48ace9d2d8eaa1270b9d530a4fc14c855	7 years ago
George Joseph	594bbbea57	Merge "app_voicemail: Fix stack overrun in append_mailbox" into 13	7 years ago
George Joseph	656b3e85cf	app_voicemail: Fix stack overrun in append_mailbox ... The append_mailbox function wasn't calculating the correct length to pass to ast_alloca and it wasn't handling the case where context might be empty. Found by the Address Sanitizer. Change-Id: I7eb51c7bd18a7a8dbdba261462a95cc69e84f161	7 years ago
George Joseph	7bdf1d3c67	app_voicemail: Cleanup mailbox topic and cache ... app_voicemail wasn't properly cleaning up the stasis cache or the mwi topic pool when the module was unloaded or when a user was deleted as a result of a reload. This resulted in leaks in both areas. * app_voicemail now calls ast_delete_mwi_state_full when it frees a user structure and ast_delete_mwi_state_full in turn now calls the new stasis_topic_pool_delete_topic function to clear the topic from the pool. Change-Id: Ide23144a4a810e7e0faad5a8e988d15947965df8	7 years ago
George Joseph	1843b0e2b5	app_voicemail: Remove need to subscribe to stasis ... app_voicemail was using the stasis cache to build and maintain a list of mailboxes that had subscribers. It then used this list to determine if a mailbox should be polled for new messages if polling was enabled. For this to work, stasis had to cache every subscription and unsubscription to the mailbox which caused a lot of overhead, both cpu and memory related. Since polling is only required when changes are being made to mailboxes outside of app_voicemail and since the number of mailboxes that don't have any subscribers is likely to be very low, all mailboxes are now polled instead of just the ones with subscribers. This paves the way for disabling the caching of stasis subscription change messages. Also fixed cleanup in some of the unit tests that not only left test users in the users list but also caused segfaults if the tests were run more than once. ASTERISK-27121 Change-Id: I5cceb737246949f9782955c64425b8bd25a9e9ee	7 years ago
Walter Doekes	d226458c5b	optional_api: Remove unused nonoptreq fields ... As they're not actively used, they only grow stale. The moduleinfo field itself is kept in Asterisk 13/15 for ABI compatibility. ASTERISK-28046 #close Change-Id: I8df66a7007f807840414bb348511a8c14c05a9fc	7 years ago
lvl	eda1af091e	app_queue: Update realtime queuemembers after wait_a_bit(), not before ... This ensures the most up-to-date information is used for the next call attempt. ASTERISK-28032 Change-Id: I02fc17c6ffb50bb60ea97c2d2e6023e8061815ce	7 years ago
Rodrigo Ramírez Norambuena	17040d1ce3	app_dial: set the comment for OPT_ARG_ANNOUNCE to really what is done ... Change-Id: I08f88adb09f7e5813f37e70fecd787468cdb32c8	7 years ago
Sean Bright	b9d9c0a8b9	app_queue: Silence GCC 8 compiler warning ... I'm only seeing an error in 14+, so I assume it is due to different compiler options: app_queue.c: In function ‘handle_queue_add_member’: app_queue.c:10234:19: error: ‘%d’ directive writing between 1 and 11 bytes into a region of size 3 [-Werror=format-overflow=] sprintf(num, "%d", state); ^~ app_queue.c:10234:18: note: directive argument in the range [-2147483648, 99] sprintf(num, "%d", state); ^~~~ Compiler: gcc version 8.0.1 20180414 (experimental) [trunk revision 259383] (Ubuntu 8-20180414-1ubuntu2) Change-Id: I18577590da46829c1ea7d8b82e41d69f105baa10	7 years ago
Ivan Poddubny	f48761907a	app_queue: set QUEUESTATUS to LEAVEEMPTY instead of CONTINUE ... When a call leaves a queue on leaveempty condition, QUEUESTATUS must be set to LEAVEEMPTY, no matter whether Queue was executed with or without the "c" (continue) option. The regression was introduced in the fix for ASTERISK_25665. The following fix (ASTERISK_27065) was incomplete, as QUEUESTATUS was overwritten in case when "c" is set, regardless of what was the cause for leaving the queue. ASTERISK-27973 #close Reported-by: Valentin Safonov Change-Id: Iec013fe6a26a4e825ca572a1dda4f3cee5f6f80c	7 years ago
Robert Mordec	447ec4e472	app_confbridge: Bridge and announcers not removed if conference ends quickly ... If a conference is ended very quickly after it was created (i.e., the first user immediately hangs up) then the conference bridge and announcer channels are not removed. When a conference is created, the push_announcer() function is added to the playback queue task processor and the conference object reference is bumped. If a conference is ended while the push_announcer() function is still going then the ao2_cleanup(conference) at the end of push_announcer() will call the destructor function - destroy_conference_bridge(). The destroy_conference_bridge() function will then add the hangup_playback() task to the playback queue and will wait for it to end. Since it is already a current task of the playback queue it will wait forever. This patch makes the conference thread call push_announcer() directly. This way the conference object reference bump is not needed. Since the playback queue task processor is only used by the conference thread itself, there is no danger of trying to play announcements before the announcer is pushed to the bridge. ASTERISK-27870 #close Change-Id: I947a50fb121422d90fd1816d643a54d75185a477	7 years ago
Richard Mudgett	6a1626c265	AMI PlayDTMF Action: Make not compete with channel's media thread. ... There can be one and only one thread handling a channel's media at a time. Otherwise, we don't know which thread is going to handle the media frames. ASTERISK-27625 Change-Id: Ia341f1a6f4d54f2022261abec9021fe5b2eb4905	7 years ago
Sam Wierema	bb0ce22b2b	app_mp3: remove 10 seconds of silence after mp3 playback ... This patch changes the way asterisk polls output from mpg123, instead of waiting for 10 seconds(when playing an http url) it now uses a timeout of one second and iterates 10 times using this same timeout. The main difference is that for every timeout asterisk receives it now checks if mpg123 is still running before poll again. ASTERISK-27752 Change-Id: Ib7df8462e3e380cb328011890ad9270d9e9b4620	7 years ago
George Joseph	499867d006	Merge "app_confbridge: Add talking indicator for ConfBridgeList AMI response" into 13	7 years ago
Joshua Colp	f17d09ae63	Merge "app_meetme: Fix manager event documentation for several events." into 13	7 years ago
George Joseph	db2413b446	app_sendtext: Allow content types other than text/plain ... There was no real reason to limit the conteny type to text/plain other than that's what it was limited to before. Now any text/* content type will be allowed for channel drivers that don't support enhanced messaging and any type will be allowed for channel drivers that do support enhanced messaging. Change-Id: I94a90cfee98b4bc8e22aa5c0b6afb7b862f979d9	7 years ago
William McCall	9ff4779f03	app_confbridge: Add talking indicator for ConfBridgeList AMI response ... When an AMI client connects, it cannot determine if a user was talking prior to a transition in the user speaking state (which would generate a ConfbridgeTalking event). This patch causes app_confbridge to track the talking state and make this state available via ConfBridgeList. ASTERISK-27877 #close Change-Id: I19b5284f34966c3fda94f5b99a7e40e6b89767c6	7 years ago
Richard Mudgett	071232244a	app_meetme: Fix manager event documentation for several events. ... The MeetmeJoin, MeetmeLeave, MeetmeEnd, MeetmeMute, MeetmeTalking, and MeetmeTalkRequest AMI events were documented with sending out a Usernum header when the User header was actually output. * Change the online documentation to match reality. ASTERISK-27873 ASTERISK-25261 Change-Id: I437bc70618d07c183c9624b7069c2fcae7f17a39	7 years ago
Joshua Colp	6dbecc2319	Merge "app_voicemail: Fix data-type mismatch between app_voicemail and database" into 13	7 years ago
Kevin Harwell	835cbbe38c	Merge "app_voicemail: Fix incorrect msg leaving/retrieving an ODBC voicemail" into 13	7 years ago
Nic Colledge	436d17fa50	app_voicemail: Fix incorrect msg leaving/retrieving an ODBC voicemail ... Correct the log warning message shown when ODBC voicemail retrieve_file is called and there is a null value in the category column. A more meaningfull message is now written at debug level. ASTERISK-27853 Change-Id: Ic36e97d5eb070a23a12ba45972f6b53e2182a3f4	7 years ago
Nic Colledge	36f08075da	app_voicemail: Fix data-type mismatch between app_voicemail and database ... Fix data-type mismatch between app_voicemail and database columns exposed by new version of MariaDB ASTERISK-27760 Change-Id: I8543ad480a08c98be78bde1ee870e6e6c84b2c5b	7 years ago
Corey Farrell	d893e57c90	Fix GCC 8 build issues. ... This fixes build warnings found by GCC 8. In some cases format truncation is intentional so the warning is just suppressed. ASTERISK-27824 #close Change-Id: I724f146cbddba8b86619d4c4a9931ee877995c84	7 years ago
Corey Farrell	5dffdf79d1	app_macro: Prevent infinite loop in find_matching_priority. ... Use AST_PBX_MAX_STACK to escape if we recurse 128 times. This will prevent crash if dialplan contains an include loop. Log an error when this occurs, at most one message per call to Macro() so we avoid logger spam. ASTERISK-26570 #close Change-Id: I6c71b76998c31434391b150de055ae9a531e31da	7 years ago
George Joseph	3c2249fd37	app_sendtext: Enhance SendText to support Enhanced Messaging ... SendText now accepts new channel variables that can be used to override the To and From display names and set the Content-Type of a message. Since you can now set Content-Type, other text/* content types are now valid. Change-Id: I648b4574478119f95de09d9f08e9595831b02830	7 years ago
Richard Mudgett	906db6a3ff	app_agent_pool.c: Fix off nominal ref leak. ... Change-Id: Ib427ffc2c802620eaafb08b1c2a17dddd8fb8eb6	7 years ago
Joshua Colp	599f326b41	Merge "BuildSystem: Remove unused dependency on libltdl." into 13	7 years ago
Jenkins2	9e21d04755	Merge "app_dial: Enable early-media video" into 13	7 years ago
Alexander Traud	0f634c1446	BuildSystem: Remove unused dependency on libltdl. ... Asterisk does not need the development package of libltdl, because it does not use any symbol of -lltdl directly. Instead, it uses the runtime package via the shared library -lodbc. On the supported platforms, that shared library declares its dependency on -lltdl correctly, otherwise AST_EXT_LIB_CHECK would have failed. ASTERISK-27745 Change-Id: Icd315809b8e7978203431f3afb66240dd3a040ba	7 years ago
George Joseph	d5af24bb14	app_voicemail: Fix json blob errors ... When app_voicemail calls ast_test_suite_notify with the results of a user keypress, it formats the keypress as '%c'. If the user hung up or some other error occurrs, the result of the keypress is a non printable character. This ultimately causes json_vpack_ex to think it's being passed a non utf-8 string and return an error. * Keypress results passed to ast_test_suite_notify are now checked with isprint() and a '?' is substituted if the check fails. Change-Id: I78ee188916bbac840f3d03f40201b692347ea865	7 years ago
Florian Floimair	69463c612d	app_dial: Enable early-media video ... Certain applications (e.g. door-phone) require that also video is transmitted before a call is accepted. Change-Id: I9842e1dc2f6e1c2c49dc33fe615255007d2f821e	7 years ago
Joshua Colp	f05ac26d4a	Merge "Replace direct checks of option_debug with DEBUG_ATLEAST macro." into 13	7 years ago
Kevin Harwell	4135226f0b	Merge "voicemail: Fixed wrong voicemail message count" into 13	7 years ago
Corey Farrell	b81eadcefc	Replace direct checks of option_debug with DEBUG_ATLEAST macro. ... Checking option_debug directly is incorrect as it ignores file/module specific debug settings. This system-wide change replaces nearly all direct checks for option_debug with the DEBUG_ATLEAST macro. Change-Id: Ic342d4799a945dbc40ac085ac142681094a4ebf0	7 years ago
Sungtae Kim	0597e72e1d	voicemail: Fixed wrong voicemail message count ... Fixed wrong voicemail mailbox reference for Action: VoicemailUsersList. ASTERISK-27703 Change-Id: I99bfec14bd4ae475b0fa1fac5a7992f3e2e8d64a	7 years ago
Michael Cargile	bb973aeceb	apps/app_amd.c: Fixed total time and silence calculations ... Between Asterisk 11 and Asterisk 13 there was a significant increase in the number of AST_FRAME_NULL frames being processed by app_amd.c's main loop. Each AST_FRAME_NULL frame was being counted as 100ms towards the total time and silence. This may have been accurate when app_amd.c was orginally added, but it is not in Asterisk 13. As such the total analysis time and silence calculations were way off effectively breaking app_amd.c * Additional debug messages were added * AST_FRAME_NULL are now ignored ASTERISK-27610 Change-Id: I18aca01af98f87c1e168e6ae0d85c136d1df5ea9	7 years ago
Richard Mudgett	1ff580bb25	app_confbridge: ConfbridgeList event has standard channel shapshot headers. ... * Made the AMI ConfbridgeList action's ConfbridgeList events output all the standard channel snapshot headers instead of a few hand-coded channel snapshot headers. The benefit is that the CallerIDName gets disruptive characters like CR, LF, Tab, and a few others escaped. However, an empty CallerIDName is now output as "<unknown>" instead of "<no name>". ASTERISK-27651 Change-Id: Iaf7d54a9d40194c2db060bc9b4979fab6720d977	7 years ago
Richard Mudgett	0cf7a9e0ca	app_confbridge: Add the Muted header to ConfbridgeJoin AMI event. ... ASTERISK-27651 Change-Id: Idef2ca54d242d1b894efd3fc7b360bc6fd5bdc34	7 years ago
Jenkins2	62d491527e	Merge "app_confbridge: Update dsp_silence_threshold and dsp_talking_threshold docs." into 13	7 years ago
Richard Mudgett	4a337b1a76	app_confbridge: Update dsp_silence_threshold and dsp_talking_threshold docs. ... The dsp_talking_threshold does not represent time in milliseconds. It represents the average magnitude per sample in the audio packets. This is what the DSP uses to determine if a packet is silence or talking/noise. Change-Id: If6f939c100eb92a5ac6c21236559018eeaf58443	7 years ago