4739 Commits (50c3bb2631edc5063833d51a5d1091d5344d1b77)

Author	SHA1	Message	Date
zuul	be42630f5b	Merge "ConfBridge: Make some announcements asynchronous."	9 years ago
zuul	cc7e978149	Merge "apps/app_dial: Fix crash on non-connect call paths for Privacy/Screening option"	9 years ago
zuul	c6a8710ceb	Merge "apps/app_dial: Set the DIALSTATUS to NOANSWER on privacy option 5"	9 years ago
Mark Michelson	ac02bbd9a0	ConfBridge: Make some announcements asynchronous. ... Confbridge announcements tend to block a channel while they are being played. In some circumstances, this is warranted since you want that particular channel not to hear the announcement (Example: "John Doe has entered the conference"). For others it makes less sense. This change first introduces methods for playing sounds asynchronously into the conference. This is very similar to how synchronous sounds are played, except the channel initiating the playback does not wait for the sound to complete before moving on. Asynchronous announcements are used for two circumstances: * Sounds played for a user after they have left the bridge * Sounds that play first to a single user and then the rest of the conference (if the channel and conference use the same language) ASTERISK-26289 #close Reported by Mark Michelson Change-Id: Ie486bb3de1646d50894489030326a423e594ab0a	9 years ago
Matt Jordan	730cb3b0b7	apps/app_dial: Fix crash on non-connect call paths for Privacy/Screening option ... In any scenario in which the callee is not connected to the caller, the current code in app_dial will crash due to raising a Dial End Stasis Message after the callee channel has been hung up. This patch corrects the error by simply moving the explicit hangup of the callee (peer) channel until after the dial end message. ASTERISK-25691 #close Change-Id: I816a414014424d0d8c80e2a3cbef13ef8c63798d	9 years ago
Matt Jordan	6e1a3b924e	apps/app_dial: Set the DIALSTATUS to NOANSWER on privacy option 5 ... If the callee selects option '5' using the Dial application's privacy (P) option, the DIALSTATUS is erroneously set to ANSWER. This option reflects the callee sending the caller to VoiceMail one time; the call is definitely *not* ANSWERed in such a scenario. With this patch, the DIALSTATUS is instead set to NOANSWER, which is the same DIALSTATUS that is set when the 'send to VoiceMail every time' option is set. ASTERISK-25691 Change-Id: Iaf0c9f0fa00545e7366443875e2bb7d9a89a1358	9 years ago
Michael Kuron	48fd4c815c	app_mp3: Use correct buffer size and the same sample rate as the channel ... Previously, the buffer used for MP3 streamed from HTTP servers had a size of 1 MB. For 8 kHz mono audio at 16 bit resolution, such a buffer covers about 1 minute. Only when the buffer is full does audio start to play. For MP3 files streamed from a server, that is usually not a big deal as long as the connection to the server is fast enough to supply that much data within a second or two. For MP3 live streams however, it takes 1 minute to download 1 minute of audio, so without this change, app_mp3 wasn't really usable for MP3 live streams. This commit changes the buffer size so that it covers 6 seconds of an MP3 file streamed from a server and 0.5 seconds of an MP3 live stream. The latter is identified by the use of a .m3u file extension. app_mp3 so far only supported 8 kHz audio. Now it always runs at the sample rate of the channel. ASTERISK-26085 #close Change-Id: Id1ee274733cd804a0edecf7450329b72f1235af0	9 years ago
zuul	8bdd5b63df	Merge "app_queue: Ensure member is removed from pending when hanging up."	9 years ago
Joshua Colp	c21e6764f1	app_queue: Ensure member is removed from pending when hanging up. ... When dialing channels it is possible that they may not ever leave the not in use state (Local channels in particular) by the time we cancel them. If this occurs but we know they were dialed we explicitly remove them from the pending members container so that subsequent call attempts occur. ASTERISK-26299 #close Change-Id: I6ad0d17c36480c92cebf840626228ce3f7e4bd65	9 years ago
chrisderock	93b7533d74	app_macro: Consider '~~s~~' as a macro start extension. ... As described in issue ASTERISK-26282 the AEL parser creates macros with extension '~~s~~'. app_macro searches only for extension 's' so the created extension cannot be found. with this patch app_macro searches for both extensions and performs the right extension. ASTERISK-26282 #close Change-Id: I939aa2a694148cc1054dd75ec0c47c47f47c90fb	9 years ago
Mark Michelson	ded22c712a	ConfBridge: Rework announcer channel methodology ... NOTE: This patch was submitted earlier and reverted because of a failing test. The test has been patched so that it adjusts for the changes here, so this is being resubmitted for review. One feature that confbridge has is the ability to play sounds to all participants in the conference. Prior to this commit, the algorithm for this was as follows: * Grab the playback lock * Push the conference announcer channel into the bridge * Play back the sound * Pull the conference announcer channel from the bridge * Release the playback lock The issue here is that the act of adding the playback channel to the bridge and removing it for each announcement is expensive. Amongst the expenses: * The announcer channel is imparted into the bridge, meaning a new thread is spun up for each playback. * When the announcer is added or removed from the bridge, it results in the BRIDGEPEER channel variable being set on all channels in the bridge. This requires keeping the bridge locked and locking each individual channel in order to set it. * There's also just the general overhead of adding the channel and removing it from the bridge. The bridge potentially has to reconfigure every single time With this commit, the paradigm for playing back announcements has shifted. * The announcer channel is now added to the bridge when the conference is allocated, and it is hung up when the conference is destroyed. * A taskprocessor is used to queue playbacks onto the announcer channel. This keeps the behavior from before where playbacks do not overlap. * The announcer channel is no longer placed into the bridge as departable. Since we are not constantly removing the channel from the bridge, it is safe to add the channel using an independent thread and simply hang the channel up when it is time for the conference to be destroyed. The use of the taskprocessor for playbacks opens up the interesting possibility of having asynchronous announcements played. In this commit, however, the behavior is still exactly the same as it previously was. ASTERISK-26289 Reported by Mark Michelson Change-Id: Ica9fa4907c2f3728cdd1cf0bc564ef4eb40754a0	9 years ago
Joshua Colp	11ef7f34bf	Merge "Revert "ConfBridge: Rework announcer channel methodology""	9 years ago
Joshua Colp	065d810d3f	Revert "ConfBridge: Rework announcer channel methodology" ... This reverts commit 5aa8773052. Change-Id: I9ab45776e54a54ecf1bac9ae62d976dec30ef491	9 years ago
zuul	c9df806f24	Merge "ConfBridge: Rework announcer channel methodology"	9 years ago
Mark Michelson	5aa8773052	ConfBridge: Rework announcer channel methodology ... One feature that confbridge has is the ability to play sounds to all participants in the conference. Prior to this commit, the algorithm for this was as follows: * Grab the playback lock * Push the conference announcer channel into the bridge * Play back the sound * Pull the conference announcer channel from the bridge * Release the playback lock The issue here is that the act of adding the playback channel to the bridge and removing it for each announcement is expensive. Amongst the expenses: * The announcer channel is imparted into the bridge, meaning a new thread is spun up for each playback. * When the announcer is added or removed from the bridge, it results in the BRIDGEPEER channel variable being set on all channels in the bridge. This requires keeping the bridge locked and locking each individual channel in order to set it. * There's also just the general overhead of adding the channel and removing it from the bridge. The bridge potentially has to reconfigure every single time With this commit, the paradigm for playing back announcements has shifted. * The announcer channel is now added to the bridge when the conference is allocated, and it is hung up when the conference is destroyed. * A taskprocessor is used to queue playbacks onto the announcer channel. This keeps the behavior from before where playbacks do not overlap. * The announcer channel is no longer placed into the bridge as departable. Since we are not constantly removing the channel from the bridge, it is safe to add the channel using an independent thread and simply hang the channel up when it is time for the conference to be destroyed. The use of the taskprocessor for playbacks opens up the interesting possibility of having asynchronous announcements played. In this commit, however, the behavior is still exactly the same as it previously was. ASTERISK-26289 Reported by Mark Michelson Change-Id: Ic5cd2c4b98a1eaa1715eb7a5b35d62f1a76d78a5	9 years ago
Tzafrir Cohen	046069011b	followme: initialize all config items on reload ... Some configuration directives were not initialized on reload, and hence were not reset to default if they were removed from followme.conf. ASTERISK-26288 #close Change-Id: Ief829e16374ad1e0ecfd63e6ee4923b5a1d1c150	9 years ago
zuul	3117d150fa	Merge "manager: Add <see-also> tags to relate UserEvent actions/apps/events"	9 years ago
Matt Jordan	9202ca34a8	app_dial: Improve documentation ... * Add some helpful <literal> and other embedded paragraph tags * Document some of the lesser known channel variables set by Dial * Add examples for some common Dial uses, along with some more challenging but useful options Change-Id: Ib2fb9301e8e044d14fbb2815ec64161f19bbfbc1	9 years ago
Matt Jordan	243f0cf99a	manager: Add <see-also> tags to relate UserEvent actions/apps/events ... Change-Id: I80f8a981f62f50e74609c69c49edcaca6c95efa4	9 years ago
Matt Jordan	225fd1003f	app_queue: Prevent crash when a call is forwarded to an invalid location ... When a call forward attempt is made from a Queue member, the current code will hang up the forwarding channel in an off-nominal condition prior to raising the Stasis events informing the rest of Asterisk that the call was forwarded. This will result in a slew of dreaded FRACKs, most likely leading to a crash. This patch modifies the code such that we don't hang up the forwarding channel even in an off-nominal condition until we've safely raised the Stasis messages. ASTERISK-25797 #close Change-Id: Ife5abed351691fd79105321636eaa8ea8dcdba38	9 years ago
Alexei Gradinari	9042ad40f2	app_voicemail: Add taskprocessor alert level options. ... On heavy loaded system with IMAP or DB storage, 'app_voicemail' taskprocessor queue could reach 500 scheduled tasks. It could happen when the IMAP or DB server dies or is unreachable. It could happen on startup when there are many (thousands) realtime endpoints configured with unsolicited mwi. If the taskprocessor queue reaches the high water level then the alert is triggered and pjsip stops processing new requests until the queue reaches the low water level to clear the alert. This patch adds 2 new 'general' configuration options to tune taskprocessor alert levels: 'tps_queue_high' - Taskprocessor high water alert trigger level. 'tps_queue_low' - Taskprocessor low water clear alert level ASTERISK-26229 #close Change-Id: I766294fbffedf64053c0d9ac0bedd3109f043ee8	9 years ago
Corey Farrell	cf1188a1be	Unit tests: Use AST_TEST_DEFINE in conditional code only. ... If AST_TEST_DEFINE is not conditional to TEST_FRAMEWORK it produces dead code. This places all existing unit tests into a conditional block if they weren't already. ASTERISK-26211 #close Change-Id: I8ef83ee11cbc991b07b7a37ecb41433e8c734686	9 years ago
Joshua Colp	943bb48b59	Merge "pbx: Create pbx_include.c for management of 'struct ast_include'."	9 years ago
Corey Farrell	be36bd7ca5	pbx: Create pbx_include.c for management of 'struct ast_include'. ... This changes context includes from a linked list to a vector, makes 'struct ast_include' opaque to pbx.c. Although ast_walk_context_includes is maintained the procedure is no longer efficient except for the first call (inc==NULL). This functionality is replaced by two new functions implemented by vector macros. * ast_context_includes_count (AST_VECTOR_SIZE) * ast_context_includes_get (AST_VECTOR_GET) As with ast_walk_context_includes callers of these functions are expected to have locked contexts. Only a few places in Asterisk walked the includes, they have been converted to use the new functions. const have been applied where possible to parameters for ast_include functions. Change-Id: Ib5c882e27cf96fb2aec67a39c18b4c71c9c83b60	9 years ago
Joshua Colp	31967dacdf	app_queue: Only remove queue member from pending when state changes. ... It is possible for a not in use state change to occur multiple times causing a queue member to be removed from the pending call container prematurely. The first not in use state change will remove the queue member from the container. At this moment the member may be called and placed in the pending container. After this another not in use state change can be received which will remove it from the container. Despite being called at this point the code will incorrectly see that there are no pending calls to it. This change only removes it from the pending container if the state has actually changed. ASTERISK-26133 #close patches: app_queue.diff submitted by Richard Miller (license 5685) Change-Id: Ie5a7f17a44f98e9159e9b85009ce3f8393aa78c0	9 years ago
Richard Mudgett	0a30008224	app_voicemail.c: Fix IMAP compile error. ... Fix compile error introduced by the patch for ASTERISK-26045 Change-Id: I5b02876266f2824f4cec2b54d6ff4db5de5778d3	9 years ago
Joshua Colp	5c949d009e	Merge "Fixes to include signal.h"	9 years ago
Timo Teräs	39b69ab537	Fixes to include signal.h ... POSIX defines signal.h. sys/signal.h should not be used as it is c-library internal header which may or may not exist. Notably with musl it generates warning of being incorrect. Change-Id: Ia56b0aa1d84b5c590114867b1b384a624f39a6fc	9 years ago
Örn Arnarson	60caebc738	apps/app_voicemail.c and main/say.c: Add support for Icelandic language ... Icelandic has some weird grammar rules when dealing with dates and numbers. There are different genders used depending on which number you're dealing with, and only a handful of numbers do change depending on the gender. There is also an implied gender in several cases. This patch was originally written for asterisk 1.6, and has been in use for several years without crashes. I cleaned it up a bit and rewrote what was necessary for Asterisk 13. The functions were copied from other similar languages and modified where appropriate. If i recall correctly, the German and Danish functions were used as a base. ASTERISK-26087 Reported by: Örn Arnarson Tested by: Örn Arnarson Change-Id: Ib7d8bd7b0fede5767921ed821315b5b508c0e665	9 years ago
Alexei Gradinari	3e8d523d88	core/dial: New channel variable FORWARDERNAME ... Added a new channel variable FORWARDERNAME which indicates which channel was responsible for a forwarding requests received on dial attempt. Fixed a bug in the app_queue: FORWARD_CONTEXT is not used. ASTERISK-26059 #close Change-Id: I34e93e8c1b5e17776a77b319703c48c8ca48e7b2	9 years ago
Joshua Colp	608e0267e8	Merge "Expand the scope of Dial Events"	9 years ago
Joshua Colp	ad31e5bb1c	Merge "followme: allow disabling callee prompt"	9 years ago
Mark Michelson	205a31f86c	Expand the scope of Dial Events ... Dial events up to this point have come in two flavors * A Dial event with no status to indicate that dialing has begun * A Dial event with a status to indicate that dialing has ended With this change, Dial events have been expanded to also give intermediate events, such as "RINGING", "PROCEEDING", and "PROGRESS". This is especially useful for ARI dialing, as it gives the application writer the opportunity to place a channel into an early bridge when early media is detected. AMI handles these in-progress dial events by sending a new event called "DialState" that simply indicates that dial state has changed but has not ended. ARI never distinguished between DialBegin and DialEnd, so no change was made to the event itself. Another change here relates to dial forwards. A forward-related event was previously only sent when a channel was successfully able to forward a call to a new channel. With this set of changes, if forwarding is blocked, we send a Dial event with a forwarding destination but no forwarding channel, since we were prevented from creating one. This is again useful for ARI since application writers can now handle call forward attempts from within their own application. ASTERISK-25925 #close Reported by Mark Michelson Change-Id: I42cbec7730d84640a434d143a0d172a740995543	9 years ago
Alexei Gradinari	b3142e99e4	app_voicemail: fix bugs, imap mm_status log change to debug ... Fixed some bugs: - create dirpath when save downloading message from IMAP storage. - create IMAP folder if not exists when saving to IMAP storage - check if file successfully opened before write to it - some IMAP checks - remove non-standard flag 'Unseen' etc Change to debug IMAP mm_status log instead of verbose. Remove unused X-Asterisk-VM-Caller-channel message header for security reason. The clients should not know name of peer/endpoint. ASTERISK-26045 #close Change-Id: I7f83d88b69b36934e2539c114b9fb612deed971b	9 years ago
Tzafrir Cohen	1d60bfcdf1	followme: allow disabling callee prompt ... Add the option 'enable_callee_prompt' to followme.conf. Enabled by default. If disabled, a callee is not prompted to accept or reject the forwarded call. ASTERISK-26064 #close Change-Id: I0a8b19d4cf95c86a07c992813babb9e4a4acfff5 Signed-off-by: Tzafrir Cohen <tzafrir.cohen@xorcom.com>	9 years ago
Tzafrir Cohen	b5c471b339	followme: delete the right recorded name file ... FollowMe with the option a records the name of the caller and plays it to the callee. However it has failed to clean up that recorded file as it tried to delete the file name without the '.sln' extension. ASTERISK-26008 #close Change-Id: I79d7b1be7d5cde57bf076d9389e2a8a4422776ec Signed-off-by: Tzafrir Cohen <tzafrir.cohen@xorcom.com>	9 years ago
Joshua Colp	b4e10e7c90	Merge "app_confbridge: Add a regcontext option for confbridge bridge profiles."	9 years ago
Jaco Kroon	8923c9ac96	app_confbridge: Add a regcontext option for confbridge bridge profiles. ... This patch allows for having app_confbridge register the name of the conference as an extension into a specific context, similar to regcontext for chan_sip. This variant is not quite as involved as the one in chan_sip and doesn't allow for multiple contexts or custom extensions, you can only specify the context and the conference name will always be used as the extension to register. ASTERISK-25989 #close Change-Id: Icacf94d9f2b5dfd31ef36f6cb702392619a7902f	9 years ago
Joshua Colp	122895f864	Merge "app_chanspy: fix audiohook options in non read-only mode"	9 years ago
Jean Aunis	0c9faaee47	app_chanspy: fix audiohook options in non read-only mode ... When option 'o' was not set, ChanSpy created its audiohook with the flag AST_AUDIOHOOK_MUTE_WRITE, which caused ChanSpy to listen audio from one direction only. ASTERISK-25866 #close Change-Id: I5c745855eea29a3fbc4e4aed0b0c0f53580535e0	9 years ago
Andrew Nagy	080c6216b6	app_voicemail: always copy dynamic struct to avoid race condition ... Voicemail email addresses can be corrupt or voicemail emails can end up being sent to the wrong email address if asterisk is reading voicemail.conf during a reload and processing an email at the same time. This patch always copies the struct that would otherwise only be copied once. ASTERISK-24463 #close Reported by: John Campbell Tested by: Etienne Lessard Tested by: Andrew Nagy Change-Id: I3a0643813116da84e2617291903d0d489b7425fb	9 years ago
Jean Aunis	7281770710	app_chanspy: reduce audio loss on the spying channel. ... ChanSpy was creating its audiohook with the flags AST_AUDIOHOOK_TRIGGER_SYNC and AST_AUDIOHOOK_SMALL_QUEUE, which caused audio frames to be lost when queues grow too large or when read and write queues go out of sync. Now these flags are set conditionally: - AST_AUDIOHOOK_TRIGGER_SYNC is not set if the option "o" is set - a new option "l" is created: if set, AST_AUDIOHOOK_SMALL_QUEUE will not be set on the audiohook ASTERISK-25866 Change-Id: I9c7652f41d9fa72c8691e4e70ec4fd16b047a4dd	9 years ago
Joshua Colp	8ae69cffef	app_queue: Fix crash when unloading module. ... When unloading the app_queue module the members in each queue are destroyed and as part of this they are removed from the pending members container. Unfortunately a crash would occur as the container was destroyed before the members were removed. This change tweaks ordering so the container destruction occurs after the members are destroyed. ASTERISK-16115 Change-Id: I48c728668c55aee3d05b751a5d450fb57e87f44b	9 years ago
zuul	811e24f595	Merge "Bridge system: Fix memory leaks and double frees on impart failure."	9 years ago
Joshua Colp	456600a641	Merge "app_queue: queue members can receive multiple calls"	9 years ago
DarkS	f99ec857c8	Fix case sensitive actions in AMI QueueSummary and QueueStatus ... ASTERISK-25954 #close Reported by: Javier Acosta Change-Id: I00be83d45cc7e8385de2523012bd196aafeeb256 (cherry picked from commit c0688a6398)	9 years ago
Kevin Harwell	30ab21d5fa	app_queue: queue members can receive multiple calls ... It was possible for a queue member that is a member of at least 2 or more queues to receive mulitiple calls at the same time. This happened because of a race between when a member was being rung and when the device state notified the other queue(s) member object of the state change. This patch makes it so when a queue member is being rung it gets added to a global pool of queue members. If that same member is tried again, e.g. from another queue, and it is found to already exist in the pending member container then it will not ring that member. ASTERISK-16115 #close Change-Id: I546dd474776d158c2b6be44205353dee5bac7e48	9 years ago
Richard Mudgett	a63656b419	Bridge system: Fix memory leaks and double frees on impart failure. ... You cannot reference the passed in features struct after calling ast_bridge_impart(). Even if the call fails. Change-Id: I902b88ba0d5d39520e670fb635078a367268ea21	9 years ago
Joshua Colp	d95512a7dd	app_talkdetect: Make the module core supported. ... This module is used as part of testsuite tests to confirm stuff works. I'm accordingly marking it as core as it is required by those tests. Change-Id: I558e7af7679b22b8ed641d7dd37ee4ca35b11e88	9 years ago
ibercom	dbb47e0a47	app_queue: Frequent segfaults in function can_ring_entry() ... ASTERISK-25888 #close Change-Id: I007a2f2dd99823e04fb5be3ff01f02b0a2956117	9 years ago