46 Commits (a66fa4db24553d6ec6c8978c528081a94b1715a1)

Author	SHA1	Message	Date
Richard Mudgett	99f439dc01	ARI POST DTMF: 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: I4d6a2fe7386ea447ee199003bf8ad681cb30454e	7 years ago
Corey Farrell	17480f6ea4	res_stasis: Reduce RAII_VAR usage. ... In addition to being a micro-optimization (RAII_VAR has overhead), this change improves output of REF_DEBUG. Unfortunately when RAII_VAR calls ao2_cleanup it does so from a generated _dtor_varname function. For example this caused _dtor_app to release a reference instead of __stasis_app_unregister. Change-Id: I4ce67120583a446babf9adeec678b71d37fcd9e5	7 years ago
George Joseph	186ef1a657	stasis/control: Fix possible deadlock with swap channel ... If an error occurs during a bridge impart it's possible that the "bridge_after" callback might try to run before control_swap_channel_in_bridge has been signalled to continue. Since control_swap_channel_in_bridge is holding the control lock and the callback needs it, a deadlock will occur. * control_swap_channel_in_bridge now only holds the control lock while it's actually modifying the control structure and releases it while the bridge impart is running. * bridge_after_cb is now tolerant of impart failures. Change-Id: Ifd239aa93955b3eb475521f61e284fcb0da2c3b3	8 years ago
Richard Mudgett	a036c35903	res_stasis: Handle re-enter stasis bridge with swap channel. ... We lose the fact that there is a swap channel if there is one. We currently wind up rejoining the stasis bridge as a normal join after the swap channel has already been kicked from the bridge. This patch preserves the swap channel so the AMI/ARI events can note that the channel joining the bridge is swapping with another channel. Another benefit to swaqpping in one operation is if there are any channels that get lonely (MOH, bridge playback, and bridge record channels). The lonely channels won't leave before the joining channel has a chance to come back in under stasis if the swap channel is the only reason the lonely channels are staying in the bridge. ASTERISK-25947 #close Reported by: Richard Mudgett ASTERISK-24649 Reported by: John Bigelow ASTERISK-24782 Reported by: John Bigelow Change-Id: If37ea508831d1fed6dbfac2f191c638fc0a850ee	9 years ago
Richard Mudgett	ecf4102d02	res_stasis: Add control ref to playback and recording structs. ... The stasis_app_playback and stasis_app_recording structs need to have a struct stasis_app_control ref. Other threads can get a reference to the playback and recording structs from their respective global container. These other threads can then use the control pointer they contain after the control struct has gone. * Add control ref to stasis_app_playback and stasis_app_recording structs. With the refs added, the control command queue can now have a circular control reference which will cause the control struct to never get released if the control's command queue is not flushed when the channel leaves the Stasis application. Also the command queue needs better protection from adding commands if the control->is_done flag is set. * Flush the control command queue on exit. ASTERISK-25882 #close Change-Id: I3cf1fb59cbe6f50f20d9e35a2c07ac07d7f4320d	9 years ago
Richard Mudgett	a179aba65e	res_stasis: Fix crash on a hanging up channel. ... * Give the struct stasis_app_control ao2 object a ref to the channel held in the object. Now the channel will still be around if a thread needs to post a stasis message instead of crash because the topic was destroyed. * Moved stopping any lingering silence generator out of the struct stasis_app_control destructor and made it a part of exiting the Stasis application. Who knows which thread the destructor will be called under so it cannot affect the channel's silence generator. Not only was the channel unprotected when the silence generator was stopped, stasis may no longer even control the channel. ASTERISK-25882 Change-Id: I21728161b5fe638cef7976fa36a605043a7497e4	9 years ago
Mark Michelson	4a3275abb9	Stasis: Use custom structure when setting variables. ... A recent change to queue channel variable setting to the Stasis control queue caused a regression. When setting channel variables, it is possible to give a NULL channel variable value in order to unset the variable (i.e. remove it from the channel variable list). The change introduced a call to ast_variable_new(), which is not tolerant of NULL channel variable values. This new change switches from using ast_variable to using a custom channel variable struct that is lighter weight and NULL value-tolerant. Change-Id: I784d7beaaa3c036ea936d103e7caf0bb1562162d	9 years ago
Joshua Colp	fcb6c1737d	Merge "Stasis: Use control queue to prevent crash." into 13	9 years ago
Mark Michelson	1003c2eb05	Stasis: Fix potential memory leak of control data. ... When queuing tasks onto the Stasis control queue, you can pass an arbitrary data pointer and a function to free that data. All ARI commands that use the Stasis control queue made the assumption that the destructor function would be called in all paths, whether the task was queued successfully or not. However, this was not correct. If a task was queued onto a control structure that was already completed, the allocated data would not be freed properly. This patch corrects this by making sure that all return paths call the data destructor. Change-Id: Ibf06522094f8e5c4cce652537dc5d7222b1c4fcb	9 years ago
Mark Michelson	eedd77fda0	Stasis: Use control queue to prevent crash. ... A crash occurred when attempting to set a channel variable on a channel that had already been hung up. This is because there is a small window between when a control is grabbed and when the channel variable is set that the channel can be hung up. The fix here is to queue the setting of the channel variable onto the control queue. This way, the manipulation of the channel happens in a thread where it is safe to be done. In this change, I also noticed that the setting of bridge roles on channels was being done outside of the control queue, so I also changed those operations to be done in the control queue. ASTERISK-25709 #close Reported by Mark Michelson Change-Id: I2a0a4d51bce6fba6f1d9954e40935e42f366ea78	9 years ago
Mark Michelson	9015bb4c8c	Resolve race conditions involving Stasis bridges. ... This resolves two observed race conditions. First, a bit of background on what the Stasis application does: 1a Creates a stasis_app_control structure. This structure is linked into a global container and can be looked up using a channel's unique ID. 2a Puts the channel in an event loop. The event loop can exit either because the stasis_app_control structure has been marked done, or because of some other factor, such as a hangup. In the event loop, the stasis_app_control determines if any specific ARI commands need to be run on the channel and will run them from this thread. 3a Checks if the channel is bridged. If the channel is bridged, then ast_bridge_depart() is called since channels that are added to Stasis bridges are always imparted as departable. 4a Unlink the stasis_app_control from the container. When an ARI command is received by Asterisk, the following occurs 1b A thread is spawned to handle the HTTP request 2b The stasis_app_control(s) that corresponds to the channel(s) in the request is/are retrieved. If the stasis_app_control cannot be retrieved, then it is assumed that the channel in question has exited the Stasis app or perhaps was never in Stasis in the first place. 3b A command is queued onto the stasis_app_control, and the channel's event loop thread is signaled to run the command. 4b While most ARI commands do nothing further, some, such as adding or removing channels from a bridge, will block until the command they issued has been completed by the channel's event loop. The first race condition that is solved by this patch involves a crash that can occur due to faulty detection of the channel's bridged status in step 3a. What can happen is that in step 2a, the event loop may run the ast_bridge_impart() function to asynchronously place the channel into a bridge, then immediately exit the event loop because the channel has hung up. In step 3a, we would detect that the channel was not bridged and would not call ast_bridge_depart(). The reason that the channel did not appear to be bridged was that the depart_thread that is spawned by ast_bridge_impart() had not yet started. That is the thread where the channel is marked as being bridged. Since we did not call ast_bridge_depart(), the Stasis application would exit, and then the channel would be destroyed Then the depart_thread would start up and try to manipulate the destroyed channel, causing a crash. The fix for this is to switch from using ast_channel_is_bridged() to checking the NULLity of ast_channel_internal_bridge_channel() to determine if ast_bridge_depart() needs to be called. The channel's internal bridge_channel is set when ast_bridge_impart() is called and is NULLed by the call to ast_bridge_depart(). If the channel's internal bridge_channel is non-NULL, then the channel must have been imparted into the bridge and needs to be departed, even if the actual bridging operation has not yet started. By departing the channel when necessary, the thread that is running the Stasis application will block until the bridge gives the okay that the depart_thread has exited. The second race condition that is solved by this patch involves a leak of HTTP handler threads. The problem was that step 2b would successfully retrieve a stasis_app_control structure. Then step 2a would exit the channel from the event loop due to a hangup. Steps 3a and 4a would execute, and then finally steps 3b and 4b would. The problem is that at step 4b, when attempting to add a channel to a bridge, the thread would block forever since the channel would never execute the queued command since it was finished with the event loop. This meant that the HTTP handling thread would be leaked, along with any references that thread may have owned (in my case, I was seeing bridges leaked). The fix for this is to hone in better on when the channel has exited the event loop. The stasis_app_control structure has an is_done field that is now set at each point where the channel may exit the event loop. If step 2b retrieves a valid stasis_app_control structure but the control is marked as done, then the attempted operation exits immediately since there will be nothing to service the attempted command. ASTERISK-25091 #close Reported by Ilya Trikoz Change-Id: If66265b73b4c9f8f58599124d777fedc54576628	10 years ago
Matthew Jordan	1995baad71	ARI/PJSIP: Add the ability to redirect (transfer) a channel in a Stasis app ... This patch adds a new feature to ARI to redirect a channel to another server, and fixes a few bugs in PJSIP's handling of the Transfer dialplan application/ARI redirect capability. *New Feature* A new operation has been added to the ARI channels resource, redirect. With this, a channel in a Stasis application can be redirected to another endpoint of the same underlying channel technology. *Bug fixes* In the process of writing this new feature, two bugs were fixed in the PJSIP stack: (1) The existing .transfer channel callback had the limitation that it could only transfer channels to a SIP URI, i.e., you had to pass 'PJSIP/sip:foo@my_provider.com' to the dialplan application. While this is still supported, it is somewhat unintuitive - particularly in a world full of endpoints. As such, we now also support specifying the PJSIP endpoint to transfer to. (2) res_pjsip_multihomed was, unfortunately, trying to 'help' a 302 redirect by updating its Contact header. Alas, that resulted in the forwarding destination set by the dialplan application/ARI resource/whatever being rewritten with very incorrect information. Hence, we now don't bother updating an outgoing response if it is a 302. Since this took a looong time to find, some additional debug statements have been added to those modules that update the Contact headers. Review: https://reviewboard.asterisk.org/r/4316/ ASTERISK-24015 #close Reported by: Private Name ASTERISK-24703 #close Reported by: Matt Jordan git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@431717 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
Jonathan Rose	712907eec6	ARI: Fix a crash caused by hanging during playback to a channel in a bridge ... ASTERISK-24147 #close Reported by: Edvin Vidmar Review: https://reviewboard.asterisk.org/r/3908/ ........ Merged revisions 421879 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@421880 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Mark Michelson	5de3fa2c60	Ensure after-bridge behavior is correct when moving from Stasis to a non-Stasis bridge. ... Because of the departable state of channels that enter Stasis bridges, Stasis has to take responsibility for directing the channel to its intended after-bridge destination if the channel moves from a Stasis bridge to a non-Stasis bridge. This change ensures that when such a move occurs, when the channel leaves the bridging system, any after bridge gotos are honored. Review: https://reviewboard.asterisk.org/r/3920 ........ Merged revisions 421792 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@421794 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
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
Joshua Colp	10f4d0f65d	res_stasis: Add progress indications to operations which perform media. ... This change fixes operations which did not account for the fact that they may be executed on channels which have not been answered. These operations will now indicate progress when invoked. ASTERISK-23560 #close ASTERISk-23560 #comment Reported by: Jan Svoboda Review: https://reviewboard.asterisk.org/r/3495/ ........ Merged revisions 413121 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@413122 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Jonathan Rose	b9d7dfcc62	ARI: Make bridges/{bridgeID}/play queue sound files ... Previously multiple play actions against a bridge at one time would cause the sounds to play simultaneously on the bridge. Now if a sound is already playing, the play action will queue playback to occur after the completion of other sounds currently on the queue. (closes issue ASTERISK-22677) Reported by: John Bigelow Review: https://reviewboard.asterisk.org/r/3379/ ........ Merged revisions 412639 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@412641 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Kinsey Moore	5247a0d990	ARI: Ensure managing application receives ChannelEnteredBridge messages ... This fixes an issue where a Stasis application running over ARI and subscribed to ari/events could miss the ChannelEnteredBridge event because it did not subscribe to the new bridge fast enough. To accomplish this, it subscribes the application controlling the channel to the new bridge before adding it to that bridge which required the stasis_app_control structure to maintain a reference to the stasis_app. (closes issue ASTERISK-23295) Review: https://reviewboard.asterisk.org/r/3336/ ........ Merged revisions 410527 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@410528 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Scott Griepentrog	80ef9a21b9	uniqueid: channel linkedid, ami, ari object creation with id's ... Much needed was a way to assign id to objects on creation, and much change was necessary to accomplish it. Channel uniqueids and linkedids are split into separate string and creation time components without breaking linkedid propgation. This allowed the uniqueid to be specified by the user interface - and those values are now carried through to channel creation, adding the assignedids value to every function in the chain including the channel drivers. For local channels, the second channel can be specified or left to default to a ;2 suffix of first. In ARI, bridge, playback, and snoop objects can also be created with a specified uniqueid. Along the way, the args order to allocating channels was fixed in chan_mgcp and chan_gtalk, and linkedid is no longer lost as masquerade occurs. (closes issue ASTERISK-23120) Review: https://reviewboard.asterisk.org/r/3191/ ........ Merged revisions 410157 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@410158 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Kevin Harwell	ce18946de4	ARI: Adding a channel to a bridge while a live recording is active blocks ... Added the ability to have rules that are checked when adding and/or removing channels to/from a bridge. In this case, if a channel is currently recording and someone attempts to add it to a bridge an "is recording" rule is checked, fails, and a 409 conflict is returned. Also command functions now return an integer value that can be descriptive of what kind of problems, if any, occurred before or during execution. (closes issue ASTERISK-22624) Reported by: Joshua Colp Review: https://reviewboard.asterisk.org/r/2947/ ........ Merged revisions 403749 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403750 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
David M. Lee	fc70db3a81	ari: Fix deadlock problem with functions that use autoservice. ... The code for getting channel variables from ARI assumed that you needed to lock the channel in order to properly execute functions and read channel variables. Apparently, this is not the case, since any dialplan function that puts the channel into autoservice deadlocks when attempting to remove the channel from autoservice. ........ Merged revisions 403342 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403403 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
David M. Lee	d1ad4a95f8	ari: Add silence generator controls ... This patch adds the ability to start a silence generator on a channel via ARI. This generator will play silence on the channel (avoiding audio timeouts on the peer) until it is stopped, or some other media operation is started (like playing media, starting music on hold, etc.). (closes issue ASTERISK-22514) Review: https://reviewboard.asterisk.org/r/3019/ ........ Merged revisions 402926 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@402928 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Joshua Colp	67b650543c	res_ari_channels: Add the ability to stop locally generated ringing on a channel. ... Using the 'ring' operation it is possible to start locally generated ringback if the channel is answered. This change adds the ability to stop it by using DELETE. ........ Merged revisions 402804 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@402805 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Joshua Colp	7678fd040e	res_ari_channels: Add ring operation, dtmf operation, hangup reasons, and tweak early media. ... The ring operation sends ringing to the specified channel it is invoked on. The dtmf operation can be used to send DTMF digits to the specified channel of a specific length with a wait time in between. Finally hangup reasons allow you to specify why a channel is being hung up (busy, congestion). Early media behavior has also been tweaked slightly. When playing media to a channel it will no longer automatically answer. If it has not been answered a progress indication is sent instead. (closes issue ASTERISK-22701) Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/2916/ ........ Merged revisions 402358 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@402359 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Joshua Colp	d17a780333	res_stasis: Ensure the channel is always departed from the bridge when it leaves. ... This change adds a command to the command queue to explicitly depart the channel from the bridge when it is told it has left. If the channel has already been departed or has entered a different bridge this command will become a no-op. (closes issue ASTERISK-22703) Reported by: John Bigelow (closes issue ASTERISK-22634) Reported by: Kevin Harwell Review: https://reviewboard.asterisk.org/r/2965/ ........ Merged revisions 402336 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@402337 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Joshua Colp	83d55f85ec	Allow specifying a channel to dial an extension and context in an ARI dial operation. ... (issue ASTERISK-22625) Reported by: Scott Griepentrog ........ Merged revisions 400254 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@400255 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	2a371cd80b	Restore Dial, Queue, and FollowMe 'I' option support. ... The Dial, Queue, and FollowMe applications need to inhibit the bridging initial connected line exchange in order to support the 'I' option. * Replaced the pass_reference flag on ast_bridge_join() with a flags parameter to pass other flags defined by enum ast_bridge_join_flags. * Replaced the independent flag on ast_bridge_impart() with a flags parameter to pass other flags defined by enum ast_bridge_impart_flags. * Since the Dial, Queue, and FollowMe applications are now the only callers of ast_bridge_call() and ast_bridge_call_with_flags(), changed the calling contract to require the initial COLP exchange to already have been done by the caller. * Made all callers of ast_bridge_impart() check the return value. It is important. As a precaution, I also made the compiler complain now if it is not checked. * Did some cleanup in parking_tests.c as a result of checking the ast_bridge_impart() return value. An independent, but associated change is: * Reduce stack usage in ast_indicate_data() and add a dropping redundant connected line verbose message. (closes issue ASTERISK-22072) Reported by: Joshua Colp Review: https://reviewboard.asterisk.org/r/2845/ ........ Merged revisions 399136 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@399138 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
David M. Lee	7a581537e8	ARI: Correct error codes for bridge operations ... This patch adds error checking to ARI bridge operations, when adding/removing channels to/from bridges. In general, the error codes fall out as follows: * Bridge not found - 404 Not Found * Bridge not in Stasis - 409 Conflict * Channel not found - 400 Bad Request * Channel not in Stasis - 422 Unprocessable Entity * Channel not in this bridge (on remove) - 422 Unprocessable Entity (closes issue ASTERISK-22036) Review: https://reviewboard.asterisk.org/r/2769/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397565 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
David M. Lee	987fdfb444	ARI: allow other operations to happen while bridged ... This patch changes ARI bridging to allow other channel operations to happen while the channel is bridged. ARI channel operations are designed to queue up and execute sequentially. This meant, though, that while a channel was bridged, any other channel operations would queue up and execute only after the channel left the bridge. This patch changes ARI bridging so that channel commands can execute while the channel is bridged. For most operations, things simply work as expected. The one thing that ended up being a bit odd is recording. The current recording implementation will fail when one attempts to record a channel that's in a bridge. Note that the bridge itself may be recording; it's recording a specific channel in the bridge that fails. While this is an annoying limitation, channel recording is still very useful for use cases such as voice mail, and bridge recording makes up much of the difference for other use cases. (closes issue ASTERISK-22084) Review: https://reviewboard.asterisk.org/r/2726/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396568 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Jonathan Rose	e47794ead1	ARI: bridges/{bridgeID}/addChannel: add roles parameter ... Roles are now cleared with each entry into a bridge with addChannel. If the roles parameter is present, the role specified will be applied to all channels being added with the addChannel command. (closes issue ASTERISK-21973) Reported by: Matt Jordan https://reviewboard.asterisk.org/r/2691/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396182 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	38236e54a8	Remove dead code from features.c; refactor pickup code into pickup.c ... This patch does the following: * It moves the pickup code out of features.c and into pickup.c * It removes the vast majority of dead code out of features.c. In particular, this includes the parking code. (issue ASTERISK-22134) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396060 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
David M. Lee	e1b959ccbb	Split caching out from the stasis_caching_topic. ... In working with res_stasis, I discovered a significant limitation to the current structure of stasis_caching_topics: you cannot subscribe to cache updates for a single channel/bridge/endpoint/etc. To address this, this patch splits the cache away from the stasis_caching_topic, making it a first class object. The stasis_cache object is shared amongst individual stasis_caching_topics that are created per channel/endpoint/etc. These are still forwarded to global whatever_all_cached topics, so their use from most of the code does not change. In making these changes, I noticed that we frequently used a similar pattern for bridges, endpoints and channels: single_topic ----------------> all_topic ^ | single_topic_cached ----+----> all_topic_cached | +----> cache This pattern was extracted as the 'Stasis Caching Pattern', defined in stasis_caching_pattern.h. This avoids a lot of duplicate code between the different domain objects. Since the cache is now disassociated from its upstream caching topics, this also necessitated a change to how the 'guaranteed' flag worked for retrieving from a cache. The code for handling the caching guarantee was extracted into a 'stasis_topic_wait' function, which works for any stasis_topic. (closes issue ASTERISK-22002) Review: https://reviewboard.asterisk.org/r/2672/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@395954 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	cafc115896	A great big renaming patch ... This patch renames the bridging* files to bridge*. This may seem pedantic and silly, but it fits better in line with current Asterisk naming conventions: * channel is not "channeling" * monitor is not "monitoring" etc. A bridge is an object. It is a first class citizen in Asterisk. "Bridging" is the act of using a bridge on a set of channels - and the API that fulfills that role is more than just the action. (closes issue ASTERISK-22130) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@395378 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Jonathan Rose	a6329a3acf	ARI: MOH start and stop for a channel ... (issue ASTERISK-21974) Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/2680/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@394810 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Jonathan Rose	17c546173f	ARI: Bridge Playback, Bridge Record ... Adds a new channel driver for creating channels for specific purposes in bridges, primarily to act as either recorders or announcers. Adds ARI commands for playing announcements to ever participant in a bridge as well as for recording a bridge. This patch also includes some documentation/reponse fixes to related ARI models such as playback controls. (closes issue ASTERISK-21592) Reported by: Matt Jordan (closes issue ASTERISK-21593) Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/2670/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@394809 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Jason Parker	c1a7567d24	ARI: Add support for suppressing media streams. ... Also convert res_mutestream to use the core feature behind this. (closes issue ASTERISK-21618) Review: https://reviewboard.asterisk.org/r/2652/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@394715 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Joshua Colp	7ee5b025f4	Fix building. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393807 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Jason Parker	87973eecff	ARI: Add support for getting/setting channel and global variables. ... This allows for reading and writing of functions on channels. (closes issue ASTERISK-21868) Review: https://reviewboard.asterisk.org/r/2641/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393806 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Jason Parker	f820d24db1	ARI: Implement channel hold/unhold. ... This puts the channel on hold (rather than queueing a frame from the channel). (closes issue ASTERISK-21619) Review: https://reviewboard.asterisk.org/r/2647/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393332 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Jason Parker	f41faf0b7d	ARI: Implement channel dial. ... This creates a new outbound channel, and bridges it to a channel already in the Stasis application. (closes issue ASTERISK-21620) Review: https://reviewboard.asterisk.org/r/2634/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393326 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Jason Parker	609c42c854	ARI: Add support for continuing to a different location in dialplan. ... This allows going elsewhere in the dialplan, so that the location can be specified after exiting the Stasis application. (closes issue ASTERISK-21870) Review: https://reviewboard.asterisk.org/r/2644/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@392987 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kinsey Moore	a5bbc790e7	Stasis-HTTP: Flesh out bridge-related capabilities ... This adds support for Stasis applications to receive bridge-related messages when the application shows interest in a given bridge. To supplement this work and test it, this also adds support for the following bridge-related Stasis-HTTP functionality: * GET stasis/bridges * GET stasis/bridges/{bridgeId} * POST stasis/bridges * DELETE stasis/bridges/{bridgeId} * POST stasis/bridges/{bridgeId}/addChannel * POST stasis/bridges/{bridgeId}/removeChannel Review: https://reviewboard.asterisk.org/r/2572/ (closes issue ASTERISK-21711) (closes issue ASTERISK-21621) (closes issue ASTERISK-21622) (closes issue ASTERISK-21623) (closes issue ASTERISK-21624) (closes issue ASTERISK-21625) (closes issue ASTERISK-21626) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@391199 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
David M. Lee	557125664d	This patch adds support for controlling a playback operation from the ... Asterisk REST interface. This adds the /playback/{playbackId}/control resource, which may be POSTed to to pause, unpause, reverse, forward or restart the media playback. Attempts to control a playback that is not currently playing will either return a 404 Not Found (because the playback object no longer exists) or a 409 Conflict (because the playback object is still in the queue to be played). This patch also adds skipms and offsetms parameters to the /channels/{channelId}/play resource. (closes issue ASTERISK-21587) Review: https://reviewboard.asterisk.org/r/2559 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@389603 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
David M. Lee	10ba6bf8a8	This patch implements the REST API's for POST /channels/{channelId}/play ... and GET /playback/{playbackId}. This allows an external application to initiate playback of a sound on a channel while the channel is in the Stasis application. /play commands are issued asynchronously, and return immediately with the URL of the associated /playback resource. Playback commands queue up, playing in succession. The /playback resource shows the state of a playback operation as enqueued, playing or complete. (Although the operation will only be in the 'complete' state for a very short time, since it is almost immediately freed up). (closes issue ASTERISK-21283) (closes issue ASTERISK-21586) Review: https://reviewboard.asterisk.org/r/2531/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@389587 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
David M. Lee	e8f4ac6c61	Break res_stasis into smaller files. ... When implementing playback for stasis-http, the monolithicedness of res_stasis really started to get in my way. This patch breaks the major components of res_stasis.c into individual files. * res/stasis/app.c - Stasis application tracking * res/stasis/control.c - Channel control objects * res/stasis/command.c - Channel command object This refactoring also allows res_stasis applications to be loaded as independent modules, such as the new res_stasis_answer module. The bulk of this patch is simply moving code from one file to another, adjusting names and adding accessors as necessary. Review: https://reviewboard.asterisk.org/r/2530/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@388729 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago