77 Commits (398e5ec16c5b919e17708e26d2e045c389afbd1c)

Author	SHA1	Message	Date
Sean Bright	ea8a610776	cli: Fix various CLI documentation and completion issues ... * app_minivm: Use built-in completion facilities to complete optional arguments. * app_voicemail: Use built-in completion facilities to complete optional arguments. * app_confbridge: Add missing colons after 'Usage' text. * chan_alsa: Use built-in completion facilities to complete optional arguments. * chan_sip: Use built-in completion facilities to complete optional arguments. Add completions for 'load' for 'sip show user', 'sip show peer', and 'sip qualify peer.' * chan_skinny: Correct and extend completions for 'skinny reset' and 'skinny show line.' * func_odbc: Correct completions for 'odbc read' and 'odbc write' * main/asterisk: Correct and extend completions for 'core show file version.' * main/astmm: Use built-in completion facilities to complete arguments for 'memory' commands. * main/bridge: Correct completions for 'bridge kick.' * main/ccss: Use built-in completion facilities to complete arguments for 'cc cancel' command. * main/cli: Add 'all' completion for 'channel request hangup.' Correct completions for 'core set debug channel.' Correct completions for 'core show calls.' * main/pbx_app: Remove redundant completions for 'core show applications.' * main/pbx_hangup_handler: Remove unused completions for 'core show hanguphandlers all.' * res_sorcery_memory_cache: Add completion for 'reload' argument of 'sorcery memory cache stale' and properly implement. Change-Id: Iee58c7392f6fec34ad9d596109117af87697bbca	8 years ago
Matt Jordan	d23b4af477	res/ari/resource_bridges: Add the ability to manipulate the video source ... In multi-party bridges, Asterisk currently supports two video modes: * Follow the talker, in which the speaker with the most energy is shown to all participants but the speaker, and the speaker sees the previous video source * Explicitly set video sources, in which all participants see a locked video source Prior to this patch, ARI had no ability to manipulate the video source. This isn't important for two-party bridges, in which Asterisk merely relays the video between the participants. However, in a multi-party bridge, it can be advantageous to allow an external application to manipulate the video source. This patch provides two new routes to accomplish this: (1) setVideoSource: POST /bridges/{bridgeId}/videoSource/{channelId} Sets a video source to an explicit channel (2) clearVideoSource: DELETE /bridges/{bridgeId}/videoSource Removes any explicit video source, and sets the video mode to talk detection ASTERISK-26595 #close Change-Id: I98e455d5bffc08ea5e8d6b84ccaf063c714e6621	9 years ago
Matt Jordan	7c824b955d	main/bridge: Add some verbose logging for video source changes ... It's actually quite useful to see the source of a video stream change. This doesn't happen terribly often, even with talk detection - but when it does, it's nice to know which channel is now providing your video stream. As a verbose 5 level message, it shouldn't be terribly spammy or costly to have, and is 'lower level' then most other verbose messages that the bridge system emits. ASTERISK-26555 Change-Id: Ia1c20ecafa9670171fd38bddcf3beccae47fb15c	9 years ago
Matt Jordan	232d4fe24f	manager: Add <see-also> tags to relate Bridge related events,actions, and apps ... Change-Id: I67e6b79fa3102e494b5fe6cc7510472249080e85	9 years ago
Joshua Colp	4efc6b4315	Merge changes from topic 'system_stress_patches' into 13 ... * changes: Bridge system: Fix memory leaks and double frees on impart failure. bridge_softmix.c: Fix crash if channel fails to join mixing tech.	9 years ago
Richard Mudgett	1e93f3d723	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
Richard Mudgett	5e388d4188	bridge_softmix.c: Fix crash if channel fails to join mixing tech. ... softmix_bridge_join() failed because of an allocation failure. To address this, the softmix bridge technology now checks if the channel failed to join softmix successfully. In addition, the bridge now begins the process of kicking the channel out of the bridge so we don't have channels partially in the bridge for very long. * Fix the test_channel_feature_hooks.c unit tests. The test channel must have a valid codec to join the simple_bridge technology. This patch makes joining a bridge more strict by not allowing partially joined channels to remain in the bridge. Change-Id: I97e2ade6a2bcd1214f24fb839fda948825b61a2b	9 years ago
Richard Mudgett	9942d50aa5	bridge: Hold off more than one imparting channel at a time. ... An earlier patch blocked the ast_bridge_impart() call until the channel either entered the target bridge or it failed. Unfortuantely, if the target bridge is stasis and the imprted channel is not a stasis channel, stasis bounces the channel out of the bridge to come back into the bridge as a proper stasis channel. When the channel is bounced out, that released the block on ast_bridge_impart() to continue. If the impart was a result of a transfer, then it became a race to see if the swap channel would get hung up before the imparted channel could come back into the stasis bridge. If the imparted channel won then everything is fine. If the swap channel gets hung up first then the transfer will fail because the swap channel is leaving the bridge. * Allow a chain of ast_bridge_impart()'s to happen before any are unblocked to prevent the race condition described above. When the channel finally joins the bridge or completely fails to join the bridge then the ast_bridge_impart() instances are unblocked. ASTERISK-25947 Reported by: Richard Mudgett ASTERISK-24649 Reported by: John Bigelow ASTERISK-24782 Reported by: John Bigelow Change-Id: I8fef369171f295f580024ab4971e95c799d0dde1	9 years ago
Kevin Harwell	40d9e9e238	bridge.c: Crash during attended transfer when missing a local channel half ... It's possible for the transferer channel to get hung up early during the attended transfer process. For instance, a phone may send a "bye" immediately upon receiving a sip notify that contains a sip frag 100 (I'm looking at you Jitsi). When this occurs a race begins between the transferer being hung up and completion of the transfer code. If the channel hangs up too early during a transfer involving stasis bridging for instance, then when the created local channel goes to look up its swap channel (and associated datastore) it can't find it (since it is no longer in the bridge) thus it fails to enter the stasis application. Consequently, the created local channel(s) hang up as well. If the timing is just right then the bridging code attempts to add the message link with missing local channel(s). Hence the crash. Unfortunately, there is no great way to solve the problem of the unexpected "bye". While we can't guarantee we won't receive an early hangup, and in this case still fail to enter the stasis application, we can make it so asterisk does not crash. This patch does just that by locking the local channel structure, checking that the local channel's peer has not been lost, and then continuing. This keeps the local channel's peer from being ripped out from underneath it by the local/unreal hangup code while attempting to set the stasis message link. ASTERISK-25771 Change-Id: Ie6d6061e34c7c95f07116fffac9a09e5d225c880	9 years ago
Richard Mudgett	513638a5f4	bridge_channel: Don't settle owed events on an optimization. ... Local channel optimization could cause DTMF digits to be duplicated. Pending DTMF end events would be posted to a bridge when the local channel optimizes out and is replaced by the channel further down the chain. When the real digit ends, the channel would get another DTMF end posted to the bridge. A -- LocalA;1/n -- LocalA;2/n -- LocalB;1 -- LocalB;2 -- B 1) LocalA has the /n flag to prevent optimization. 2) B is sending DTMF to A through the local channel chain. 3) When LocalB optimizes out it can move B to the position of LocalB;1 4) Without this patch, when B swaps with LocalB;1 then LocalB;1 would settle an owed DTMF end to the bridge toward LocalA;2. 5) When B finally ends its DTMF it sends the DTMF end down the chain. 6) Without this patch, A would hear the DTMF digit end when LocalB optimizes out and when B ends the original digit. ASTERISK-25582 Change-Id: I1bbd28b8b399c0fb54985a5747f330a4cd2aa251	9 years ago
Jonathan Rose	eadad24b59	Unset BRIDGEPEER when leaving a bridge ... Currently if a channel is transferred out of a bridge, the BRIDGEPEER variable (also BRIDGEPVTCALLID) remain set even once the channel is out of the bridge. This patch removes these variables when leaving the bridge. ASTERISK-25600 #close Reported by: Mark Michelson Change-Id: I753ead2fffbfc65427ed4e9244c7066610e546da	10 years ago
Kevin Harwell	97ee0ee6c6	bridge.c: Fixed race condition during attended transfer ... During an attended transfer a thread is started that handles imparting the bridge channel. From the start of the thread to when the bridge channel is ready exists a gap that can potentially cause problems (for instance, the channel being swapped is hung up before the replacement channel enters the bridge thus stopping the transfer). This patch adds a condition that waits for the impart thread to get to a point of acceptable readiness before allowing the initiating thread to continue. ASTERISK-24782 Reported by: John Bigelow Change-Id: I08fe33a2560da924e676df55b181e46fca604577	10 years ago
Kevin Harwell	35a99b6394	bridge.c: Hangup attended transfer target if bridged ... After completing an attended transfer the transfer target channel was not being hung up after leaving the bridge. Added an explicit softhangup to hangup said channel, but only if it was previously bridged. ASTERISK-24782 #close Reported by: John Bigelow Change-Id: Idde9543d56842369384a5e8c00d72a22bbc39ada	10 years ago
Kevin Harwell	b56c1914fa	bridge.c: NULL app causes crash during attended transfer ... Due to a race condition there was a chance that during an attended transfer the channel's application would return NULL. This, of course, would cause a crash when attempting to access the memory. This patch retrieves the channel's app at an earlier time in processing in hopes that the app name is available. However, if it is not then "unknown" is used instead. Since some string value is now always present the crash can no longer occur. ASTERISK-24869 #close Reported by: viniciusfontes Review: Change-Id: I5134b84c4524906d8148817719d76ffb306488ac	10 years ago
Richard Mudgett	4441bb6a25	Bridging: Eliminate the unnecessary make channel compatible with bridge operation. ... When a channel enters the bridging system it is first made compatible with the bridge and then the bridge technology makes the channel compatible with the technology. For all but the DAHDI native and softmix bridge technologies the make channel compatible with the bridge step is an effective noop because the other technologies allow all audio formats. For the DAHDI native bridge technology it doesn't matter because it is not an initial bridge technology and chan_dahdi allows only one native format per channel. For the softmix bridge technology, it is a noop at best and harmful at worst because the wrong translation path could be setup if the channel's native formats allow more than one audio format. This is an intermediate patch for a series of patches aimed at improving translation path choices. * Removed code dealing with the unnecessary step of making the channel compatible with the bridge. ASTERISK-24841 Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/4600/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@434424 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
Kevin Harwell	d754f70239	bridge.c: Hangup attended transfer target after it has been swapped out ... After completing an attended transfer the transfer target channel (the one that gets swapped out) was not being hung up after leaving the bridge. This resulted in a channel possibly being left around. Added an explicit softhangup for the channel in question after the transfer is successfully completed in order to make sure the channel is hung up. ASTERISK-24782 #close Reported by: John Bigelow Review: https://reviewboard.asterisk.org/r/4575/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@434240 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
Corey Farrell	6adf26f14d	Replace most uses of ast_register_atexit with ast_register_cleanup. ... Since 'core stop now' and 'core restart now' do not stop modules, it is unsafe for most of the core to run cleanups. Originally all cleanups used ast_register_atexit, and were only changed when it was shown to be unsafe. ast_register_atexit is now used only when absolutely required to prevent corruption and close child processes. Exceptions that need to use ast_register_atexit: * CDR: Flush records. * res_musiconhold: Kill external applications. * AstDB: Close the DB. * canary_exit: Kill canary process. ASTERISK-24142 #close Reported by: David Brillert ASTERISK-24683 #close Reported by: Peter Katzmann ASTERISK-24805 #close Reported by: Badalian Vyacheslav ASTERISK-24881 #close Reported by: Corey Farrell Review: https://reviewboard.asterisk.org/r/4500/ Review: https://reviewboard.asterisk.org/r/4501/ ........ Merged revisions 433495 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@433497 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
Richard Mudgett	feddab7944	HTTP: Stop accepting requests on final system shutdown. ... There are three CLI commands to stop and restart Asterisk each. 1) core stop/restart now - Hangup all calls and stop or restart Asterisk. New channels are prevented while the shutdown request is pending. 2) core stop/restart gracefully - Stop or restart Asterisk when there are no calls remaining in the system. New channels are prevented while the shutdown request is pending. 3) core stop/restart when convenient - Stop or restart Asterisk when there are no calls in the system. New calls are not prevented while the shutdown request is pending. ARI has made stopping/restarting Asterisk more problematic. While a shutdown request is pending it is desirable to continue to process ARI HTTP requests for current calls. To handle the current calls while a shutdown request is pending, a new committed to shutdown phase is needed so ARI applications can deal with the calls until the system is fully committed to shutdown. * Added a new shutdown committed phase so ARI applications can deal with calls until the final committed to shutdown phase is reached. * Made refuse new HTTP requests when the system has reached the final system shutdown phase. Starting anything while the system is actively releasing resources and unloading modules is not a good thing. * Split the bridging framework shutdown to not cleanup the global bridging containers when shutting down in a hurry. This is similar to how other modules prevent crashes on rapid system shutdown. * Moved ast_begin_shutdown(), ast_cancel_shutdown(), and ast_shutting_down(). You should not have to include channel.h just to access these system functions. ASTERISK-24752 #close Reported by: Matthew Jordan Review: https://reviewboard.asterisk.org/r/4399/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@431692 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
Scott Griepentrog	f7d23dfcc6	stasis transfer: fix stasis bridge push race part two ... When swapping a Local channel in place of one already in a bridge (to complete a bridge attended transfer), the channel that was swapped out can actually be hung up before the stasis bridge push callback executes on the independant transfer thread. This results in the stasis app loop dropping out and removing the control that has the the app name which the local replacement channel needs so it can re-enter stasis. To avoid this race condition a new push_peek callback has been added, and called from the ast_bridge_impart thread before it launches the independant thread that will complete the transfer. Now the stasis push_peek callback can copy the stasis app name before the swap channel can hang up. ASTERISK-24649 Review: https://reviewboard.asterisk.org/r/4382/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@431450 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
Richard Mudgett	355eb9d22f	Bridge core: Pass a ref with the swap channel when joining a bridge. ... When code imparts a channel into a bridge to swap with another channel, a ref needs to be held on the swap channel to ensure that it cannot dissapear before finding it in the bridge. * The ast_bridge_join() swap channel parameter now always steals a ref for the swap channel. This is the only change to the bridge framework's public API semantics. * bridge_channel_internal_join() now requires the bridge_channel->swap channel to pass in a ref. ASTERISK-24649 Reported by: John Bigelow Review: https://reviewboard.asterisk.org/r/4354/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@430975 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
Richard Mudgett	4b363688d4	AMI: Make AMI actions that generate event lists consistent. ... * Made the following AMI actions use list API calls for consistency: Agents BridgeInfo BridgeList BridgeTechnologyList ConfbridgeLIst ConfbridgeLIstRooms CoreShowChannels DAHDIShowChannels DBGet DeviceStateList ExtensionStateList FAXSessions Hangup IAXpeerlist IAXpeers IAXregistry MeetmeList MeetmeListRooms MWIGet ParkedCalls Parkinglots PJSIPShowEndpoint PJSIPShowEndpoints PJSIPShowRegistrationsInbound PJSIPShowRegistrationsOutbound PJSIPShowResourceLists PJSIPShowSubscriptionsInbound PJSIPShowSubscriptionsOutbound PresenceStateList PRIShowSpans QueueStatus QueueSummary ShowDialPlan SIPpeers SIPpeerstatus SIPshowregistry SKINNYdevices SKINNYlines Status VoicemailUsersList * Incremented the AMI version to 2.7.0. * Changed astman_send_listack() to not use the listflag parameter and always set the value to "Start" so the start capitalization is consistent. i.e., The FAXSessions used "Start" while the rest of the system used "start". The corresponding complete event always used "Complete". * Fixed ami_show_resource_lists() "PJSIPShowResourceLists" to output the AMI ActionID for all of its list events. * Fixed off-nominal AMI protocol error in manager_bridge_info(), manager_parking_status_single_lot(), and manager_parking_status_all_lots(). Use of astman_send_error() after responding to the original AMI action request violates the action response pattern by sending two responses. * Fixed minor protocol error in action_getconfig() when no requested categories are found. Each line needs to be formatted as "Header: text". * Fixed off-nominal memory leak in manager_build_parked_call_string(). * Eliminated unnecessary use of RAII_VAR() in ami_subscription_detail(). ASTERISK-24049 #close Reported by: Jonathan Rose Review: https://reviewboard.asterisk.org/r/4315/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@430434 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	1583ea4832	DTMF hooks: Leaving channels need to push any collected digits into the bridge. ... Any partially collected DTMF digits for a DTMF hook need to be pushed into the bridge when a channel leaves the bridging system as if there were a timeout. Review: https://reviewboard.asterisk.org/r/4199/ ........ Merged revisions 428601 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@428602 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Mark Michelson	1536b0ecb6	Fix race condition that could result in ARI transfer messages not being sent. ... From reviewboard: "During blind transfer testing, it was noticed that tests were failing occasionally because the ARI blind transfer event was not being sent. After investigating, I detected a race condition in the blind transfer code. When blind transferring a single channel, the actual transfer operation (i.e. removing the transferee from the bridge and directing them to the proper dialplan location) is queued onto the transferee bridge channel. After queuing the transfer operation, the blind transfer Stasis message is published. At the time of publication, snapshots of the channels and bridge involved are created. The ARI subscriber to the blind transfer Stasis message then attempts to determine if the bridge or any of the involved channels are subscribed to by ARI applications. If so, then the blind transfer message is sent to the applications. The way that the ARI blind transfer message handler works is to first see if the transferer channel is subscribed to. If not, then iterate over all the channel IDs in the bridge snapshot and determine if any of those are subscribed to. In the test we were running, the lone transferee channel was subscribed to, so an ARI event should have been sent to our application. Occasionally, though, the bridge snapshot did not have any channels IDs on it at all. Why? The problem is that since the blind transfer operation is handled by a separate thread, it is possible that the transfer will have completed and the channels removed from the bridge before we publish the blind transfer Stasis message. Since the blind transfer has completed, the bridge on which the transfer occurred no longer has any channels on it, so the resulting bridge snapshot has no channels on it. Through investigation of the code, I found that attended transfers can have this issue too for the case where a transferee is transferred to an application." The fix employed here is to decouple the creation of snapshots for the transfer messages from the publication of the transfer messages. This way, snapshots can be created to reflect what they are at the time of the transfer operation. Review: https://reviewboard.asterisk.org/r/4135 ........ Merged revisions 427848 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@427870 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	294ff83152	main/bridge: Destroy features struct on off nominal path during bridge impart ... When a channel is imparted to a bridge, the invocation of the function may provide an ast_bridge_features struct. Upon passing this to ast_bridge_impart, the caller must assume that ownership has passed to the function, as in all paths the function destroys the struct prior to returning (as its purpose is to configure the behavior of the channel while in the bridge). On one off nominal path - where the channel already has a PBX thread - the struct was not being destroyed. This patch fixes that glitch. ASTERISK-24437 #close Reported by: Scott Griepentrog ........ Merged revisions 426431 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@426432 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Joshua Colp	743ad19699	bridge: During a smart bridge operation provide a more complete bridge to the old technology. ... When a smart bridge operation occurs and a bridge transitions from one technology to another the old technology is provided the channels formerly in it and told that they are leaving. Unfortunately the bridge provided along with them is incomplete. The bridge, despite there being channels in it, contains none. This forces technology implementations to have additional logic when channels are leaving or to store their own duplicated state. This change makes the bridge more complete so it contains the expected channels. Now that the bridge is complete special logic within bridge_native_rtp is no longer needed and has been removed. Review: https://reviewboard.asterisk.org/r/4057/ ........ Merged revisions 425242 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@425243 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	e8b72c6f4b	chan_pjsip: Update media translation paths when new SDP negotiated. ... On a SIP reinvite that changes media strams, the PJSIP channel driver was flooding the log with "Asked to transmit frame type %s, while native formats is %s" warnings. * Fixes PJSIP not setting up translation paths when the formats change on a reinvite. AFS-63 was effectively reintroduced because of the media formats work. res_pjsip_sdp_rtp.c:set_caps() * Improved the unexpected frame format WARNING message to include more information. * Added protective locking while altering formats on a channel. Reworked set_format() to simplify and protect the formats under manipulation. * Restored some code that got lost in the media_formats work. (channel.c:set_format() and res_pjsip_sdp_rtp.c:set_caps()) AFS-137 #close Reported by: Mark Michelson Review: https://reviewboard.asterisk.org/r/3906/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@421645 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Kinsey Moore	04f478212c	Stasis: Add information to blind transfer event ... When a blind transfer occurs that is forced to create a local channel pair to satisfy the transfer request, information about the local channel pair is not published. This adds a field to describe that channel to the blind transfer message struct so that this information is conveyed properly to consumers of the blind transfer message. This also fixes a bug in which Stasis() was unable to properly identify the channel that was replacing an existing Stasis-controlled channel due to a blind transfer. Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/3921/ ........ Merged revisions 421537 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/13@421538 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Mark Michelson	cb55679aed	Ensure bridges exist when trying to determine bridged parties when publishing transfer information. ... ........ Merged revisions 420387 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@420388 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Kinsey Moore	0ac7f96057	Stasis: Convey transfer information to applications ... This fixes a class of issues where Stasis applications were not made aware that their channels were being manipulated or replaced by external entitiessuch as transfers, AMI commands, or dialplan applications such as Bridge(). Inconsistent information such as StasisEnd events with unknown channels as a result of masquerades has also been corrected. To accomplish these fixes, several new fields were added to blind and attended transfer messages as well as StasisStart and BridgeAttendedTransfer Stasis events. ASTERISK-23941 #close Review: https://reviewboard.asterisk.org/r/3865/ Review: https://reviewboard.asterisk.org/r/3857/ Review: https://reviewboard.asterisk.org/r/3852/ Review: https://reviewboard.asterisk.org/r/3816/ Review: https://reviewboard.asterisk.org/r/3731/ Review: https://reviewboard.asterisk.org/r/3729/ Review: https://reviewboard.asterisk.org/r/3728/ ........ Merged revisions 420325 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@420338 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Kinsey Moore	f1036f40dc	Stasis: Allow message types to be blocked ... This introduces stasis.conf and a mechanism to prevent certain message types from being published. Internally, this works by preventing the chosen message types from being created which ensures that those message types can never be published. This patch also adjusts message publishers such that message payloads are not created if the related message type is not available. ASTERISK-23943 #close Review: https://reviewboard.asterisk.org/r/3823/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@420124 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Joshua Colp	b2d6a9e076	bridge: Make "bridge destroy" only available in developer mode and add "all" to "bridge kick". ... The "bridge destroy" CLI command is invasive to bridges and can leave them in an unexpected state for the users of them. Since this command may be useful for developers it is now only available when developer mode is available. To take its place "all" has been added as a valid option to the "bridge kick" CLI command. It will kick all of the channels in the bridge out. ASTERISK-23987 Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/3840/ ........ Merged revisions 419536 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@419537 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	a2ce95d9d2	accountcode: Slightly change accountcode propagation. ... The previous behavior was to simply set the accountcode of an outgoing channel to the accountcode of the channel initiating the call. It was done this way a long time ago to allow the accountcode set on the SIP/100 channel to be propagated to a local channel so the dialplan execution on the Local;2 channel would have the SIP/100 accountcode available. SIP/100 -> Local;1/Local;2 -> SIP/200 Propagating the SIP/100 accountcode to the local channels is very useful. Without any dialplan manipulation, all channels in this call would have the same accountcode. Using dialplan, you can set a different accountcode on the SIP/200 channel either by setting the accountcode on the Local;2 channel or by the Dial application's b(pre-dial), M(macro) or U(gosub) options, or by the FollowMe application's b(pre-dial) option, or by the Queue application's macro or gosub options. Before Asterisk v12, the altered accountcode on SIP/200 will remain until the local channels optimize out and the accountcode would change to the SIP/100 accountcode. Asterisk v1.8 attempted to add peeraccount support but ultimately had to punt on the support. The peeraccount support was rendered useless because of how the CDR code needed to unconditionally force the caller's accountcode onto the peer channel's accountcode. The CEL events were thus intentionally made to always use the channel's accountcode as the peeraccount value. With the arrival of Asterisk v12, the situation has improved somewhat so peeraccount support can be made to work. Using the indicated example, the the accountcode values become as follows when the peeraccount is set on SIP/100 before calling SIP/200: SIP/100 ---> Local;1 ---- Local;2 ---> SIP/200 acct: 100 \/ acct: 200 \/ acct: 100 \/ acct: 200 peer: 200 /\ peer: 100 /\ peer: 200 /\ peer: 100 If a channel already has an accountcode it can only change by the following explicit user actions: 1) A channel originate method that can specify an accountcode to use. 2) The calling channel propagating its non-empty peeraccount or its non-empty accountcode if the peeraccount was empty to the outgoing channel's accountcode before initiating the dial. e.g., Dial and FollowMe. The exception to this propagation method is Queue. Queue will only propagate peeraccounts this way only if the outgoing channel does not have an accountcode. 3) Dialplan using CHANNEL(accountcode). 4) Dialplan using CHANNEL(peeraccount) on the other end of a local channel pair. If a channel does not have an accountcode it can get one from the following places: 1) The channel driver's configuration at channel creation. 2) Explicit user action as already indicated. 3) Entering a basic or stasis-mixing bridge from a peer channel's peeraccount value. You can specify the accountcode for an outgoing channel by setting the CHANNEL(peeraccount) before using the Dial, FollowMe, and Queue applications. Queue adds the wrinkle that it will not overwrite an existing accountcode on the outgoing channel with the calling channels values. Accountcode and peeraccount values propagate to an outgoing channel before dialing. Accountcodes also propagate when channels enter or leave a basic or stasis-mixing bridge. The peeraccount value only makes sense for mixing bridges with two channels; it is meaningless otherwise. * Made peeraccount functional by changing accountcode propagation as described above. * Fixed CEL extracting the wrong ie value for the peeraccount. This was done intentionally in Asterisk v1.8 when that version had to punt on peeraccount. * Fixed a few places dealing with accountcodes that were reading from channels without the lock held. AFS-65 #close Review: https://reviewboard.asterisk.org/r/3601/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@419520 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	a2c912e997	media formats: re-architect handling of media for performance improvements ... In the old times media formats were represented using a bit field. This was fast but had a few limitations. 1. Asterisk was limited in how many formats it could handle. 2. Formats, being a bit field, could not include any attribute information. A format was strictly its type, e.g., "this is ulaw". This was changed in Asterisk 10 (see https://wiki.asterisk.org/wiki/display/AST/Media+Architecture+Proposal for notes on that work) which led to the creation of the ast_format structure. This structure allowed Asterisk to handle attributes and bundle information with a format. Additionally, ast_format_cap was created to act as a container for multiple formats that, together, formed the capability of some entity. Another mechanism was added to allow logic to be registered which performed format attribute negotiation. Everywhere throughout the codebase Asterisk was changed to use this strategy. Unfortunately, in software, there is no free lunch. These new capabilities came at a cost. Performance analysis and profiling showed that we spend an inordinate amount of time comparing, copying, and generally manipulating formats and their related structures. Basic prototyping has shown that a reasonably large performance improvement could be made in this area. This patch is the result of that project, which overhauled the media format architecture and its usage in Asterisk to improve performance. Generally, the new philosophy for handling formats is as follows: * The ast_format structure is reference counted. This removed a large amount of the memory allocations and copying that was done in prior versions. * In order to prevent race conditions while keeping things performant, the ast_format structure is immutable by convention and lock-free. Violate this tenet at your peril! * Because formats are reference counted, codecs are also reference counted. The Asterisk core generally provides built-in codecs and caches the ast_format structures created to represent them. Generally, to prevent inordinate amounts of module reference bumping, codecs and formats can be added at run-time but cannot be removed. * All compatibility with the bit field representation of codecs/formats has been moved to a compatibility API. The primary user of this representation is chan_iax2, which must continue to maintain its bit-field usage of formats for interoperability concerns. * When a format is negotiated with attributes, or when a format cannot be represented by one of the cached formats, a new format object is created or cloned from an existing format. That format may have the same codec underlying it, but is a different format than a version of the format with different attributes or without attributes. * While formats are reference counted objects, the reference count maintained on the format should be manipulated with care. Formats are generally cached and will persist for the lifetime of Asterisk and do not explicitly need to have their lifetime modified. An exception to this is when the user of a format does not know where the format came from *and* the user may outlive the provider of the format. This occurs, for example, when a format is read from a channel: the channel may have a format with attributes (hence, non-cached) and the user of the format may last longer than the channel (if the reference to the channel is released prior to the format's reference). For more information on this work, see the API design notes: https://wiki.asterisk.org/wiki/display/AST/Media+Format+Rewrite Finally, this work was the culmination of a large number of developer's efforts. Extra thanks goes to Corey Farrell, who took on a large amount of the work in the Asterisk core, chan_sip, and was an invaluable resource in peer reviews throughout this project. There were a substantial number of patches contributed during this work; the following issues/patch names simply reflect some of the work (and will cause the release scripts to give attribution to the individuals who work on them). Reviews: https://reviewboard.asterisk.org/r/3814 https://reviewboard.asterisk.org/r/3808 https://reviewboard.asterisk.org/r/3805 https://reviewboard.asterisk.org/r/3803 https://reviewboard.asterisk.org/r/3801 https://reviewboard.asterisk.org/r/3798 https://reviewboard.asterisk.org/r/3800 https://reviewboard.asterisk.org/r/3794 https://reviewboard.asterisk.org/r/3793 https://reviewboard.asterisk.org/r/3792 https://reviewboard.asterisk.org/r/3791 https://reviewboard.asterisk.org/r/3790 https://reviewboard.asterisk.org/r/3789 https://reviewboard.asterisk.org/r/3788 https://reviewboard.asterisk.org/r/3787 https://reviewboard.asterisk.org/r/3786 https://reviewboard.asterisk.org/r/3784 https://reviewboard.asterisk.org/r/3783 https://reviewboard.asterisk.org/r/3778 https://reviewboard.asterisk.org/r/3774 https://reviewboard.asterisk.org/r/3775 https://reviewboard.asterisk.org/r/3772 https://reviewboard.asterisk.org/r/3761 https://reviewboard.asterisk.org/r/3754 https://reviewboard.asterisk.org/r/3753 https://reviewboard.asterisk.org/r/3751 https://reviewboard.asterisk.org/r/3750 https://reviewboard.asterisk.org/r/3748 https://reviewboard.asterisk.org/r/3747 https://reviewboard.asterisk.org/r/3746 https://reviewboard.asterisk.org/r/3742 https://reviewboard.asterisk.org/r/3740 https://reviewboard.asterisk.org/r/3739 https://reviewboard.asterisk.org/r/3738 https://reviewboard.asterisk.org/r/3737 https://reviewboard.asterisk.org/r/3736 https://reviewboard.asterisk.org/r/3734 https://reviewboard.asterisk.org/r/3722 https://reviewboard.asterisk.org/r/3713 https://reviewboard.asterisk.org/r/3703 https://reviewboard.asterisk.org/r/3689 https://reviewboard.asterisk.org/r/3687 https://reviewboard.asterisk.org/r/3674 https://reviewboard.asterisk.org/r/3671 https://reviewboard.asterisk.org/r/3667 https://reviewboard.asterisk.org/r/3665 https://reviewboard.asterisk.org/r/3625 https://reviewboard.asterisk.org/r/3602 https://reviewboard.asterisk.org/r/3519 https://reviewboard.asterisk.org/r/3518 https://reviewboard.asterisk.org/r/3516 https://reviewboard.asterisk.org/r/3515 https://reviewboard.asterisk.org/r/3512 https://reviewboard.asterisk.org/r/3506 https://reviewboard.asterisk.org/r/3413 https://reviewboard.asterisk.org/r/3410 https://reviewboard.asterisk.org/r/3387 https://reviewboard.asterisk.org/r/3388 https://reviewboard.asterisk.org/r/3389 https://reviewboard.asterisk.org/r/3390 https://reviewboard.asterisk.org/r/3321 https://reviewboard.asterisk.org/r/3320 https://reviewboard.asterisk.org/r/3319 https://reviewboard.asterisk.org/r/3318 https://reviewboard.asterisk.org/r/3266 https://reviewboard.asterisk.org/r/3265 https://reviewboard.asterisk.org/r/3234 https://reviewboard.asterisk.org/r/3178 ASTERISK-23114 #close Reported by: mjordan media_formats_translation_core.diff uploaded by kharwell (License 6464) rb3506.diff uploaded by mjordan (License 6283) media_format_app_file.diff uploaded by kharwell (License 6464) misc-2.diff uploaded by file (License 5000) chan_mild-3.diff uploaded by file (License 5000) chan_obscure.diff uploaded by file (License 5000) jingle.diff uploaded by file (License 5000) funcs.diff uploaded by file (License 5000) formats.diff uploaded by file (License 5000) core.diff uploaded by file (License 5000) bridges.diff uploaded by file (License 5000) mf-codecs-2.diff uploaded by file (License 5000) mf-app_fax.diff uploaded by file (License 5000) mf-apps-3.diff uploaded by file (License 5000) media-formats-3.diff uploaded by file (License 5000) ASTERISK-23715 rb3713.patch uploaded by coreyfarrell (License 5909) rb3689.patch uploaded by mjordan (License 6283) ASTERISK-23957 rb3722.patch uploaded by mjordan (License 6283) mf-attributes-3.diff uploaded by file (License 5000) ASTERISK-23958 Tested by: jrose rb3822.patch uploaded by coreyfarrell (License 5909) rb3800.patch uploaded by jrose (License 6182) chan_sip.diff uploaded by mjordan (License 6283) rb3747.patch uploaded by jrose (License 6182) ASTERISK-23959 #close Tested by: sgriepentrog, mjordan, coreyfarrell sip_cleanup.diff uploaded by opticron (License 6273) chan_sip_caps.diff uploaded by mjordan (License 6283) rb3751.patch uploaded by coreyfarrell (License 5909) chan_sip-3.diff uploaded by file (License 5000) ASTERISK-23960 #close Tested by: opticron direct_media.diff uploaded by opticron (License 6273) pjsip-direct-media.diff uploaded by file (License 5000) format_cap_remove.diff uploaded by opticron (License 6273) media_format_fixes.diff uploaded by opticron (License 6273) chan_pjsip-2.diff uploaded by file (License 5000) ASTERISK-23966 #close Tested by: rmudgett rb3803.patch uploaded by rmudgetti (License 5621) chan_dahdi.diff uploaded by file (License 5000) ASTERISK-24064 #close Tested by: coreyfarrell, mjordan, opticron, file, rmudgett, sgriepentrog, jrose rb3814.patch uploaded by rmudgett (License 5621) moh_cleanup.diff uploaded by opticron (License 6273) bridge_leak.diff uploaded by opticron (License 6273) translate.diff uploaded by file (License 5000) rb3795.patch uploaded by rmudgett (License 5621) tls_fix.diff uploaded by mjordan (License 6283) fax-mf-fix-2.diff uploaded by file (License 5000) rtp_transfer_stuff uploaded by mjordan (License 6283) rb3787.patch uploaded by rmudgett (License 5621) media-formats-explicit-translate-format-3.diff uploaded by file (License 5000) format_cache_case_fix.diff uploaded by opticron (License 6273) rb3774.patch uploaded by rmudgett (License 5621) rb3775.patch uploaded by rmudgett (License 5621) rtp_engine_fix.diff uploaded by opticron (License 6273) rtp_crash_fix.diff uploaded by opticron (License 6273) rb3753.patch uploaded by mjordan (License 6283) rb3750.patch uploaded by mjordan (License 6283) rb3748.patch uploaded by rmudgett (License 5621) media_format_fixes.diff uploaded by opticron (License 6273) rb3740.patch uploaded by mjordan (License 6283) rb3739.patch uploaded by mjordan (License 6283) rb3734.patch uploaded by mjordan (License 6283) rb3689.patch uploaded by mjordan (License 6283) rb3674.patch uploaded by coreyfarrell (License 5909) rb3671.patch uploaded by coreyfarrell (License 5909) rb3667.patch uploaded by coreyfarrell (License 5909) rb3665.patch uploaded by mjordan (License 6283) rb3625.patch uploaded by coreyfarrell (License 5909) rb3602.patch uploaded by coreyfarrell (License 5909) format_compatibility-2.diff uploaded by file (License 5000) core.diff uploaded by file (License 5000) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@419044 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Kinsey Moore	e977b7936b	Bridging: Allow channels to define bridging hooks ... This patch allows the current owner of a channel to define various feature hooks to be made available once the channel has entered a bridge. This includes any hooks that are setup on the ast_bridge_features struct such as DTMF hooks, bridge event hooks (join, leave, etc.), and interval hooks. Review: https://reviewboard.asterisk.org/r/3649/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@417361 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Kinsey Moore	bd36288efa	Fix build warnings with TEST_FRAMEWORK enabled ... ........ Merged revisions 416732 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 416733 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 416734 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@416736 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	fb5690ce4b	Logger/CLI/etc.: Fix some aesthetic issues; reduce chatty verbose messages ... This patch addresses some aesthetic issues in Asterisk. These are all just minor tweaks to improve the look of the CLI when used in a variety of settings. Specifically: * A number of chatty verbose messages were removed or demoted to DEBUG messages. Verbose messages with a verbosity level of 5 or higher were - if kept as verbose messages - demoted to level 4. Several messages that were emitted at verbose level 3 were demoted to 4, as announcement of dialplan applications being executed occur at level 3 (and so the effects of those applications should generally be less). * Some verbose messages that only appear when their respective 'debug' options are enabled were bumped up to always be displayed. * Prefix/timestamping of verbose messages were moved to the verboser handlers. This was done to prevent duplication of prefixes when the timestamp option (-T) is used with the CLI. * Verbose magic is removed from messages before being emitted to non-verboser handlers. This prevents the magic in multi-line verbose messages (such as SIP debug traces or the output of DumpChan) from being written to files. * _Slightly_ better support for the "light background" option (-W) was added. This includes using ast_term_quit in the output of XML documentation help, as well as changing the "Asterisk Ready" prompt to bright green on the default background (which stands a better chance of being displayed properly than bright white). Review: https://reviewboard.asterisk.org/r/3547/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@414798 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Jonathan Rose	d00882108f	res_pjsip_refer: Fix bugs involving Parking/PJSIP/transfers ... PJSIP would never send the final 200 Notify for a blind transfer when transferring to parking. This patch fixes that. In addition, it fixes a reference leak when performing blind transfers to non-bridging extensions. Review: https://reviewboard.asterisk.org/r/3485/ ........ Merged revisions 414400 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@414403 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Kinsey Moore	abd3e4040b	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 ........ Merged revisions 413587 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 413588 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@413589 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Kinsey Moore	f7caf4e249	Bridging: Don't lock NULL bridges ... When bridge locking was added for bridge snapshot creation, some locations where bridge locking was added were not guaranteed to actually have a bridge and locking NULL AO2 objects tends to cause segfaults. This ensures that NULL bridges aren't locked. ........ Merged revisions 413073 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@413074 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Kinsey Moore	d6e2c50058	bridging: Ensure locking during snapshot creation ... While the vast majority of bridge snapshot creation is locked properly, there are currently some instances that are not. This adds the missing locking to ensure bridge state is not malleable during snapshot creation. (closes issue ASTERISK-22904) Review: https://reviewboard.asterisk.org/r/3415/ Reported by: Matt Jordan ........ Merged revisions 412193 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@412194 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	28c0cb28d0	channel locking: Add locking for channel snapshot creation ... Original commit message by mmichelson (asterisk 12 r403311): "This adds channel locks around calls to create channel snapshots as well as other functions which operate on a channel and then end up creating a channel snapshot. Functions that expect the channel to be locked prior to being called have had their documentation updated to indicate such." The above was initially committed and then reverted at r403398. The problem was found to be in core_local.c in the publish_local_bridge_message function. The ast_unreal_lock_all function locks and adds a reference to the returned channels and while they were being unlocked they were not being unreffed when no longer needed. Fixed by unreffing the channels. Also in bridge.c a lock was obtained on "other->chan", but then an attempt was made to unlock "other" and not the previously locked channel. Fixed by unlocking "other->chan" (closes issue ASTERISK-22709) Reported by: John Bigelow ........ Merged revisions 404237 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@404260 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Jonathan Rose	b0bb03e916	bridging: Give bridges a name and a known creator ... Bridges have two new optional properties, a creator and a name. Certain consumers of bridges will automatically provide bridges that they create with these properties. Examples include app_bridgewait, res_parking, app_confbridge, and app_agent_pool. In addition, a name may now be provided as an argument to the POST function for creating new bridges via ARI. (closes issue AFS-47) Review: https://reviewboard.asterisk.org/r/3070/ ........ Merged revisions 404042 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@404043 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Jonathan Rose	00dcee2a64	transfers: Fix bug setting both BLINDTRANSFER and ATTENDEDTRANSFER ... The ast_bridge_set_transfer_variables function is supposed to wipe whichever variable isn't being set. Instead it was setting both to the new value. Oops. (issue AFS-24) ........ Merged revisions 403957 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403958 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Jonathan Rose	63b2c28c4b	Transfers: Make Asterisk set ATTENDEDTRANSFER/BLINDTRANSFER more reliably ... There were still a few cases in which ATTENDEDTRANSFER and BLINDTRANSFER wouldn't be set on channels involved with blind and attended transfers. This would happen with features that were initialized by channel driver specific mechanisms in multiparty calls. This patch resolves those cases while attempted to keep the behavior for setting those variables as consistent as possible. (closes issue AFS-24) Review: https://reviewboard.asterisk.org/r/3040/ ........ Merged revisions 403781 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403783 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
David M. Lee	1212906351	Reverting r403311. It's causing ARI tests to hang. ... ........ Merged revisions 403398 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403404 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Mark Michelson	8e8b329e14	Add channel locking for channel snapshot creation. ... This adds channel locks around calls to create channel snapshots as well as other functions which operate on a channel and then end up creating a channel snapshot. Functions that expect the channel to be locked prior to being called have had their documentation updated to indicate such. ........ Merged revisions 403311 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403314 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Scott Griepentrog	3b36687a56	Manager: Add equivalent AMI actions for the bridge CLI commands. ... Adds the following AMI events, closely following their CLI counterparts: BridgeDestroy BridgeKick BridgeTechnologyList BridgeTechnologySuspend BridgeTechnologyUnsuspend BridgeDestroy kicks an entire bridge, where BridgeKick kicks just one channel off the bridge. When kicking a channel, specifying the bridge also (optional) insures it is not removed from the wrong bridge. The BridgeTechnology events allow viewing and changing suspension status, which affects only subsequent not active bridging. (closes ASTERISK-22356) Reported by: Richard Mudgett Review: https://reviewboard.asterisk.org/r/2973/ ........ Merged revisions 402387 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@402388 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	d5db1f76f8	Bridging: Fix orphaned bridge if neither of the joining channels can join. ... The original issue noted that the bridge is orphaned when res_parking.so is not loaded and a call uses the dial kK flags. A similar issue happens when only one of the park flags is used. In this case you have the bridge with one or the other channel left in it. The channel and bridge will stay around until the channel hangs up. * Fixed the initial bridge channel push failure to act as if the channel were kicked out of the bridge. The bridge then decides if it needs to be dissolved. (closes issue ASTERISK-22629) Reported by: Kevin Harwell Review: https://reviewboard.asterisk.org/r/2928/ ........ Merged revisions 401424 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@401425 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Mark Michelson	c30170d9a2	Resolve some memory leaks due to incorrect for loop / ao2 ref usage. ... A common idiom in Asterisk is to due something like: for (ao2_obj = list_beginning; ao2_obj = next_item; ao2_ref(ao2_obj, -1)) { ...do stuff... } This is nice because it automatically takes care of the object references for you. However, there is a pitfall here. If a break statement is in the for loop, then the current reference is not cleaned up. In some cases, this is on purpose, but in others there is a leak. This commit fixes the leak cases. ........ Merged revisions 401248 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@401249 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago