122 Commits (55e8c9eff24b5d6f22cbb0f9a9aa165e63f3d538)

Author	SHA1	Message	Date
Richard Mudgett	71dfa35540	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	ff3af764de	bridge_softmix.c: Fix crash if could not allocate the dsp. ... Fix off nominal crash where we could not setup the channel to process frames for the softmix bridge technology because of allocation failure. Change-Id: Ic307a8386e46bf551e48fcd1eb97276714d56372	9 years ago
Matt Jordan	75c800eb28	Revert "bridges/bridge_t38: Add a bridging module for managing T.38 state" ... This reverts commit f42d22d3a1. Unfortunately, using a bridge to manage T.38 state will cause severe deadlocks in core_unreal/chan_local. Local channels attempt to reach across both their peer and the peer's bridge to inspect T.38 state. Given the propensity of Local channel chains, managing the locking situation in such a scenario is practically infeasible. Change-Id: I932107387c13aad2c75a7a4c1e94197a9d6d8a51	10 years ago
Matt Jordan	f42d22d3a1	bridges/bridge_t38: Add a bridging module for managing T.38 state ... When 4875e5ac32 was merged, it fixed several issues with a direct media bridge transitioning to handling a T.38 fax. However, it uncovered a race condition caused by the bridging core. When a channel involved in a T.38 fax leaves a bridge, the frame queued by the channel driver that should inform the far side that it is no longer in a T.38 fax may not make it across the bridge. The bridging framework is *extremely* aggressive in tearing down the bridge, and control frames that are currently in flight *may* get dropped. This patch adds a new module to the bridging framework, bridge_t38. This module maintains some notion of the T.38 state for the two channels in a bridge. When the bridge detects that it is being torn down or when one of the two channels leaves, it informs the respective channel(s) that they should stop faxing. This ensures that channels switch back to audio if they survive and are ejected out of a bridge while faxing. ASTERISK-25582 Change-Id: If5b0bb478eb01c4607c9f4a7fc17c7957d260ea0	10 years ago
Alexander Traud	cf79b62778	ast_format_cap_get_names: To display all formats, the buffer was increased. ... ASTERISK-25533 #close Change-Id: Ie1a9d1a6511b3f1a56b93d04475fbf8a4e40010a	10 years ago
Jonathan Rose	687597ca8c	holding_bridge: ensure moh participants get frames ... Currently, if a blank musiconhold.conf is used, musiconhold will fail to start for a channel going into a holding bridge with an anticipation of getting music on hold. That being the case, no frames will be written to the channel and that can pose a problem for blind transfers in PJSIP which may rely on frames being written to get past the REFER framehook. This patch makes holding bridges start a silence generator if starting music on hold fails and makes it so that if no music on hold functions are installed that the ast_moh_start function will report a failure so that consumers of that function will be able to respond appropriately. ASTERISK-25271 #close Change-Id: I06f066728604943cba0bb0b39fa7cf658a21cd99 (cherry picked from commit 8458b8d441)	10 years ago
Joshua Colp	4a25d55416	bridge_native_rtp.c: Don't start native RTP bridging after attended transfer. ... The bridge_native_rtp module adds a frame hook to channels which are in a native RTP bridge. This frame hook is used to intercept when a hold or unhold frame traverses the bridge so native RTP can be stopped or started as appropriate. This is expected but exposes a specific bug when attended transfers are involved. Upon completion of an attended transfer an unhold frame is queued up to take one of the channels involved off hold. After this is done the channel is moved between bridges. When the frame hook is involved in this case for the unhold it releases the channel lock and acquires the bridge lock. This allows the bridge core to step in and move the channel (potentially changing the bridging techology) from another thread. Once completed the bridge lock is released by the bridge core. The frame hook is then able to acquire the bridge lock and wrongfully starts native RTP again, despite the channel no longer being in the bridge or needing to start native RTP. In fact at this point the frame hook is no longer attached to the channel. This change makes it so the native RTP bridge data is available to the frame hook when it is invoked. Whether the frame hook has been detached or not is stored on the native RTP bridge data and is checked by the frame hook before starting or stopping native RTP bridging. If the frame hook has been detached it does nothing. ASTERISK-25240 #close Change-Id: I13a73186a05f4e5a764f81e5cd0ccec1ed1891d2	10 years ago
Matt Jordan	4a58261694	git migration: Refactor the ASTERISK_FILE_VERSION macro ... Git does not support the ability to replace a token with a version string during check-in. While it does have support for replacing a token on clone, this is somewhat sub-optimal: the token is replaced with the object hash, which is not particularly easy for human consumption. What's more, in practice, the source file version was often not terribly useful. Generally, when triaging bugs, the overall version of Asterisk is far more useful than an individual SVN version of a file. As a result, this patch removes Asterisk's support for showing source file versions. Specifically, it does the following: * Rename ASTERISK_FILE_VERSION macro to ASTERISK_REGISTER_FILE, and remove passing the version in with the macro. Other facilities than 'core show file version' make use of the file names, such as setting a debug level only on a specific file. As such, the act of registering source files with the Asterisk core still has use. The macro rename now reflects the new macro purpose. * main/asterisk: - Refactor the file_version structure to reflect that it no longer tracks a version field. - Remove the "core show file version" CLI command. Without the file version, it is no longer useful. - Remove the ast_file_version_find function. The file version is no longer tracked. - Rename ast_register_file_version/ast_unregister_file_version to ast_register_file/ast_unregister_file, respectively. * main/manager: Remove value from the Version key of the ModuleCheck Action. The actual key itself has not been removed, as doing so would absolutely constitute a backwards incompatible change. However, since the file version is no longer tracked, there is no need to attempt to include it in the Version key. * UPGRADE: Add notes for: - Modification to the ModuleCheck AMI Action - Removal of the "core show file version" CLI command Change-Id: I6cf0ff280e1668bf4957dc21f32a5ff43444a40e	10 years ago
Richard Mudgett	c499cabf53	chan_pjsip/res_pjsip/bridge_softmix/core: Improve translation path choices. ... With this patch, chan_pjsip/res_pjsip now sets the native formats to the codecs negotiated by a call. * The changes in chan_pjsip.c and res_pjsip_sdp_rtp.c set the native formats to include all the negotiated audio codecs instead of only the initial preferred audio codec and later the currently received audio codec. * The audio frame handling in channel.c:ast_read() is more streamlined and will automatically adjust to changes in received frame formats. The new policy is to remove translation and pass the new frame format to the receiver except if the translation was to a signed linear format. A more long winded version is commented in ast_read() along with some caveats. * The audio frame handling in channel.c:ast_write() is more streamlined and will automatically adjust any needed translation to changes in the frame formats sent. Frame formats sent can change for many reasons such as a recording is being played back or the bridged peer changed the format it sends. Since it is a normal expectation that sent formats can change, the codec mismatch warning message is demoted to a debug message. * Removed the short circuit check in channel.c:ast_channel_make_compatible_helper(). Two party bridges need to make channels compatible with each other. However, transfers and moving channels among bridges can result in otherwise compatible channels having sub-optimal translation paths if the make compatible check is short circuited. A result of forcing the reevaluation of channel compatibility is that the asterisk.conf:transcode_via_slin and codecs.conf:genericplc options take effect consistently now. It is unfortunate that these two options are enabled by default and negate some of the benefits to the changes in channel.c:ast_read() by forcing translation through signed linear on a two party bridge. * Improved the softmix bridge technology to better control the translation of frames to the bridge. All of the incoming translation is now normally handled by ast_read() instead of splitting any translation steps between ast_read() and the slin factory. If any frame comes in with an unexpected format then the translation path in ast_read() is updated for the next frame and the slin factory handles the current frame translation. This is the final patch in a series of patches aimed at improving translation path choices. The other patches are on the following reviews: https://reviewboard.asterisk.org/r/4600/ https://reviewboard.asterisk.org/r/4605/ ASTERISK-24841 #close Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/4609/ ........ Merged revisions 434671 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@434672 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
Richard Mudgett	6f1a7fe05f	bridge_softmix.c,channel.c: Minor code simplification and cleanup. ... * Made code easier to follow in bridge_softmix.c:analyse_softmix_stats() and made some debug messages more helpful. * Made some debug and warning messages more helpful in channel.c:set_format(). Review: https://reviewboard.asterisk.org/r/4607/ ........ Merged revisions 434617 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@434618 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
Richard Mudgett	459171be12	bridge_native_rtp.c: Defer allocation and check if it fails in native_rtp_bridge_compatible(). ... Review: https://reviewboard.asterisk.org/r/4601/ ........ Merged revisions 434508 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@434509 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
Richard Mudgett	09df34d880	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/ ........ Merged revisions 434424 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@434430 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
Kevin Harwell	91733b5d15	bridge_softmix: G.729 codec license held ... When more than one call using the same codec type enters into a softmix bridge and no audio is present for a channel the bridge optimizes the out frame by using the same one for all channels with the same codec type. Unfortunately, when that number (channels with same codec type) dropped to <= 1 the codec was not dereferenced. At least not until all parties left the bridge. Thus in the case of G.729 the license was not released. This patch ensures that the codec is dereferenced immediately when the optimization no longer applies. ASTERISK-24797 #close Reported by: Luke Hulsey Review: https://reviewboard.asterisk.org/r/4429/ ........ Merged revisions 432174 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 432175 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@432176 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
Corey Farrell	8cc50b1ebc	Enable REF_DEBUG for ast_module_ref / ast_module_unref. ... Add ast_module_shutdown_ref for use by modules that can only be unloaded during graceful shutdown. When REF_DEBUG is enabled: * Add an empty ao2 object to struct ast_module. * Allocate ao2 object when the module is loaded. * Perform an ao2_ref in each place where mod->usecount is manipulated. * ao2_cleanup on module unload. ASTERISK-24479 #close Reported by: Corey Farrell Review: https://reviewboard.asterisk.org/r/4141/ ........ Merged revisions 431662 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 431663 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@431672 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
George Joseph	5f60ebc004	bridge_native_rtp: Change local/remote message from debug/2 to verb/4 ... Change the "Locally bridged"/"Remotely bridged" messages from dbg/2 to verb/4. Tested-by: George Joseph Review: https://reviewboard.asterisk.org/r/4300/ ........ Merged revisions 430225 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@430226 65c4cc65-6c06-0410-ace0-fbb531ad65f3	10 years ago
Matthew Jordan	9a1ab5d548	bridge_native_rtp: Fix T.38 issues with remote bridges ... After r425242 the fax/sip/directmedia_reinvite_t38 test started failing due to the surviving channel not being re-INVITEd back from T.38 to audio. This patch fixes that bug - a deeper explanation of what happened follows. When two RTP channels are in a native bridge, the bridging layer will investigate each via the get_rtp_info glue callback. This callback returns the native bridge preference of the channel *at that moment in time* (that part is key). At different points during the bridging, the native bridging layer will inform the RTP capable channels of the status of the bridge via the update_peer glue callback. In a T.38 scenario with audio direct media, the sequence of events will often look like the following: * SIP/A and SIP/B both have audio and enter a native bridge. * Asterisk re-INVITEs audio between SIP/A and SIP/B directly (via an update_peer callback). * SIP/A sends a re-INVITE to T.38, which causes Asterisk to send a re-INVITE to T.38 to SIP/B. Assuming everyone 200 OKs the process, the UDPTL stack receives UDPTL packets in Asterisk from both endpoints. From the perspective of the channels, we are now in a local bridge for T.38, even though we are technically still in a remote bridge in bridge_native_rtp. (YAY!) * When one side hangs up, bridge_native_rtp is told to stop bridging. It then re-evaluates the channels and asks them how they are bridged - and since T.38 is enabled, they reply with a Local bridge (which is correct), but is wrong because the audio portion is still technically in a remote bridge. * Asterisk releases the surviving channel, whose audio is *not* re-INVITED back to Asterisk as bridge_native_rtp incorrectly assumes that it was in a local bridge. Ironically, prior to r425242, this used to work mostly due to a fluke in the bridging layer. The purpose of the get_rtp_info callback shouldn't be modified: it should tell the bridging layer what kind of bridge the channel prefers at that moment in time. If you have T.38 enabled, that *must* be a local bridge, as the UDPTPL stack must be in the media path. As such, this patch does not modify that part of the code. However, we have to tell the channels to re-evaluate themselves when they come out of a native bridge, since we can no longer trust the get_rtp_info callbacks when the native bridge is being stopped. Something else may have changed in the channels, and they may now be lying to us. As such, this patch makes it so that we unilaterally tell the channels that they are no longer bridged via the update_peer callback. This is actually what the channels expect anyway: code in both chan_sip and chan_pjsip's callbacks look at the T.38 state and - if they were in T.38 - send a re-INVITE to get the audio back to Asterisk. Review: https://reviewboard.asterisk.org/r/4157/ ........ Merged revisions 427582 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 427583 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@427584 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	0ed8aebda9	bridge_builtin_features: Add missing channel locks around ast_get_chan_features_general_config(). ... The feature_automonitor() and feature_automixmonitor() functions were not locking the channel around ast_get_chan_features_general_config(). Accessing the channel datastore list without the channel locked is a good way to corrupt the list or follow the pointer chain into oblivion. ........ Merged revisions 426531 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 426552 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@426553 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	df59a71b83	bridge_native_rtp: Fix audio issues when moving from remote bridge to softmix ... When a native RTP bridge that is remotely bridging its participants switches to a softmix bridge, it may not properly re-INVITE the media for one or both participants back to Asterisk. This is due to the current bridge_native_rtp code only re-INVITEs if it believes the channel will survive the bridge operation. Currently, that code is failing, as it expects the channels to have a soft hangup flag set on it indicating that a redirect has occurred or that the channel is going to leave the bridge. (The code did not take into account a smart bridge operation). This patch also renames a few things to be more reflective of the underlying types. Review: https://reviewboard.asterisk.org/r/3997/ ASTERISK-24327 #close ........ Merged revisions 425760 from http://svn.asterisk.org/svn/asterisk/branches/12 ........ Merged revisions 425761 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@425762 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Joshua Colp	98d5b7090d	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 ........ Merged revisions 425243 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@425244 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Richard Mudgett	02cf1835e3	bridge_softmix.c: Made use ao2_replace() instead of the inline equivalent. ... * Clarified some read/write format comments. * Fixed a doxygen tag typo. ........ Merged revisions 423423 from http://svn.asterisk.org/svn/asterisk/branches/13 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@423424 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
Jonathan Rose	af4cd65143	Channels: Masquerades to automatically move frame/audio hooks ... Whenever possible, audiohooks and framehooks will now be copied over to the channel that the masquerading channel gets cloned into. This should occur for all audiohooks and most framehooks. As a result, in Asterisk 12.5 and up, the AUDIOHOOK_INHERIT function is now deprecated and its behavior is essentially the new default for all audiohooks, plus some additional audiohooks/framehooks. Review: https://reviewboard.asterisk.org/r/3721/ ........ Merged revisions 418914 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@418936 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	20a14e568f	bridges/bridge_native_rtp: Reconfigure bridge on removal of framehook ... This patch is a re-do of r414122. When r414122 was merged, a major problem with it was uncovered. UNBRIDGE soft hangup flags have a catastrophic effect on the pbx core if they leak out from the bridge layer: the channel gets hung up. With the number of threads involved in a blind transfer, and with the initial patch, it was likely that this would occur. This caused a large number of test failures This patch is nearly identical with the one proposed in r414122, save for the following changes: - We explicitly clear the UNBRIDGE flag when setting an after goto on a channel in a bridge - Defensively, if we encounter an UNBRIDGE flag in the pbx core, we handle it https://reviewboard.asterisk.org/r/3585/ ........ Merged revisions 415443 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@415444 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Joshua Colp	962b78bca1	bridge_native_rtp: Take the bridge type choice of both channels into account. ... The bridge_native_rtp module currently uses the bridge result of the first channel that joins a bridge as the ultimate result. This means that if the first channel has direct media enabled but the second does not a direct media bridge will still occur. This change makes it so that both sides are taken into account. If either side forbids the bridge or responds with a local bridge result then either a generic or local bridge occurs. ASTERISK-23541 #close Reported by: Justin E Review: https://reviewboard.asterisk.org/r/3577/ ........ Merged revisions 414975 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@414976 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
Matthew Jordan	42a1dee02d	Undo r414123 ... The Test Suite caught a few problems, undoing until those are resolved git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@414138 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Matthew Jordan	17ff4d9282	bridge_native_rtp/bridge_channel: Fix direct media issues due to frame hook ... This patch fixes issues with direct media bridges that occur after a blind transfer. These issues were caught by the (currently failing) pjsip/transfers/blind_transfer/caller_direct_media test. The test currently fails primarily for two reasons: (1) When Bob and Charlie (the transfer target and the transfer destination) enter a bridge together, the framehook remains on the transfer target channel until both channels are in the bridge. As it consumes voice frames, the initial bridge type is a simple bridge. The framehook is removed when both channels are in the bridge; however, this does not currently cause the bridging framework to re-evaluate the bridge. This patch adds a AST_SOFTHANGUP_UNBRIDGE poke to the transfer target channel when a framehook is removed so the bridge can re-evaluate itself. (2) When a channel leaves a native RTP bridge, it may be leaving due to being hung up. Sending a re-INVITE to a channel that is about to be hung up is not nice - in fact, there's a good chance we'll send the BYE request before the channel has had a chance to send back a 200 OK. To be somewhat nicer, this patch adds a function to channel.h that allows the bridging framework to query for exactly why a channel is leaving a bridge via the channel's soft hangup flags. This allows it to only send the re-INVITE if there's a chance the channel will survive the native bridging experience. Review: https://reviewboard.asterisk.org/r/3535/ ........ Merged revisions 414122 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@414123 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Joshua Colp	d134150be2	framehooks: Add callback for determining if a hook is consuming frames of a specific type. ... In the past framehooks have had no capability to determine what frame types a hook is actually interested in consuming. This has meant that code has had to assume they want all frames, thus preventing native bridging. This change adds a callback which allows a framehook to be queried for whether it is consuming a frame of a specific type. The native RTP bridging module has also been updated to take advantange of this, allowing native bridging to occur when previously it would not. ASTERISK-23497 #comment Reported by: Etienne Lessard ASTERISK-23497 #close Review: https://reviewboard.asterisk.org/r/3522/ ........ Merged revisions 413681 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@413682 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Joshua Colp	e2ed86e4ca	Undoing framehook support. Issues were uncovered by Bamboo. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@413668 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Joshua Colp	3b3e4b9b95	framehooks: Add callback for determining if a hook is consuming frames of a specific type. ... In the past framehooks have had no capability to determine what frame types a hook is actually interested in consuming. This has meant that code has had to assume they want all frames, thus preventing native bridging. This change adds a callback which allows a framehook to be queried for whether it is consuming a frame of a specific type. The native RTP bridging module has also been updated to take advantange of this, allowing native bridging to occur when previously it would not. ASTERISK-23497 #comment Reported by: Etienne Lessard ASTERISK-23497 #close Review: https://reviewboard.asterisk.org/r/3522/ ........ Merged revisions 413650 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@413651 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
Russell Bryant	5b7a769fd8	(mix)monitor: Add options to enable a periodic beep ... Add an option to enable a periodic beep to be played into a call if it is being recorded. If enabled, it uses the PERIODIC_HOOK() function internally to play the 'beep' prompt into the call at a specified interval. This option is provided for both Monitor() and MixMonitor(). Review: https://reviewboard.asterisk.org/r/3424/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@412427 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Mark Michelson	d44aefeef4	Fix stuck channel in ARI through the introduction of synchronous bridge actions. ... Playing back a file to a channel in an ARI bridge would attempt to wait until the playback concluded before returning. The method used involved signaling the waiting thread in the ARI custom playback function. The problem with this is that there were some corner cases that were not accounted for: * If a bridge channel could not be found, then we never would attempt the playback but would still attempt to wait for the playback to complete. * If the bridge playfile action failed to queue, we would still attempt to wait for the playback to complete. * If the bridge playfile action were queued but some circumstance caused the playback not to occur (the bridge dies, the channel is removed from the bridge), then we would never be notified. The solution to this is to move the waiting logic into the bridge code. A new bridge API function is added to queue a synchronous action on a bridge. The waiting thread is notified when the queued frame has been freed, either due to an error occurring or due to successful playback. As a failsafe, the waiting thread has a 10 minute timeout just in case there is a frame leak somewhere. Review: https://reviewboard.asterisk.org/r/3338 ........ Merged revisions 410673 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@410684 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Kinsey Moore	b5f8f56bd0	bridge_native_rtp: Fix crash involving masquerade ... It is possible for a channel to be masqueraded out of a bridge which means it may no longer have RTP glue to check upon leaving said bridge. If this situation occurred (it's possible at least during dial and call pickup) then Asterisk would crash. This change makes sure the glue is checked before use. (closes issue AST-1290) Reported by: John Bigelow ........ Merged revisions 409900 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@409904 65c4cc65-6c06-0410-ace0-fbb531ad65f3	11 years ago
Kevin Harwell	84e1790beb	bridge_native_rtp: Deadlock during 4-way conference creation ... The change contains a slightly adjusted patch that was on the issue (submitted by kmoore). A fix was made by adding in a bridge lock while calling bridge_start/stop from the framehook callback. Since the framehook callback is not called from the bridging core the bridge is not locked, but needs to be before calling bridge_start. (closes issue ASTERISK-22749) Reported by: Kinsey Moore Review: https://reviewboard.asterisk.org/r/3066/ Patches: lock_inversion.diff uploaded by kmoore (license 6273) ........ Merged revisions 403767 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@403768 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kinsey Moore	6c417b0475	bridge_native_rtp: Ensure bridge is torn down ... When a bridge transitions away from one tech to another, the tech going away is provided a dummy bridge with no channels in it to tear down. Currently this means that the teardown code exits prematurely and does not tear anything down. This change tears down RTP bridging for the channel provided in the leave bridge tech callback. This also reverts the majority of r400403 since it is now redundant. (closes issue ASTERISK-22628) (closes issue ASTERISK-22676) Reported by: John Bigelow Reported by: Kevin Harwell Tested by: John Bigelow Review: https://reviewboard.asterisk.org/r/2905/ Patches: native_rtp_fix.diff uploaded by Kinsey Moore (License 6273) ........ Merged revisions 402148 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@402149 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	0ddcee5a46	Softmix: Fix crash when switching from softmix to another bridge technology. ... The crash is caused by a race condition when switching between native RTP and softmix bridging technologies. In this situation, the bridging technology is switched from native RTP to softmix, and then back to native RTP fast enough that the softmix private data gets destroyed before the softmix mixing thread gets started. Thanks to Kinsey Moore for the crash analysis. * Fix race condition when starting the softmix mixing thread and switching to another bridge technology. (closes issue ASTERISK-22678) Reported by: John Bigelow Patches: jira_asterisk_22678_v12.patch (license #5621) patch uploaded by rmudgett Tested by: John Bigelow ........ Merged revisions 400849 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@400850 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Mark Michelson	b11983d480	Fix assumption in bridge_native_rtp.c regarding number of participants in a bridge. ... When a party leaves a bridge, there may be more participants in the bridge than expected. As such, it is important not to make assumptions regarding the list of channels in a bridge. This change makes it so that when a party leaves a native RTP bridge, we unbridge it and the party it was bridged with. Previously, the first and last channels in the list were unbridged since it was assumed that these were the two channels that had been bridged. As previously stated, a new party had been inserted into the bridge, so this logic did not work properly. (closes issue ASTERISK-22615) reported by Matt Jordan Review: https://reviewboard.asterisk.org/r/2899 ........ Merged revisions 400403 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@400452 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Mark Michelson	ee21eee7e0	Cache string values of formats on ast_format_cap() to save processing. ... Channel snapshots have string representations of the channel's native formats. Prior to this change, the format strings were re-created on ever channel snapshot creation. Since channel native formats rarely change, this was very wasteful. Now, string representations of formats may optionally be stored on the ast_format_cap for cases where string representations may be requested frequently. When formats are altered, the string cache is marked as invalid. When strings are requested, the cache validity is checked. If the cache is valid, then the cached strings are copied. If the cache is invalid, then the string cache is rebuilt and copied, and the cache is marked as being valid again. Review: https://reviewboard.asterisk.org/r/2879 ........ Merged revisions 400356 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@400363 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Mark Michelson	212b6f668b	Fix refleaks of ast_rtp_instance structures. ... These refleaks were causing bridged calls not to close their RTP ports. Thus a call would leave open 4 ports (RTP for party A, RTCP for party A, RTP for party B, and RTCP for party B). This led to an eventual depletion of available RTP ports. ........ Merged revisions 399924 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@399925 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	656843dd34	Add a WARNING in bridge_softmix when a timing module isn't loaded ... If bridge_softmix fails to be created because no timing source is present in Asterisk, this will currently fail gracefully but with (most likely) a generic error message by whatever module tried to create the softmix bridge. This patch adds a more explicit warning so you can actually diagnose and fix the problem. Review: https://reviewboard.asterisk.org/r/2857/ ........ Merged revisions 399353 from http://svn.asterisk.org/svn/asterisk/branches/11 ........ Merged revisions 399365 from http://svn.asterisk.org/svn/asterisk/branches/12 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@399368 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Jonathan Rose	c13d0b7bdc	bridge_native_rtp: Fix hold chain bugs caused by native RTP bridge framehook ... Issuing hold/unhold would lead to odd behavior. Between two chan_sip devices, a hold could cause an endless chain of updates while with pjsip a similar chain would begin but then end somewhat randomly. This patch fixes that by no longer tweaking the RTP glue on both sides of the call for every HOLD/UNHOLD/UPDATE_RTP_PEER frame. (issue ASTERISK-22217) Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/2794/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397578 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	6ebfac8e70	Handle DTMF and hold wrapup when a channel leaves the bridging system. ... DTMF start/end and hold/unhold events have state because a DTMF begin event and hold event must be ended by something. The following cases need to be handled when a channel is moved around in the system. * When a channel leaves a bridge it may owe a DTMF end event to the bridge. * When a channel leaves a bridge it may owe an UNHOLD event to the bridge. (This case is explicitly ignored because things like transfers need explicit control over this.) * When a channel leaves the bridging system it may need to simulate a DTMF end event to the channel. * When a channel leaves the bridging system it may need to simulate an UNHOLD event to the channel. The patch also fixes the following: * Fixes playing a file and restarting MOH using the latest MOH class used. (closes issue ASTERISK-22043) Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/2791/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397577 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	c25c093c67	Minor tweaks with ast_moh_start() callers. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397494 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	477dea4661	Bridge API: Set a cause code on a channel when it is ejected from a bridge. ... The cause code needs to be passed from the disconnecting channel to the bridge peers if the disconnecting channel dissolves the bridge. * Made the call to an app_agent_pool agent disconnect with the busy cause code if the agent does not ack the call in time or hangs up before acking the call. (closes issue ASTERISK-22042) Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/2772/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397472 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	d213dfa30f	Fix several interrelated issues dealing with the holding bridge technology. ... * Added an option flags parameter to interval hooks. Interval hooks now can specify if the callback will affect the media path or not. * Added an option flags parameter to the bridge action custom callback. The action callback now can specify if the callback will affect the media path or not. * Made the holding bridge technology reexamine the participant idle mode option whenever the entertainment is restarted. * Fixed app_agent_pool waiting agents needlessly starting and stopping MOH every second by specifying the heartbeat interval hook as not affecting the media path. * Fixed app_agent_pool agent alert from restarting the MOH after the alert beep. The agent entertainment is now changed from MOH to silence after the alert beep. * Fixed holding bridge technology to defer starting the entertainment. It was previously a mixture of immediate and deferred. * Fixed holding bridge technology to immediately stop the entertainment. It was previously a mixture of immediate and deferred. If the channel left the bridging system, any deferred stopping was discarded before taking effect. * Miscellaneous holding bridge technology rework coding improvements. Review: https://reviewboard.asterisk.org/r/2761/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@397294 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	c3466db29d	Resolve some BUGBUG comments. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396783 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	20bf856ba4	bridge_native_rtp: Remove some unnecessary NULL checks on c1. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396512 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Jonathan Rose	2d87fc773b	bridge_holding: Add suspsend/unsuspend callbacks ... Suspend and unsuspend callbacks are added to the holding bridge so that entertainment can be disabled and re-enabled when operations would suspend a channel on the bridge (such as playback operations). This fixes entertainment so that when those operations end, the entertainment can pick back up and it also serves as an optimization. Also, this patch fixes a bug caused by triggering ringing frames immediately instead of pushing them to the queue which created a race condition where sometimes parking with ringing during attended transfers would cause the ringing to be interrupted by an unhold frame. (closes issue ASTERISK-22006) Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/2711/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@396189 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Joshua Colp	2feb049ba2	Fix hold/unhold in bridge_native_rtp, use tech_pvt instead of bridge_pvt, reduce bridging attempts, and fix breaking native RTP bridges. ... (closes issue ASTERISK-22128) (closes issue ASTERISK-22104) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@395866 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago