185 Commits (16.2)

Author	SHA1	Message	Date
sungtae kim	19fc99a2fb	Added ARI resource /ari/asterisk/ping ... Added ARI resource. GET /ari/asterisk/ping : It returns "pong" message with timestamp and asterisk id. It would be useful for simple heath check. Change-Id: I8d24e1dcc96f60f73437c68d9463ed746f688b29	8 years ago
George Joseph	603143bd5a	media_index.c: Refactored so it doesn't cache the index ... Testing revealed that the cache added no benefit but that it could consume excessive memory. Two new index related functions were created: ast_sounds_get_index_for_file() and ast_media_index_update_for_file() which restrict index updating to specific sound files. The original ast_sounds_get_index() and ast_media_index_update() calls are still available but since they no longer cache the results internally, developers should re-use an index they may already have instead of calling ast_sounds_get_index() repeatedly. If information for only a single file is needed, ast_sounds_get_index_for_file() should be called instead of ast_sounds_get_index(). The media_index directory scan code was elimininated in favor of using the existing ast_file_read_dirs() function. Since there's no more cache, ast_sounds_index_init now only registers the sounds cli commands instead of generating the initial index and subscribing to stasis format register/unregister messages. "sounds" is no longer a valid target for the "module reload" command. Both the sounds cli commands and the sounds ari resources were refactored to only call ast_sounds_get_index() once per invocation and to use ast_sounds_get_index_for_file() when a specific sound file is requested. Change-Id: I1cef327ba1b0648d85d218b70ce469ad07f4aa8d	8 years ago
Sebastian Damm	59cf552dd3	res/res_ari: Add additional hangup reasons ... The ARI DELETE /channels command takes a "reason" parameter Previously, there were only five reasons implemented This patch adds more reasons to choose from for more complex setups ASTERISK-28198 #close Change-Id: I85996f1076c9946d65c778413f040a845a90fecc	8 years ago
Corey Farrell	ed7a5664b6	astobj2: Eliminate usage of legacy ao2_container_alloc routine. ... Replace usage of ao2_container_alloc with ao2_container_alloc_hash or ao2_container_alloc_list. ao2_container_alloc is now restricted to modules only and is being removed from Asterisk 17. Change-Id: I0907d78bc66efc775672df37c8faad00f2f6c088	8 years ago
Sungtae Kim	cb83350230	res/res_ari: Fix null endpoint handle ... The res_ari(POST /channels/create handler) deos not check the endpoint parameter length. And it causes core dump. Fixed it to check the parameter length. Also fixed memory leak. ASTERISK-28169 Change-Id: Ibf10a9eb8a2e3a9ee1e13fbe748b2ecf955c3993	8 years ago
Moritz Fain	1f68614f3c	res_stasis: Fix stale data in ARI bridges ... Fixed an issue that resulted in "Allocation failed" each time an ARI request was made to start playing MOH on a bridge. In bridge_moh_create() we were attaching the after bridge callbacks to chan which is the ;1 channel of the unreal channel pair. We should have attached them to the ;2 channel which is pushed into the bridge by ast_unreal_channel_push_to_bridge(). The callbacks are called when the specific channel leaves the bridging system. Since the ;1 channel is never put into a bridge the callbacks never get called. The callbacks then never remove the moh_wrapper from the app_bridges_moh container. As a result we cannot find the channel associated with the wrapper to start MOH because it has hungup. This is the reason causing the reported issue. * Rather than using after bridge callbacks to cleanup, we now have moh_channel_thread() doing the cleanup when the channel hangs up. * Fixed moh_channel_thread() accumulating control frames on the stasis bridge MOH channel until MOH is stopped. Control frames are no longer accumulated while MOH is playing. * Fixed channel ref counting issue. stasis_app_bridge_moh_channel() may or may not return a channel ref. As a result ast_ari_bridges_start_moh() wouldn't know it may have a channel ref to release. stasis_app_bridge_moh_channel() will now return a ref with the channel it returns. * Eliminated RAII_VAR in bridge_moh_create(). ASTERISK-26094 #close Change-Id: Ibff479e167b3320c68aaabfada7e1d0ef7bd548c	8 years ago
Walter Doekes	78453e65fd	optional_api: Remove unused nonoptreq fields ... As they're not actively used, they only grow stale. The moduleinfo field itself is kept in Asterisk 13/15 for ABI compatibility. ASTERISK-28046 #close Change-Id: I8df66a7007f807840414bb348511a8c14c05a9fc	8 years ago
Corey Farrell	679fa5fb34	Add missing OPTIONAL_API and ARI dependences. ... I've audited all modules that include any header which includes asterisk/optional_api.h. All modules which use OPTIONAL_API now declare those dependencies in AST_MODULE_INFO using requires or optional_modules as appropriate. In addition ARI dependency declarations have been reworked. Instead of declaring additional required modules in res/ari/resource_*.c we now add them to an optional array "requiresModules" in api-docs for each module. This allows the AST_MODULE_INFO dependencies to include those missing modules. Change-Id: Ia0c70571f5566784f63605e78e1ceccb4f79c606	9 years ago
Sean Bright	fd0ca1c3f9	Remove as much trailing whitespace as possible. ... Change-Id: I873c1c6d00f447269bd841494459efccdd2c19c0	9 years ago
Corey Farrell	bf2d35931d	aco: Minimize use of regex. ... Remove nearly all use of regex from ACO users. Still remaining: * app_confbridge has a legitamate use of option name regex. * ast_sorcery_object_fields_register is implemented with regex, all callers use simple prefix based regex. I haven't decided the best way to fix this in both 13/15 and master. Change-Id: Ib5ed478218d8a661ace4d2eaaea98b59a897974b	9 years ago
Richard Mudgett	55c4d8e008	res_ari: Fix inverted test giving wrong error message. ... The patch for ASTERISK_24560 inverted a test checking if the bridge name is being updated to a different name. * Fix the test to return "Changing bridge name is not implemented" when someone attempts to change the bridge name. ASTERISK-27445 Change-Id: I4b70bf08b0e02e016108b077ff75b345dec12fc9	9 years ago
Jenkins2	759f4c1125	Merge "res_ari_events: Fix use after free / double-free of JSON message."	9 years ago
Corey Farrell	5247ba4b88	res_ari_events: Fix use after free / double-free of JSON message. ... When stasis_app_message_handler needs to queue a message for a later connection it needs to bump the message reference so it doesn't get freed when the caller releases it's reference. Change-Id: I82696df8fe723b3365c15c3f7089501da8daa892	9 years ago
Richard Mudgett	ee08f10d06	Fix ast_(v)asprintf() malloc failure usage conditions. ... When (v)asprintf() fails, the state of the allocated buffer is undefined. The library had better not leave an allocated buffer as a result or no one will know to free it. The most likely way it can return failure is for an allocation failure. If the printf conversion fails then you actually have a threading problem which is much worse because another thread modified the parameter values. * Made __ast_asprintf()/__ast_vasprintf() set the returned buffer to NULL on failure. That is much more useful than either an uninitialized pointer or a pointer that has already been freed. Many uses won't have to check for failure to ensure that the buffer won't be double freed or prevent an attempt to free an uninitialized pointer. * stasis.c: Fixed memory leak in multi_object_blob_to_ami() allocated by ast_asprintf(). * ari/resource_bridges.c:ari_bridges_play_helper(): Remove assignment to the wrong thing which is now not needed even if assigning to the right thing. Change-Id: Ib5252fb8850ecf0f78ed0ee2ca0796bda7e91c23	9 years ago
Joshua Colp	ac7ec5026c	Merge "Add missing menuselect dependencies."	9 years ago
Richard Mudgett	79ddcdbc70	Stasis/ARI: Fix off-nominal path json memory leaks. ... Change-Id: Id569c624c426e3b22a99936473c730592d8b83fb	9 years ago
Corey Farrell	606ae3484a	Add missing menuselect dependencies. ... This adds menuselect dependencies for modules that use symbols of other modules. ASTERISK-27390 Change-Id: Ia2d2849f5b87a72af7324a82edc3f283eafb5385	9 years ago
Corey Farrell	b616b7e4a9	res/ari/resource_bridges.h: Update from 'make ari-stubs'. ... A comment was updated when I ran 'make ari-stubs'. Change-Id: Ib5154ae3ad72aff53374c28ead540fe349c42175	9 years ago
Corey Farrell	f2175c5a39	res_ari_channels: Fix reference leak in channel_state_invalid. ... channel_state_invalid leaked a reference to the channel snapshot any time it was aquired. ASTERISK-27067 #close Change-Id: I8c653f00416b39978513c5605c4be0f03b1df29a	9 years ago
Seán C McCord	e8bde6916a	ari/bridge: Add mute, dtmf suppression controls ... Add bridge_features structure to bridge creation. Specifically, this implements mute and DTMF suppression, but others should be able to be easily added to the same structure. ASTERISK-27322 #close Reported by: Darren Sessions Sponsored by: AVOXI Change-Id: Id4002adfb65c9a8027ee9e1a5f477e0f01cf9d61	9 years ago
George Joseph	b55d21ad91	make ari-stubs so doc periodic jobs can run ... The periodic doc job does a make ari-stubs and checks that there are no changes before generating the docs. Since I changed the mustache template (and the generated code directly) recently and forgot to regenerate the stubs, the doc job thinks they're out of date. Change-Id: I94b97035311eccf52b0101b8590223265a7881d4	9 years ago
George Joseph	6691606723	ari: Implement 'debug all' and request/response logging ... The 'ari set debug' command has been enhanced to accept 'all' as an application name. This allows dumping of all apps even if an app hasn't registered yet. To accomplish this, a new global_debug global variable was added to res/stasis/app.c and new APIs were added to set and query the value. 'ari set debug' now displays requests and responses as well as events. This required refactoring the existing debug code. * The implementation for 'ari set debug' was moved from stasis/cli.{c,h} to ari/cli.{c,h}, and stasis/cli.{c,h} were deleted. * In order to print the body of incoming requests even if a request failed, the consumption of the body was moved from the ari stubs to ast_ari_callback in res_ari.c and the moustache templates were then regenerated. The body is now passed to ast_ari_invoke and then on to the handlers. This results in code savings since that template was inserted multiple times into all the stubs. An additional change was made to the ao2_str_container implementation to add partial key searching and a sort function. The existing cli code assumed it was already there when it wasn't so the tab completion was never working. Change-Id: Ief936f747ce47f1fb14035fbe61152cf766406bf (cherry picked from commit 1d890874f3)	10 years ago
zuul	d0474f6322	Merge "res/ari/resource_bridges: Add the ability to manipulate the video source"	10 years ago
Matt Jordan	a72ef38113	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	10 years ago
Sebastien Duthil	c6d755de11	res_ari: Add support for channel variables in ARI events. ... This works the same as for AMI manager variables. Set "channelvars=foo,bar" in your ari.conf general section, and then the channel variables "foo" and "bar" (along with their values), will appear in every Stasis websocket channel event. ASTERISK-26492 #close patches: ari_vars.diff submitted by Mark Michelson Change-Id: I5609ba239259577c0948645df776d7f3bc864229	10 years ago
Matt Jordan	c30d677333	res/stasis: Add CLI commands for displaying/debugging ARI apps ... This patch adds three new CLI commands: - ari show apps: list the registered ARI applications - ari show app: show detailed information about an ARI application - ari set debug: dump events being sent to an ARI application Note that while these CLI commands live in the res_stasis module, we use the 'ari' family for these commands. This was done as most users of Asterisk aren't aware of the semantic differences between ARI and res_stasis, and some 'ari' CLI commands already exist. ASTERISK-26488 #close Change-Id: I51ad6ff0cabee0d69db06858c13f18b1c513c9f5	10 years ago
Corey Farrell	a6e5bae3ef	Remove ASTERISK_REGISTER_FILE. ... ASTERISK_REGISTER_FILE no longer has any purpose so this commit removes all traces of it. Previously exported symbols removed: * __ast_register_file * __ast_unregister_file * ast_complete_source_filename This also removes the mtx_prof static variable that was declared when MTX_PROFILE was enabled. This variable was only used in lock.c so it is now initialized in that file only. ASTERISK-26480 #close Change-Id: I1074af07d71f9e159c48ef36631aa432c86f9966	10 years ago
Mark Michelson	3bd76dd679	ARI: Add duplicate channel ID checking for channel creation. ... This is similar to what is done for origination, but for the 14 and up channel creation method. When attempting to create a channel, if a channel ID is specified and a channel already exists with that ID, then a 409 is returned. Change-Id: I77f9253278c6947939c418073b6b31065489187c	10 years ago
Mark Michelson	e459b8dadf	ARI: Detect duplicate channel IDs ... ARI and AMI allow for an explicit channel ID to be specified when originating channels. Unfortunately, there is nothing in place to prevent someone from using the same ID for multiple channels. Further complicating things, adding ID validation to channel allocation makes it impossible for ARI to discern why channel allocation failed, resulting in a vague error code being returned. The fix for this is to institute a new method for channel errors to be discerned. The method mirrors errno, in that when an error occurs, the caller can consult the channel errno value to determine what the error was. This initial iteration of the feature only introduces "unknown" and "channel ID exists" errors. However, it's possible to add more errors as needed. ARI uses this feature to determine why channel allocation failed and can return a 409 error during origination to show that a channel with the given ID already exists. ASTERISK-26421 Change-Id: Ibba7ae68842dab6df0c2e9c45559208bc89d3d06	10 years ago
Joshua Colp	f31772ec20	ari: Update model validator based on addition of asterisk_id. ... ASTERISK-26470 Change-Id: I9c386f7a1c7d969161b28f189eb6298bbc5b7541	10 years ago
Matt Jordan	dd5129d84a	res/ari: Add the Asterisk EID field to outgoing events ... This patch adds the Asterisk EID field to all outgoing ARI events. Because this field should be added to all events as they are transmitted, it is appended to the JSON message just prior to it being handed off to the application message handler. This makes it somewhat resilient to both new events being added to ARI, as well as other potential event transport mechanisms. ASTERISK-26470 #close Change-Id: Ieff0ecc24464e83f3f44e9c3e7bd9a5d70b87a1d	10 years ago
Jean Aunis	91993ebaa5	resource_channels.c: add hangup reason "answered_elsewhere". ... In ARI, the channels API allows to hangup a channel with a hangup reason. This commit adds a new reason "answered_elsewhere". When using a SIP channel, this will eventually allow Asterisk to add a proper "Reason" header to a CANCEL message. ASTERISK-26321 Change-Id: Ia97675bd4acd6a7f58eb467953dfb94559f6583d	10 years ago
Kevin Harwell	e711e57106	resource_channels: Sync with ARI stubs ... This file was out of sync with the current ARI definitions. Change-Id: Ie7cb7d6d3c2eeb9cc9d683ca87b43b117e713d0a	10 years ago
George Joseph	886f2cab23	rest_api/channels: Fix multiple issues with create and dial ... * We weren't properly subscribing to the channel and it's originator on create. * We weren't doing a publish_dial after calling ast_call on dial. * We weren't calling depart_bridge when a channel left the dial bridge. The first 2 issues were causing events to not be generated and the third was actually causing channels to not get properly destroyed when hung up. Together these 3 issues were causing the new rest_apichannels/create_dial_bridge tests to fail. As a result of the fixes, the cdr state machine had to be slightly tweaked to allow bridge leave events without asserting and the tests themselves had to be updated to account for the channels now cleaning themselves up. Change-Id: Ibf23abf5a62de76e82afb4461af5099c961b97d8	10 years ago
Mark Michelson	11caa10cf5	ARI: Ensure announcer channels are destroyed. ... Announcer channels were not being destroyed because the stasis_app_control structure that referenced them was not being destroyed. The control structure was not being destroyed because it was not being unlinked from its container. It was not being unlinked from its container because the after bridge callback for the announcer channel was not being run. The after bridge callback was not being run because the after bridge datastore was not being removed from the channel on destruction. The channel was not being destroyed because the hangup that used to destroy the channel was now only reducing the reference count to one. The reference count of the channel was only being reduced to one because the stasis_app_control structure was holding the final reference... The control structure used to not keep a reference to the channel, so that loop described above did not happen. The solution is to manually remove the control structure from its container when the playback on a bridge is complete. ASTERISK-26083 #close Reported by Joshua Colp Change-Id: I0ddc0f64484ea0016245800b409b567dfe85cfb4	10 years ago
zuul	39e6d80937	Merge "ARI: Ensure proper channel state on operations."	10 years ago
Mark Michelson	1fd3a7849e	ARI: Ensure proper channel state on operations. ... ARI was recently outfitted with operations to create and dial channels. This leads to the ability to try funny stuff. You could create a channel and then immediately try to play back media on it. You could create a channel, dial it, and while it is ringing attempt to make it continue in the dialplan. This commit attempts to fix this by adding a channel state check to operations that should not be able to operate on outbound channels that have not yet answered. If a channel is in an invalid state, we will send a 412 response. ASTERISK-26047 #close Reported by Mark Michelson Change-Id: I2ca51bf9ef2b44a1dc5a73f2d2de35c62c37dfd8	10 years ago
Joshua Colp	643dd462ee	Merge "ari/resource_channels: Add 'formats' to channel create/originate"	10 years ago
George Joseph	a2f820e8dc	ari/resource_channels: Add 'formats' to channel create/originate ... If you create a local channel and don't specify an originator channel to take capabilities from, we automatically add all audio formats to the new channel's capabilities. When we try to make the channel compatible with another, the "best format" functions pick the best format available, which in this case will be slin192. While this is great for preserving quality, it's the worst for performance and overkill for the vast majority of applications. In the absense of any other information, adding all formats is the correct thing to do and it's not always possible to supply an originator so a new parameter 'formats' has been added to the channel create/originate functions. It's just a comma separated list of formats to make availalble for the channel. Example: "ulaw,slin,slin16". 'formats' and 'originator' are mutually exclusive. To facilitate determination of format names, the format name has been added to "core show codecs". ASTERISK-26070 #close Change-Id: I091b23ecd41c1b4128d85028209772ee139f604b	10 years ago
zuul	84a93b0d67	Merge "ARI: Re-implement the ARI dial command, allowing for early bridging."	10 years ago
Mark Michelson	88d997913f	ARI: Re-implement the ARI dial command, allowing for early bridging. ... ARI dial had been implemented using the Dial API. This made great sense when dialing was 100% separate from bridging. However, if a channel were to be added to a bridge during the dial attempt, there would be a conflict between the dialing thread and the bridging thread. Each would be attempting to read frames from the dialed channel and act on them. The initial attempt to make the two play nice was to have the Dial API suspend the channel in the bridge and stay in charge of the channel until the dial was complete. The problem with this was that it was riddled with potential race conditions. It also was not well-suited for the case where the channel changed which bridge it was in during the dial. This new approach removes the use of the Dial API altogether. Instead, the channel we are dialing is placed into an invisible ARI dialing bridge. The bridge channel thread handles incoming frames from the channel. If the channel is added to a real bridge, it is departed from the invisible bridge and then added to the real bridge. Similarly, if the channel is removed from the real bridge, it is automatically added back to the invisible bridge if the dial attempt is still active. This approach keeps the threading simple by always having the channel being handled by bridge channel threads. ASTERISK-25925 Change-Id: I7750359ddf45fcd45eaec749c5b3822de4a8ddbb	10 years ago
Matt Jordan	e773e3a9bb	ARI: Add the ability to download the media associated with a stored recording ... This patch adds a new feature to ARI that allows a client to download the media associated with a stored recording. The new route is /recordings/stored/{name}/file, and transmits the underlying binary file using Asterisk's HTTP server's underlying file transfer facilities. Because this REST route returns non-JSON, a few small enhancements had to be made to the Python Swagger generation code, as well as the mustache templates that generate the ARI bindings. ASTERISK-26042 #close Change-Id: I49ec5c4afdec30bb665d9c977ab423b5387e0181	10 years ago
Matt Jordan	03d88b5656	ARI: Add the ability to play multiple media URIs in a single operation ... Many ARI applications will want to play multiple media files in a row to a resource. The most common use case is when building long-ish IVR prompts made up of multiple, smaller sound files. Today, that requires building a small state machine, listening for each PlaybackFinished event, and triggering the next sound file to play. While not especially challenging, it is tedious work. Since requiring developers to write tedious code to do normal activities stinks, this patch adds the ability to play back a list of media files to a resource. Each of the 'play' operations on supported resources (channels and bridges) now accepts a comma delineated list of media URIs to play. A single Playback resource is created as a handle to the entire list. The operation of playing a list is identical to playing a single media URI, save that a new event, PlaybackContinuing, is raised instead of a PlaybackFinished for each non-final media URI. When the entire list is finished being played, a PlaybackFinished event is raised. In order to help inform applications where they are in the list playback, the Playback resource now includes a new, optional attribute, 'next_media_uri', that contains the next URI in the list to be played. It's important to note the following: - If an offset is provided to the 'play' operations, it only applies to the first media URI, as it would be weird to skip n seconds forward in every media resource. - Operations that control the position of the media only affect the current media being played. For example, once a media resource in the list completes, a 'reverse' operation on a subsequent media resource will not start a previously completed media resource at the appropiate offset. - This patch does not add any new operations to control the list. Hopefully, user feedback and/or future patches would add that if people want it. ASTERISK-26022 #close Change-Id: Ie1ea5356573447b8f51f2e7964915ea01792f16f	10 years ago
Sean Bright	14938184a3	res_ari: Correct Location headers returned by some ARI resources ... The Location headers returned by: * /bridges/{bridgeId}/play * /bridges/{bridgeId}/record * /channels/{channelId}/play * /channels/{channelId}/record Did not have the '/ari' prefix, and in the case of the 'play' resources, were using 'playback' instead of 'playbacks.' Change-Id: I957c58a3a1471bf477dae7c67faa1b74fcd9241c	10 years ago
Joshua Colp	72ef79dc2d	Merge "ARI: Add method to Dial a created channel."	11 years ago
Joshua Colp	3b71f09bb7	Merge "ARI: Add method to create a new channel."	11 years ago
Mark Michelson	abbb2edd4c	ARI: Add method to Dial a created channel. ... This adds a new ARI method that allows for you to dial a channel that you previously created in ARI. By combining this with the create method for channels, it allows for a workflow where a channel can be created, manipulated, and then dialed. The channel is under control of the ARI application during all stages of the Dial and can even be manipulated based on channel state changes observed within an ARI application. The overarching goal for this is to eventually be able to add a dialed channel to a Stasis bridge earlier than the "Up" state. However, at the moment more work is needed in the Dial and Bridge APIs in order to facilitate that. ASTERISK-25889 #close Change-Id: Ic6c399c791e66c4aa52454222fe4f8b02483a205	11 years ago
Mark Michelson	dd48d60c5b	ARI: Add method to create a new channel. ... This adds a new ARI method to the channels resource that allows for the creation of a new channel. The channel is created and then placed into the specified Stasis application. This is different from the existing originate method that creates a channel, dials it, and then places the answered channel into the dialplan or a Stasis application. This method does not attempt to call the channel at all. Dialing is left as a later step after channel creation. This allows for pre-dialing channel manipulation if desired. ASTERISK-25889 Change-Id: I3c96a0aba914b08e39f6256371a5bd4c92cbded8	11 years ago
zuul	e2a8033d09	Merge "res_stasis: Add control ref to playback and recording structs."	11 years ago
Richard Mudgett	0ea742d33a	res_stasis: Add control ref to playback and recording structs. ... The stasis_app_playback and stasis_app_recording structs need to have a struct stasis_app_control ref. Other threads can get a reference to the playback and recording structs from their respective global container. These other threads can then use the control pointer they contain after the control struct has gone. * Add control ref to stasis_app_playback and stasis_app_recording structs. With the refs added, the control command queue can now have a circular control reference which will cause the control struct to never get released if the control's command queue is not flushed when the channel leaves the Stasis application. Also the command queue needs better protection from adding commands if the control->is_done flag is set. * Flush the control command queue on exit. ASTERISK-25882 #close Change-Id: I3cf1fb59cbe6f50f20d9e35a2c07ac07d7f4320d	11 years ago