258 Commits (834d022da58324503f70a6cea9dadf235e1b417f)

Author	SHA1	Message	Date
Torrey Searle	4661c08549	chan_pjsip: add a flag to ignore 183 responses if no SDP present ... chan_sip will always ignore 183 responses that do not contain SDP however, chan_pjsip will currently always translate it into a 183 with SDP. This new flag allows chan_pjsip to have the same behavior as chan_sip. ASTERISK-28322 #close Change-Id: If81cfaa17c11b6ac703e3d71696f259d86c6be4a	7 years ago
Sean Bright	7b02a9617c	samples: Fix comment typo in pjsip.conf.sample ... Change-Id: I84a45c3d9fd26ca61aca99927eec83b57f1de857	7 years ago
George Joseph	c2adeb9dc2	taskprocessor: Enable subsystems and overload by subsystem ... To prevent one subsystem's taskprocessors from causing others to stall, new capabilities have been added to taskprocessors. * Any taskprocessor name that has a '/' will have the part before the '/' saved as its "subsystem". Examples: "sorcery/acl-0000006a" and "sorcery/aor-00000019" will be grouped to subsystem "sorcery". "pjsip/distributor-00000025" and "pjsip/distributor-00000026" will bn grouped to subsystem "pjsip". Taskprocessors with no '/' have an empty subsystem. * When a taskprocessor enters high-water alert status and it has a non-empty subsystem, the subsystem alert count will be incremented. * When a taskprocessor leaves high-water alert status and it has a non-empty subsystem, the subsystem alert count will be decremented. * A new api ast_taskprocessor_get_subsystem_alert() has been added that returns the number of taskprocessors in alert for the subsystem. * A new CLI command "core show taskprocessor alerted subsystems" has been added. * A new unit test was addded. REMINDER: The taskprocessor code itself doesn't take any action based on high-water alerts or overloading. It's up to taskprocessor users to check and take action themselves. Currently only the pjsip distributor does this. * A new pjsip/global option "taskprocessor_overload_trigger" has been added that allows the user to select the trigger mechanism the distributor uses to pause accepting new requests. "none": Don't pause on any overload condition. "global": Pause on ANY taskprocessor overload (the default and current behavior) "pjsip_only": Pause only on pjsip taskprocessor overloads. * The core pjsip pool was renamed from "SIP" to "pjsip" so it can be properly grouped into the "pjsip" subsystem. * stasis taskprocessor names were changed to "stasis" as the subsystem. * Sorcery core taskprocessor names were changed to "sorcery" to match the object taskprocessors. Change-Id: I8c19068bb2fc26610a9f0b8624bdf577a04fcd56	7 years ago
Joshua Colp	54a912b26d	res_odbc: Add basic query logging. ... When Asterisk is connected and used with a database the response time of the database can cause problems in Asterisk if it is long. Normally the only way to see this problem would be to retrieve a backtrace from Asterisk and examine where things are blocked, or examine the database to see if there is any indication of a problem. This change adds some basic query logging to make it easier to investigate such a problem. When logging is enabled res_odbc will now keep track of the number of queries executed, as well as the query that has taken the longest time to execute. There is also an option which will cause a WARNING message to be output if a query takes longer than a configurable amount of time to execute. This makes it easier and clearer for users that their database may be experiencing a problem that could impact Asterisk. ASTERISK-28277 Change-Id: I173cf4928b10754478a6a8c27dfa96ede0f058a6	7 years ago
Kevin Harwell	0bcaadc037	codecs.conf.sample: update codec opus docs ... The option value "sdp" for some of the settings was removed a while back, however the sample conf was not updated. This patch removes any wording with regards to the old "sdp" option value, and adjusts the defaults to what they are now. ASTERISK-28263 Change-Id: I41bfa44e9f69446bcc5c8fd92e3675c676fdc445	7 years ago
George Joseph	c6980e32ae	app_voicemail: Add Mailbox Aliases ... You can now define an "aliases" context in voicemail.conf whose entries point to actual mailboxes. These can be used anywhere the mailbox is specified. Example: [general] aliasescontext = myaliases [default] 1234 = yadayada [myaliases] 4321@devices = 1234@default Now you can use 4321@devices to refer to the 1234@default mailbox. This can be useful to provide channel drivers with constant mailbox specifications such as <extension>@devices leaving app_voicemail to control exactly which mailbox the alias points to. Now, only voicemail has to be reloaded to make changes instead of individual channel drivers which are usually more expensive to reload. Change-Id: I395b9205c91523a334fe971be0d1de4522067b04	7 years ago
Alexei Gradinari	f0546d1d87	res_pjsip: add option to enable ContactStatus event when contact is updated ... The commit I2f97ebfa79969a36a97bb7b9afd5b6268cf1a07d removed sending out the ContactStatus AMI event when a contact is updated. Thist change broke things which rely on old behavior. This patch adds a new PJSIP global configuration option 'send_contact_status_on_update_registration' to be able to preserve old ContactStatus behavior. By default new behavior, i.e. the ContactStatus event will not be sent when a device refreshes its registration. Change-Id: I706adf7584e7077eb6bde6d9799ca408bc82ce46	7 years ago
George Joseph	4f0bf0270e	Revert "app_voicemail: Remove need to subscribe to stasis" ... This reverts commit 29115e2384. That commit closed a long standing hole which allowed subscriptions to mailboxes that weren't configured in voicemail.conf. This caused an issue with FreePBX which depdended on that behavior. The commit is being reverted until FreePBX can handle the new behavior. ASTERISK-28151 Reported by: Ronald Raikes Change-Id: I57b7b85e75d7dd97c742b5c69d718a0f61260c15	7 years ago
Joshua Colp	50ac85cb40	stasis: Segment channel snapshot to reduce creation cost. ... When a channel snapshot was created it used to be done from scratch, copying all data (many strings). This incurs a cost when doing so. This change segments the channel snapshot into different components which can be reused if unchanged from the previous snapshot creation, reducing the cost. In normal cases this results in some pointers being copied with reference count being bumped, some integers being set, and a string or two copied. The other benefit is that it is now possible to determine if a channel snapshot update is redundant and thus stop it before a message is published to stasis. The specific segments in the channel snapshot were split up based on whether they are changed together, how often they are changed, and their general grouping. In practice only 1 (or 0) of the segments actually get changed in normal operation. Invalidation is done by setting a flag on the channel when the segment source is changed, forcing creation of a new segment when the channel snapshot is created. ASTERISK-28119 Change-Id: I5d7ef3df963a88ac47bc187d73c5225c315f8423	7 years ago
George Joseph	26810197c7	Merge "pjsip: new endpoint's options to control Connected Line updates"	7 years ago
Alexei Gradinari	eee935983b	pjsip: new endpoint's options to control Connected Line updates ... This patch adds new options 'trust_connected_line' and 'send_connected_line' to the endpoint. The option 'trust_connected_line' is to control if connected line updates are accepted from this endpoint. The option 'send_connected_line' is to control if connected line updates can be sent to this endpoint. The default value is 'yes' for both options. Change-Id: I16af967815efd904597ec2f033337e4333d097cd	7 years ago
Corey Farrell	90a11c4ae7	chan_sip deprecation. ... This officially deprecates chan_sip in Asterisk 17+. A warning is printed upon startup or module load to tell users that they should consider migrating. chan_sip is still built by default but the default modules.conf skips loading it at startup. Very important to note we are not scheduling a time where chan_sip will be removed. The goal of this change is to accurately inform end users of the current state of chan_sip and encourage movement to the fully supported chan_pjsip. Change-Id: Icebd8848f63feab94ef882d36b2e99d73155af93	7 years ago
Joshua Colp	bf5bb7831f	Merge "modules.conf.sample: Update preload usage documentation."	7 years ago
Richard Mudgett	96d5e444f0	modules.conf.sample: Update preload usage documentation. ... Change-Id: Id449d4435c38148b56ac4cfd61ae4d90ac66bb90	7 years ago
Nick French	37b2e68628	res_pjsip: Implement additional SIP RFCs for Google Voice trunk compatability ... This change implements a few different generic things which were brought on by Google Voice SIP. 1. The concept of flow transports have been introduced. These are configurable transports in pjsip.conf which can be used to reference a flow of signaling to a target. These have runtime configuration that can be changed by the signaling itself (such as Service-Routes and P-Preferred-Identity). When used these guarantee an individual connection (in the case of TCP or TLS) even if multiple flow transports exist to the same target. 2. Service-Routes (RFC 3608) support has been added to the outbound registration module which when received will be stored on the flow transport and used for requests referencing it. 3. P-Associated-URI / P-Preferred-Identity (RFC 3325) support has been added to the outbound registration module. If a P-Associated-URI header is received it will be used on requests as the P-Preferred-Identity. 4. Configurable outbound extension support has been added to the outbound registration module. When set the extension will be placed in the Supported header. 5. Header parameters can now be configured on an outbound registration which will be placed in the Contact header. 6. Google specific OAuth / Bearer token authentication (draft-ietf-sipcore-sip-authn-02) has been added to the outbound registration module. All functionality changes are controlled by pjsip.conf configuration options and do not affect non-configured pjsip endpoints otherwise. ASTERISK-27971 #close Change-Id: Id214c2d1c550a41fcf564b7df8f3da7be565bd58	7 years ago
Sean Bright	90bd8371f2	samples: PARKINGSLOT -> PARKING_SPACE in parking sample config ... PARKINGSLOT was deprecated in Asterisk 12 but the sample config was not updated to reflect that. Change-Id: I3e087c19d9ee587094fa5304102d8084a79c2b3c	7 years ago
Ben Ford	b11a6643cf	res_rtp_asterisk.c: Add "seqno" strictrtp option ... When networks experience disruptions, there can be large gaps of time between receiving packets. When strictrtp is enabled, this created issues where a flood of packets could come in and be seen as an attack. Another option - seqno - has been added to the strictrtp option that ignores the time interval and goes strictly by sequence number for validity. Change-Id: I8a42b8d193673899c8fc22fe7f98ea87df89be71	7 years ago
George Joseph	cdece3b637	app_voicemail: Remove need to subscribe to stasis ... app_voicemail was using the stasis cache to build and maintain a list of mailboxes that had subscribers. It then used this list to determine if a mailbox should be polled for new messages if polling was enabled. For this to work, stasis had to cache every subscription and unsubscription to the mailbox which caused a lot of overhead, both cpu and memory related. Since polling is only required when changes are being made to mailboxes outside of app_voicemail and since the number of mailboxes that don't have any subscribers is likely to be very low, all mailboxes are now polled instead of just the ones with subscribers. This paves the way for disabling the caching of stasis subscription change messages. Also fixed cleanup in some of the unit tests that not only left test users in the users list but also caused segfaults if the tests were run more than once. ASTERISK-27121 Change-Id: I5cceb737246949f9782955c64425b8bd25a9e9ee	7 years ago
Matthew Fredrickson	c8bacd45f1	sample_configs: noload res_hep.so by default ... Change disables loading of res_hep.so in default installation. Loading res_hep has a performance impact whether it's used or not. This disables loading of it in sample config files. Change-Id: I5ec150cf941634fabc72973e5bf1a965cb0ef9d0	7 years ago
Richard Mudgett	c035d0afe0	pbx_dundi: Update sample config documentation. ... Change-Id: I33d0ad0611c2124ca3440f0f811fa0f45e4e2849	7 years ago
Corey Farrell	63ca367ab9	Sample configs: Fix pjsip.conf syntax error. ... It is valid for a config file to be empty or contain only comments, but not valid for a config value to be set when no uncommented context exists. This caused an error to be loged numerous times during start when loading the default pjsip.conf. Change-Id: Icf3b0d69b4ecb6e935eecd43c99ed8b32a5a1cf6	7 years ago
Richard Mudgett	68a3d39a99	pjsip_wizard.conf.sample: Update remote_hosts description. ... Remove the note that SRV records are not supported as that is no longer true. ASTERISK-27993 Change-Id: Id0dd6ef40e52702be9727a2b6122216cb00bb4ca	7 years ago
Richard Mudgett	e6bb2efaab	res_pjsip: Update endpoint transport option documentation. ... Change-Id: I5394fdff6a296efc8e1695a156e616acd932ae52	7 years ago
George Joseph	8f42447c68	res_pjsip: Add 'suppress_q850_reason_headers' option to endpoint ... A new option 'suppress_q850_reason_headers' has been added to the endpoint object. Some devices can't accept multiple Reason headers and get confused when both 'SIP' and 'Q.850' Reason headers are received. This option allows the 'Q.850' Reason header to be suppressed. The default value is 'no'. ASTERISK-27949 Reported-by: Ross Beer Change-Id: I54cf37a827d77de2079256bb3de7e90fa5e1deb1	7 years ago
Joshua Colp	de5144e751	pjsip: Clarify certificate configuration for Websocket. ... The Websocket transport uses the built-in HTTP server. As a result the TLS configuration is done in http.conf and not in pjsip.conf. This change adds a warning if this is configured in pjsip.conf and also clarifies in the sample configuration file. Change-Id: I187d994d328c3ed274b6754fd4c2a4955bdc6dd9	7 years ago
George Joseph	880fbff6b7	res_pjsip_session: Add ability to accept multiple sdp answers ... pjproject by default currently will follow media forked during an INVITE on outbound calls if the To tag is different on a subsequent response as that on an earlier response. We handle this correctly. There have been reported cases where the To tag is the same but we still need to follow the media. The pjproject patch in this commit adds the capability to sip_inv and also adds the capability to control it at runtime. The original "different tag" behavior was always controllable at runtime but we never did anything with it and left it to default to TRUE. So, along with the pjproject patch, this commit adds options to both the system and endpoint objects to control the two behaviors, and a small logic change to session_inv_on_media_update in res_pjsip_session to control the behavior at the endpoint level. The default behavior for "different tags" remains the same at TRUE and the default for "same tag" is FALSE. Change-Id: I64d071942b79adb2f0a4e13137389b19404fe3d6 ASTERISK-27936 Reported-by: Ross Beer	7 years ago
George Joseph	e7a7506f9c	app_confbridge: Enable sending events to participants ... ConfBridge can now send events to participants via in-dialog MESSAGEs. All current Confbridge events are supported, such as ConfbridgeJoin, ConfbridgeLeave, etc. In addition to those events, a new event ConfbridgeWelcome has been added that will send a list of all current participants to a new participant. For all but the ConfbridgeWelcome event, the JSON message contains information about the bridge, such as its id and name, and information about the channel that triggered the event such as channel name, callerid info, mute status, and the MSID labels for their audio and video tracks. You can use the labels to correlate callerid and mute status to specific video elements in a webrtc client. To control this behavior, the following options have been added to confbridge.conf: bridge_profile/enable_events: This must be enabled on any bridge where events are desired. user_profile/send_events: This must be set for a user profile to send events. Different user profiles connected to the same bridge can have different settings. This allows admins to get events but not normal users for instance. user_profile/echo_events: In some cases, you might not want the user triggering the event to get the event sent back to them. To prevent it, set this to false. A change was also made to res_pjsip_sdp_rtp to save the generated msid to the stream so it can be re-used. This allows participant A's video stream to appear as the same label to all other participants. Change-Id: I26420aa9f101f0b2387dc9e2fd10733197f1318e	7 years ago
George Joseph	2bf26ce5ac	ast_coredumper: Fix output directory and variable precedence ... The OUTPUTDIR variable in ast_debug_tools.conf.sample is now set to "/tmp" instead of "/some/directory". Variables set on the command line or that are already in the environment now take predecence over variables set in the config files. ASTERISK-27846 Reported by: Ted G Change-Id: Ie8baec52d531886bf5849ec1d59bb59dc87ad387	8 years ago
Tzafrir Cohen	6301531416	chan_dahdi: Configurable dialed digit timeouts ... Analog phones dial overlap dialing and it is chan_dahdi's job to read the numbers. It has three timeout constants that this commit converts to channel-level configuration options: * firstdigit_timeout: Default time (ms) to detect first digit * interdigit_timeout: Default time (ms) to detect following digits * matchdigit_timeout: Default time (ms) to wait in case of ambiguous match. This happens when the dialed digits match a number in the current context but are also the prefix of another number. Change-Id: Ib728fa900a4f6ae56d1ed810aba61b6593fb7213	8 years ago
Joshua Colp	8de3fa2b56	bridge_softmix / app_confbridge: Add support for REMB combining. ... This change adds the ability for multiple REMB reports in bridge_softmix to be combined according to a configured behavior into a single report. This single report is sent back to the sender of video, which adjusts the encoding bitrate to be at or below the bitrate of the report. The available behaviors are: lowest, highest, and average. Lowest uses the lowest received bitrate. Highest uses the highest received bitrate. Average goes through the received bitrates adding them to the previous average and creates a new average. Other behaviors can be added in the future and the existing average one may be adjusted, but this provides the foundation to do so. Support for configuring which behavior to use has been added to app_confbridge. ASTERISK-27804 Change-Id: I9eafe4e7c1f72d67074a8d6acb26bfcf19322b66	8 years ago
Joshua Colp	0f6431e8e4	app_confbridge / bridge_softmix: Add ability to configure REMB interval. ... This change adds a configuration option to app_confbridge which can be used to set the interval at which we will send a combined REMB (remote estimated maximum bitrate) frame to sources of video. The bridging API has also been extended slightly to allow setting this so bridge_softmix can use it. ASTERISK-27786 Change-Id: I0e49eae60f369c86434414f3cb8278709c793c82	8 years ago
George Joseph	5d097f8236	channel.c: Allow generic plc then channel formats are equal ... If the two formats on a channel are equal, we don't transcode and since the generic plc needs slin to work, it doesn't get invoked. * A new configuration option "genericplc_on_equal_codecs" was added to the "plc" section of codecs.conf to allow generic packet loss concealment even if no transcoding was originally needed. Transcoding via SLIN is forced in this case. ASTERISK-27743 Change-Id: I0577026a179dea34232e63123254b4e0508378f4	8 years ago
Jenkins2	3afc2c52af	Merge "core: Fix handling of maximum length lines in config files."	8 years ago
Richard Mudgett	1a36a452bd	pjproject: Add cache_pools debugging option. ... The pool cache gets in the way of finding use after free errors of memory pool contents. Tools like valgrind and MALLOC_DEBUG don't know when a pool is released because it gets put into the cache instead of being freed. * Added the "cache_pools" option to pjproject.conf. Disabling the option helps track down pool content mismanagement when using valgrind or MALLOC_DEBUG. The cache gets in the way of determining if the pool contents are used after free and who freed it. To disable the pool caching simply disable the cache_pools option in pjproject.conf and restart Asterisk. Sample pjproject.conf setting: [startup] cache_pools=no * Made current users of the caching pool factory initialization and destruction calls call common routines to create and destroy cached pools. ASTERISK-27704 Change-Id: I64d5befbaeed2532f93aa027a51eb52347d2b828	8 years ago
Corey Farrell	a7927471ad	core: Fix handling of maximum length lines in config files. ... When a line is the maximum length "\n" is found at sizeof(buf) - 2 since the last character is actually the null terminator. In addition if a line was exactly 8190 plus a multiple of 8192 characters long the config parser would skip the following line. Additionally fix comment in voicemail.conf sample config. It previously stated that emailbody can only contain up to 512 characters which is always wrong. The buffer is normally 8192 characters unless LOW_MEMORY is enabled then it is 512 characters. The updated comment states that the line can be up to 8190 or 510 characters since the line feed and NULL terminator each use a character. ASTERISK-26688 #close Change-Id: I80864a0d40d2e2d8cd79d72af52a8f0a3a99c015	8 years ago
Richard Mudgett	b9024197ab	app_confbridge: Update dsp_silence_threshold and dsp_talking_threshold docs. ... The dsp_talking_threshold does not represent time in milliseconds. It represents the average magnitude per sample in the audio packets. This is what the DSP uses to determine if a packet is silence or talking/noise. Change-Id: If6f939c100eb92a5ac6c21236559018eeaf58443	8 years ago
Corey Farrell	cf21e9fc97	Sample modules.conf: comment out example load statement. ... The sample modules.conf explicitly loaded res_musiconhold.so. This is redundent as autoload=yes is already set. It causes warnings if res_musiconhold.so was not installed and results in an unexpected load if the admin disables autoload without remembering to remove the res_musiconhold load statement. Also remove reference to unknown module pbx_gtkconsole. Change-Id: Ib01888994d9f1364b14d3c9fb6ff96774a6e580a	8 years ago
ghjm	86b484dec7	app_followme: Add a prompt to be read when a call is connected ... This patch adds the ability to configure a prompt which will be read to the "winner" who pressed 1 (or the configured value) and received the call. ASTERISK-24372 #close Change-Id: I6ec1c6c883347f7d1e1f597189544993c8d65272	8 years ago
Richard Mudgett	8494e78010	res_pjsip: Split type=identify to IP address and SIP header matching priorities ... The type=identify endpoint identification method can match by IP address and by SIP header. However, the SIP header matching has limited usefulness because you cannot specify the SIP header matching priority relative to the IP address matching. All the matching happens at the same priority and the order of evaluating the identify sections is indeterminate. e.g., If you had two type=identify sections where one matches by IP address for endpoint alice and the other matches by SIP header for endpoint bob then you couldn't predict which endpoint is matched when a request comes in that matches both. * Extract the SIP header matching criteria into its own "header" endpoint identification method so the user can specify the relative priority of the SIP header and the IP address matching criteria in the global endpoint_identifier_order option. The "ip" endpoint identification method now only matches by IP address. ASTERISK-27491 Change-Id: I9df142a575b7e1e3471b7cda5d3ea156cef08095	8 years ago
Richard Mudgett	8f3167c5f1	res_pjsip.c: Update the endpoint identification documentation. ... * Endpoint identify_by documentation. * IP/Header endpoint identifier documentation. Change-Id: Id92f00b495acca7be945daf749d2abd7f76a0b5a	8 years ago
Sean Bright	fd0ca1c3f9	Remove as much trailing whitespace as possible. ... Change-Id: I873c1c6d00f447269bd841494459efccdd2c19c0	8 years ago
Rodrigo Ramírez Norambuena	5335ad117d	app_queue: Add feature to set wrapuptime on the queue member ... This patch adds the ability to set the wrapuptime on the queue member config. When the option is set the wrapuptime on the queue member is used instead of the queue's wrapuptime. ASTERISK-27483 #close Change-Id: I11c85809537f974eb44dc5bbf82bcedd8a458902	8 years ago
Joshua Colp	1769d4a5c6	confbridge: Clarify mute sound documentation. ... The mute/unmute sounds are only played when the action is initiated using the DTMF menu. ASTERISK-24756 Change-Id: I55b3dd5bc166096bf5e2f547ddd0ce355f36e3dc	8 years ago
Jenkins2	c9bcd888a2	Merge "res_rtp_asterisk.c: Disable packet flood detection for video streams."	8 years ago
Richard Mudgett	98f7e9251f	res_rtp_asterisk.c: Disable packet flood detection for video streams. ... We should not do flood detection on video RTP streams. Video RTP streams are very bursty by nature. They send out a burst of packets to update the video frame then wait for the next video frame update. Really only audio streams can be checked for flooding. The others are either bursty or don't have a set rate. * Added code to selectively disable packet flood detection for video RTP streams. ASTERISK-27440 Change-Id: I78031491a6e75c2d4b1e9c2462dc498fe9880a70	8 years ago
Sean Bright	dac5e3a0df	configs: Comment out and change IP of iax.conf [demo] ... This no longer appears to exist, so no sense in causing confusion. ASTERISK-27175 #close Reported by: Tzafrir Cohen Change-Id: Idde967924c69f6a741dc9a5ab7dacb44d22cf100	8 years ago
Alexander Traud	e0354bbe82	res_rtp_asterisk: Correct default in sample configuration file. ... With Asterisk 12 (commit 866d968), the default of "icesupport" changed to - "yes" in the module "res_rtp_asterisk" and - "no" in the module "chan_sip". The latter was reflected in the sample configuration file for "sip.conf". The former did not make it into "rtp.conf.sample". ASTERISK-20643 Change-Id: I2a2e0a900455d0767a99ea576e30adc6d7608a36	8 years ago
Richard Mudgett	02a9952709	features.conf.sample: Clarify ActivatedBy documentation wording. ... Change-Id: Id2899331fe05d1909a862ea879742879d086bc64	8 years ago
Joshua Colp	ccbf176c30	Merge "ast_coredumper: Add ability to use directory other than /tmp"	8 years ago
Nir Simionovich	d995064fb7	This patch adds a beanstalk CEL backend. ... Beanstalkd is a simple to use job queue. It provides a means to create multiple job queues called "tubes". Each tube can store multiple jobs, with varying priorities with the queue. Queue processing is available via a simple TCP socket or via well defined libraries, avaialble at https://github.com/kr/beanstalkd/wiki/client-libraries This module is based upon the beanstalk-client library, available for download at: https://github.com/deepfryed/beanstalk-client This module currently doesn't support user defined events. Change-Id: Ic3a087faeeac045d69a2a018e60e29831ddb95ab	8 years ago