1136 Commits (77a5383957504d58889d4347ce72792aac6e51ea)

Author	SHA1	Message	Date
Asterisk Development Team	77a5383957	Update CHANGES and UPGRADE.txt for 13.35.0	6 years ago
Asterisk Development Team	07bf7890dc	Update CHANGES and UPGRADE.txt for 13.34.0	6 years ago
Asterisk Development Team	074d766fb3	Update CHANGES and UPGRADE.txt for 13.33.0	6 years ago
Asterisk Development Team	077cc24d66	Update CHANGES and UPGRADE.txt for 13.32.0	6 years ago
Asterisk Development Team	fe72f4e216	Update CHANGES and UPGRADE.txt for 13.31.0	7 years ago
Asterisk Development Team	51482dec75	Update CHANGES and UPGRADE.txt for 13.30.0	7 years ago
Asterisk Development Team	18fbd8d8c3	Update CHANGES and UPGRADE.txt for 13.29.0	7 years ago
Asterisk Development Team	b3d0754f82	Update CHANGES and UPGRADE.txt for 13.28.0	7 years ago
Asterisk Development Team	fba341af8b	Update CHANGES and UPGRADE.txt for 13.27.0	7 years ago
Ben Ford	251348cbbb	build: Revise CHANGES and UPGRADE.txt handling. ... This changes the way that we handle adding changes to CHANGES and UPGRADE.txt. The reason for this is because whenever someone needed to make a change to one of these files and someone else had already done so, you would run into merge conflicts. With this new setup, there will never be merge conflicts since all changes will be documented in the doc/<file>-staging directory. The release script is now responsible for merging all of these changes into the appropriate files. There is a special format that these files have to follow in order to be parsed. The files do not need to have a meaningful name, but it is strongly recommended. For example, if you made a change to pjsip, you may have something like this "res_pjsip_relative_title", where "relative_title" is something more descriptive than that. Inside each file, you will need a subject line for your change, followed by a description. There can be multiple subject lines. The file may look something like this: Subject: res_pjsip Subject: Core A description that explains the changes made and why. The release script will handle the bulleting and section separators! You can still separate with new lines within your description. The headers ("Subject" and "Master-Only") are case sensative, but the value for "Master-Only" ("true" or "True") is not. For more information, check out the wiki page: https://wiki.asterisk.org/wiki/display/AST/CHANGES+and+UPGRADE.txt ASTERISK-28111 #close Change-Id: I19cf4b569321c88155a65e9b0b80f6d58075dd47	7 years ago
Friendly Automation	4f6daa0aff	Merge "chan_pjsip: add a flag to ignore 183 responses if no SDP present" into 13	7 years ago
Torrey Searle	cbc704c5ec	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
Ben Ford	65170ba8f0	res_stasis: Add ability to switch applications. ... Added the ability to move between Stasis applications within Stasis. This can be done by calling 'move' in an application, providing (at minimum) the channel's id and the application to switch to. If the application is not registered or active, nothing will happen and the channel will remain in the current application, and an event will be triggered to let the application know that the move failed. The event name is "ApplicationMoveFailed", and provides the "destination" that the channel was attempting to move to, as well as the usual channel information. Optionally, a list of arguments can be passed to the function call for the receiving application. A full example of a 'move' call would look like this: client.channels.move(channelId, app, appArgs) The control object used to control the channel in Stasis can now switch which application it belongs to, rather than belonging to one Stasis application for its lifetime. This allows us to use the same control object instead of having to tear down the current one and create another. ASTERISK-28267 #close Change-Id: I43d12b10045a98a8d42541889b85695be26f288a	7 years ago
George Joseph	4aa55a8ca6	res_mwi_devstate.c: New module to allow presence subs to VM boxes ... This module allows presence subscriptions to voicemail boxes. This allows common BLF keys to act as voicemail waiting indicators. ASTERISK-28301 Change-Id: I62a246c24f3d7d432e33e22d7a4a57c15c292fdd	7 years ago
George Joseph	bae3fd04c1	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
Kevin Harwell	da93d17af8	ARI event type filtering ... Event type filtering is now enabled, and configurable per application. An app is now able to specify which events are sent to the application by configuring an allowed and/or disallowed list(s). This can be done by issuing the following: PUT /applications/{applicationName}/eventFilter And then enumerating the allowed/disallowed event types as a body parameter. ASTERISK-28106 Change-Id: I9671ba1fcdb3b6c830b553d4c5365aed5d588d5b	7 years ago
George Joseph	33ebaae23f	Merge "res_stasis: Auto-create context and extens on Stasis app launch." into 13	7 years ago
Joshua C. Colp	41da4ed8b2	Merge "media_index.c: Refactored so it doesn't cache the index" into 13	7 years ago
Ben Ford	26a04477f4	res_stasis: Auto-create context and extens on Stasis app launch. ... At AstriCon, there was a strong desire for the ability to completely bypass dialplan when using ARI. This is possible through the automatic creation of a context and a couple of extensions whenever an application is started. For example, if you have an application named 'ari-example', a context named 'stasis-ari-example' will be automatically created whenever this application is started as long as one does not already exist. Two extensions (a match-all extension for Stasis and a 'h' extension) are created within this context. Any endpoint that registers to Asterisk within this context will send all calls to the corresponding Stasis application. When the application is destroyed, the context is removed. ASTERISK-28104 #close Change-Id: Ie35bd93075e05b05e3ae129a83c9426931b7ebac	8 years ago
George Joseph	66982824bf	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. ast_sounds_reindex() is now a no-op but left for backwards compatibility. loader.c no longer registers "sounds" as a special reload target. 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
Friendly Automation	9514829403	Merge "app_voicemail: Add Mailbox Aliases" into 13	8 years ago
George Joseph	d0eccac3ff	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	8 years ago
Alexei Gradinari	4a8564cafa	res_pjsip: add option to disable ContactStatus event when contact is updated ... This patch adds a new PJSIP global configuration option 'send_contact_status_on_update_registration' to be able to have the same performance benefits as version 16. By default old behavior, i.e. the ContactStatus event will be sent when a device refreshes its registration. Change-Id: I706adf7584e7077eb6bde6d9799ca408bc82ce46	8 years ago
Sean Bright	92cc094d4e	bridge_builtin_features.c: Set auto(mix)mon variables on both channels ... This is how features behaved up through Asterisk 11, but was changed when the new bridging framework was implemented in Asterisk 12. Reported by rrittgarn in #asterisk. Change-Id: I72cf86223947a8118c75f46e2c603dbc11e3125b	8 years ago
George Joseph	0deaf81bff	backtrace: Refactor ast_bt_get_symbols so it doesn't crash ... We've been seeing crashes in libbfd when we attempt to generate a stack trace from multiple threads. It turns out that libbfd is NOT thread-safe. It can cache the bfd structure and give it to multiple threads without protecting itself. To get around this, we've added a global mutex around the bfd functions and also have refactored the use of those functions to be more efficient and to provide more information about inlined functions. Also added a few more tests to test_pbx.c. One just calls ast_assert() and the other calls ast_log_backtrace(). Neither are run by default. WARNING: This change necessitated changing the return value of ast_bt_get_symbols() from an array of strings to a VECTOR of strings. However, the use of this function outside Asterisk is not likely. ASTERISK-28140 Change-Id: I79d02862ddaa2423a0809caa4b3b85c128131621	8 years ago
Alexei Gradinari	b6d0fbda9d	pjsip: New function PJSIP_PARSE_URI to parse URI and return part of URI ... New dialplan function PJSIP_PARSE_URI added to parse an URI and return a specified part of the URI. This is useful when need to get part of the URI instead of cutting it using a CUT function. For example to get 'user' part of Remote URI ${PJSIP_PARSE_URI(${CHANNEL(pjsip,remote_uri)},user)} ASTERISK-28144 #close Change-Id: I5d828fb87f6803b6c1152bb7b44835f027bb9d5a	8 years ago
Corey Farrell	1709f6be77	pbx_config: Only the first [globals] section is seen. ... If multiple [globals] sections are used (for example via separate included files), only the first one is processed. This can result in lost global variables when using a modular extensions.conf. ASTERISK-28146 #close Change-Id: Iaac810c0a7c4d9b1bf8989fcc041cdb910ef08a0	8 years ago
Alexei Gradinari	bfe3821800	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	8 years ago
Ben Ford	aa31657e28	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	8 years ago
Richard Mudgett	8baedc2204	res_pjsip_endpoint_identifier_ip.c: Added regex support to match_header ... This patch adds regular expression support to make the identify section's match_header option more useful when attempting to match complex headers like the 'To' or 'From' headers. The 'From' header has variable components such as the tag parameter that you cannot predict. To specify a regular expression put slashes around the regular expression in place of the header value. [identify-alice] type=identify endpoint=alice match_header=From: /<sip:alice@127\\.0\\.0\\.1>/ * Added regex support to match_header so you could match a 'To' header among other complex headers. Fixed reported crashes when trying to match special headers like 'Contact'. The identify section's match_header method used code that assumed you were matching a generic header. Any other type of header could cause a crash if the header structure variant did not match the generic header enough. * Made use code that will work for any header type instead of code specific to generic headers. Other fixes while in the area: * Made check all headers of the requested name. * Added some more sanity checks to the configured identify matching options when applying the configuration. ASTERISK-27548 Change-Id: I27dfd4ff5e2259b906640e3c330681b76b4ed1f1	8 years ago
George Joseph	3470409dd6	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	8 years ago
George Joseph	06966e91fe	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	8 years ago
William McCall	9ff4779f03	app_confbridge: Add talking indicator for ConfBridgeList AMI response ... When an AMI client connects, it cannot determine if a user was talking prior to a transition in the user speaking state (which would generate a ConfbridgeTalking event). This patch causes app_confbridge to track the talking state and make this state available via ConfBridgeList. ASTERISK-27877 #close Change-Id: I19b5284f34966c3fda94f5b99a7e40e6b89767c6	8 years ago
George Joseph	2b50b31b68	Merge "app_sendtext: Enhance SendText to support Enhanced Messaging" into 13	8 years ago
George Joseph	c5d19565fe	Merge "bridge_softmix: Forward TEXT frames" into 13	8 years ago
George Joseph	3c2249fd37	app_sendtext: Enhance SendText to support Enhanced Messaging ... SendText now accepts new channel variables that can be used to override the To and From display names and set the Content-Type of a message. Since you can now set Content-Type, other text/* content types are now valid. Change-Id: I648b4574478119f95de09d9f08e9595831b02830	8 years ago
George Joseph	be7d4faed5	bridge_softmix: Forward TEXT frames ... Core bridging and, more specifically, bridge_softmix have been enhanced to relay received frames of type TEXT or TEXT_DATA to all participants in a softmix bridge. res_pjsip_messaging and chan_pjsip have been enhanced to take advantage of this so when res_pjsip_messaging receives an in-dialog MESSAGE message from a user in a conference call, it's relayed to all other participants in the call. res_pjsip_messaging already queues TEXT frames to the channel when it receives an in-dialog MESSAGE from an endpoint and chan_pjsip will send an MESSAGE when it gets a TEXT frame. On a normal point-to-point call, the frames are forwarded between the two correctly. bridge_softmix was not though so messages weren't getting forwarded to conference bridge participants. Even if they were, the bridging code had no way to tell the participants who sent the message so it would look like it came from the bridge itself. * The TEXT frame type doesn't allow storage of any meta data, such as sender, on the frame so a new TEXT_DATA frame type was added that uses the new ast_msg_data structure as its payload. A channel driver can queue a frame of that type when it receives a message from outside. A channel driver can use it for sending messages by implementing the new send_text_data channel tech callback and setting the new AST_CHAN_TP_SEND_TEXT_DATA flag in its tech properties. If set, the bridging/channel core will use it instead of the original send_text callback and it will get the ast_msg_data structure. Channel drivers aren't required to implement this. Even if a TEXT_DATA enabled driver uses it for incoming messages, an outgoing channel driver that doesn't will still have it's send_text callback called with only the message text just as before. * res_pjsip_messaging now creates a TEXT_DATA frame for incoming in-dialog messages and sets the "from" to the display name in the "From" header, or if that's empty, the caller id name from the channel. This allows the chat client user to set a friendly name for the chat. * bridge_softmix now forwards TEXT and TEXT_DATA frames to all participants (except the sender). * A new function "ast_sendtext_data" was added to channel which takes an ast_msg_data structure and calls a channel's send_text_data callback, or if that's not defined, the original send_text callback. * bridge_channel now calls ast_sendtext_data for TEXT_DATA frame types and ast_sendtext for TEXT frame types. * chan_pjsip now uses the "from" name in the ast_msg_data structure (if it exists) to set the "From" header display name on outgoing text messages. Change-Id: Idacf5900bfd5f22ab8cd235aa56dfad090d18489	8 years ago
Nathan Bruning	6a4afe09ce	res_pjsip_notify.c: enable in-dialog NOTIFY ... This patch adds support to send in-dialog SIP NOTIFY commands on chan_pjsip channels, similar to the functionality recently added for chan_sip (ASTERISK_27461). This extends res_pjsip_notify to allow for in-dialog messages. ASTERISK-27697 Change-Id: If7f3151a6d633e414d5dc319d5efc1443c43dd29	8 years ago
Alexander Traud	83353997f4	BuildSystem: Add support for building RADIUS with radcli. ... Radcli is yet another RADIUS client library, generally compatible with freeradius and radiusclient-ng. This commit adds autoconf option for detecting it as well and changes cdr_radius and cel_radius to use its header file in that case. ASTERISK-26540 Reported by: Tzafrir Cohen Change-Id: Icc056d476b7acf481309219e9abdca416866c6ec	8 years ago
George Joseph	373e7e3fb0	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
Richard Mudgett	104468ad3a	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
George Joseph	5947fd148b	AST-2018-005: res_pjsip_transport_management: Move to core ... Since res_pjsip_transport_management provides several attack mitigation features, its functionality moved to res_pjsip and this module has been removed. This way the features will always be available if res_pjsip is loaded. ASTERISK-27618 Reported By: Sandro Gauci Change-Id: I21a2d33d9dda001452ea040d350d7a075f9acf0d	8 years ago
Richard Mudgett	1ff580bb25	app_confbridge: ConfbridgeList event has standard channel shapshot headers. ... * Made the AMI ConfbridgeList action's ConfbridgeList events output all the standard channel snapshot headers instead of a few hand-coded channel snapshot headers. The benefit is that the CallerIDName gets disruptive characters like CR, LF, Tab, and a few others escaped. However, an empty CallerIDName is now output as "<unknown>" instead of "<no name>". ASTERISK-27651 Change-Id: Iaf7d54a9d40194c2db060bc9b4979fab6720d977	9 years ago
Richard Mudgett	0cf7a9e0ca	app_confbridge: Add the Muted header to ConfbridgeJoin AMI event. ... ASTERISK-27651 Change-Id: Idef2ca54d242d1b894efd3fc7b360bc6fd5bdc34	9 years ago
George Joseph	7debdd285c	res_pjsip_pubsub: Prune subs with reliable transports at startup ... In an earlier release, inbound registrations on a reliable transport were pruned on Asterisk restart since the TCP connection would have been torn down and become unusable when Asterisk stopped. This same process is now also applied to inbound subscriptions. Also fixed issues in res_pjsip_registrar where it wasn't handling the monitoring correctly when multiple registrations came in over the same transport. To accomplish this, the pjsip_transport_event feature needed to be refactored to allow multiple monitors (multiple subcriptions or registrations from the same endpoint) to exist on the same transport. Since this changed the API, any external modules that may have used the transport monitor feature (highly unlikey) will need to be changed. ASTERISK-27612 Reported by: Ross Beer Change-Id: Iee87cf4eb9b7b2b93d5739a72af52d6ca8fbbe36	9 years ago
Corey Farrell	698b28e9c7	res_pjsip_registrar_expire: Mark module deprecated. ... The functionality of this module was already moved to res_pjsip_registrar, mark it deprecated and add message to CHANGES. Change-Id: I90c7d52c7e15e85fde3389d5eaccb05b97848813	9 years ago
krells	d5bcbd460e	pbx: Reduce verbosity while loading extensions ... Each time the dial plan is reloaded, a lot of logs like these are generated: "Added extension 'XXXXX' priority 1 to YYYYYYYYYYY" This patch changes the log level for those logs. ASTERISK-27084 Change-Id: I5662902161c50890997ddc56835d4cafb456c529	9 years ago
ghjm	0b399013c6	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	9 years ago
Richard Mudgett	f35960d55b	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	9 years ago
Sean Bright	ce3d56920b	Remove as much trailing whitespace as possible. ... Change-Id: I873c1c6d00f447269bd841494459efccdd2c19c0	9 years ago