3465 Commits (be36bd7ca565531c09ca444e59b6d057131a9ab0)

Author	SHA1	Message	Date
Corey Farrell	be36bd7ca5	pbx: Create pbx_include.c for management of 'struct ast_include'. ... This changes context includes from a linked list to a vector, makes 'struct ast_include' opaque to pbx.c. Although ast_walk_context_includes is maintained the procedure is no longer efficient except for the first call (inc==NULL). This functionality is replaced by two new functions implemented by vector macros. * ast_context_includes_count (AST_VECTOR_SIZE) * ast_context_includes_get (AST_VECTOR_GET) As with ast_walk_context_includes callers of these functions are expected to have locked contexts. Only a few places in Asterisk walked the includes, they have been converted to use the new functions. const have been applied where possible to parameters for ast_include functions. Change-Id: Ib5c882e27cf96fb2aec67a39c18b4c71c9c83b60	9 years ago
Mark Michelson	273052f404	Update support for SILK format. ... This commit adds scaffolding in order to support the SILK audio format on calls. Roughly, this is what is added: * Cached silk formats. One for each possible sample rate. * ast_codec structures for each possible sample rate. * RTP payload mappings for "SILK". In addition, this change overhauls the res_format_attr_silk file in the following ways: * The "samplerate" attribute is scrapped. That's native to the format. * There are far more checks to ensure that attributes have been allocated before attempting to reference them. * We do not SDP fmtp lines for attributes set to 0. These changes make way to be able to install a codec_silk module and have it actually work. It also should allow for passthrough silk calls in Asterisk. Change-Id: Ieeb39c95a9fecc9246bcfd3c45a6c9b51c59380e	9 years ago
zuul	bea3e9b6fb	Merge "BuildSystem: Avoid obsolete warning with pthread.m4 on autoconf."	9 years ago
Joshua Colp	89f0a7d3f4	Merge "res_rtp_asterisk: Enable Forward Secrecy (PFS) for DTLS."	9 years ago
Alexander Traud	85212f2799	res_rtp_asterisk: Enable Forward Secrecy (PFS) for DTLS. ... Since July 2014, TLS based protocols (SIP over TLS, Secure WebSockets, HTTPS) support PFS thanks to ASTERISK-23905. In July 2015, the same feature was added for DTLS. The source code from main/tcptls.c should have been re-used to ease security audits. Therefore, this change rolls back the change from July 2015 and re-uses the code from July 2014. This has the additional benefits to work under CentOS 7 and enabling not just ECDHE but DHE based cipher suites as well. ASTERISK-25659 #close Reported by: StefanEng86, urbaniak, pay123 Tested by: sarumjanuch, traud patches: res_rtp_asterisk.patch submitted by sarumjanuch dtls_centos_step_1.patch submitted by traud dtls_centos_step_2.patch submitted by traud Change-Id: I537cadf4421f092a613146b230f2c0ee1be28d5c	9 years ago
Matt Jordan	f12311ee69	res/res_corosync: Raise a Stasis message on node join/leave events ... When res_corosync detects that a node leaves or joins, it currently is informed of this via Corosync callbacks. However, there are a few limitations with the information presented: (1) While we have information that Corosync is aware of - such as the Corosync nodeid - that information is really only useful inside of Corosync or res_corosync. There's no way to translate a Corosync nodeid to some other internally useful unique identifier for the Asterisk instance that just joined or left the cluster. (2) While res_corosync is notified of the instance joining or leaving the cluster, it has no mechanism to inform the Asterisk core or other modules of this event. This limits the usefulness of res_corosync as a heartbeat mechanism for other modules. This patch addresses both issues. First, it adds the notion of a cluster discovery message both within the Stasis message bus, as well as the binary event messages that res_corosync uses to transmit data back and forth within the cluster. When Asterisk joins the cluster, it sends a discovery message to the other nodes in the cluster, which correlates the Corosync nodeid along with the Asterisk EID. res_corosync now maintains a hash of Corosync nodeids to Asterisk EIDs, such that it can map changes in cluster state with the Asterisk instance that has that nodeid. Likewise, when an Asterisk instance receives a discovery message from a node in the cluster, it now sends its own discovery message back to the originating node with the local Asterisk EID. This lets Asterisk instances within the cluster build a complete picture of the other Asterisk instances within the cluster. Second, it publishes the discovery messages onto the Stasis message bus. Said messages are published whenever a node joins or leaves the cluster. Interested modules can subscribe for the ast_cluster_discovery_type() message under the ast_system_topic() and be notified when changes in cluster state occur. Change-Id: I9015f418d6ae7f47e4994e04e18948df4d49b465	9 years ago
Alexander Traud	a3f4141f6f	BuildSystem: Avoid obsolete warning with pthread.m4 on autoconf. ... Updated the macro-set autoconf/ax_pthread.m4 to its latest upstream version. ASTERISK-26046 #close Change-Id: I11abc11d17acd2b6a8a5a5be8ae8e0949dab9cc7	9 years ago
Joshua Colp	e049248161	Merge "res_pjsip: Fix statsd regression."	9 years ago
Joshua Colp	90d4ebbb40	Merge "res_pjsip: Added "subscribe_context" to endpoint"	9 years ago
Joshua Colp	8654727eb7	Merge "BuildSystem: Avoid obsolete warning with libcurl.m4 on autoconf."	9 years ago
Richard Mudgett	b85446d039	res_pjsip: Fix statsd regression. ... The ASTERISK-25904 change-id I8fad8aae9305481469c38d2146e1ba3a56d3108f patch introduced several regressions when the newly created "Updated" state goes out for each endpoint registration refresh. 1) It restarted any OPTIONS RTT ping cycle. 2) It would interfere with a currently active ping and throw off that ping's resulting RTT calculation. 3) It cleared the RTT time each time the endpoint was refreshed. 4) The cleared RTT time was sent out as a statsd update each time. 5) It created two AMI events for each update. * Revert the original patch and reimplement it. Now the current contact status state is re-sent instead of the state being momentarily toggled every time the endpoint refreshes its registration. The statsd events are not created for the re-sent refresh because they are sent after every OPTIONS ping. ASTERISK-26160 #close Reported by: Matt Jordan Change-Id: Ie072be790fbb2a8f5c1c874266e4143fa31f66d1	9 years ago
Joshua Colp	4ad333bb0e	func_odbc: Fix connection deadlock. ... The func_odbc module was modified to ensure that the previous behavior of using a single database connection was maintained. This was done by getting a single database connection and holding on to it. With the new multiple connection support in res_odbc this will actually starve every other thread from getting access to the database as it also maintains the previous behavior of having only a single database connection. This change disables the func_odbc specific behavior if the res_odbc module is running with only a single database connection active. The connection is only kept for the duration of the request. ASTERISK-26177 #close Change-Id: I9bdbd8a300fb3233877735ad3fd07bce38115b7f	9 years ago
Alexei Gradinari	1c949eea6c	res_pjsip: Added "subscribe_context" to endpoint ... If specified, incoming SUBSCRIBE requests will be searched for the matching extension in the indicated context. If no "subscribe_context" is specified, then the "context" setting is used. ASTERISK-25471 #close Change-Id: I3fb7a15f5bc154079bd348c08b7ad1cdd2d5e514	9 years ago
Alexander Traud	32cb981d04	BuildSystem: Avoid obsolete warning with libcurl.m4 on autoconf. ... Updated the macro-set autoconf/libcurl.m4 to its latest upstream version. This avoids a warning about an obsolete macro on AC_HELP_STRING, because Asterisk is using AS_HELP_STRING everywhere else already. ASTERISK-26046 Change-Id: I8299faf504ceaeee3e39930c59293809e116c631	9 years ago
Joshua Colp	040a11cecd	Merge "res_pjsip: improve realtime performance #2"	9 years ago
zuul	fab67b8b4d	Merge "codecs: Fix ABI incompatibility created by adding format_name to ast_codec"	9 years ago
zuul	6aaba96aca	Merge "BuildSystem: Avoid obsolete warning with AC_TYPE_SIGNAL on autoconf."	9 years ago
George Joseph	4045e6d8ba	codecs: Fix ABI incompatibility created by adding format_name to ast_codec ... Adding format_name even to the end of ast_codec caused issued with binary codec modules because the pointer would be garbage in asterisk when they registered. So, the ast_codec structure was reverted and an internal_ast_codec structure was created just for use in codec.c. A new internal-only API was also added (__ast_codec_register_with_format) so that codec_builtin could register codecs with the format_name in a separate parameter rather than in the ast_codec structure. ASTERISK-26144 #close Reported-by: Alexei Gradinari Change-Id: I6df1b08f6a6ae089db23adfe1ebc8636330265ba	9 years ago
Alexander Traud	6e87bf746a	BuildSystem: Avoid obsolete warning with AC_TYPE_SIGNAL on autoconf. ... Removed the obsolete macro AC_TYPE_SIGNAL because Asterisk does not use K&R C but requires ANSI C anyway. ASTERISK-26046 Change-Id: I914c014385e1862102d90fe7650621def78db02e	9 years ago
Joshua Colp	8b85b05092	Merge "Fix Alembic upgrades."	9 years ago
Alexei Gradinari	6fa3ed0679	res_pjsip: improve realtime performance #2 ... The patch removes updating all Endpoints' status on startup. Instead, only non-qualified aors with static contact and non-qualified non-expired contacts are retrieved from the realtime to update the endpoint status to ONLINE. The endpoint name was added to the contact object to simply find the endpoint that created this contact. The status of endpoints with qualified aors will be updated by 'qualify' functions. ASTERISK-26061 #close Change-Id: Id324c1776fa55d3741e0c5457ecac0304cb1a0df	9 years ago
Mark Michelson	b6bd97eea2	Fix Alembic upgrades. ... A non-existent constraint was being referenced in the upgrade script. This patch corrects the problem by removing the reference. In addition, the head of the alembic branch referred to a non-existent revision. This has been fixed by referring to the proper revision. This patch fixes another realtime problem as well. Our Alembic scripts store booleans as yes or no values. However, Sorcery tries to insert "true" or "false" instead. This patch introduces a new boolean type that translates to "yes" or "no" instead. ASTERISK-26128 #close Change-Id: I51574736a881189de695a824883a18d66a52dcef	9 years ago
George Joseph	b57cd01404	res_pjsip_pubsub: Address SEGV when attempting to terminate a subscription ... Occasionally under load we'll attempt to send a final NOTIFY on a subscription that's already been terminated and a SEGV will occur down in pjproject's evsub_destroy function. This is a result of a race condition between all the paths that can generate a notify and/or destroy the underlying pjproject evsub object: * The client can send a SUBSCRIBE with Expires: 0. * The client can send a SUBSCRIBE/refresh. * The subscription timer can expire. * An extension state can change. * An MWI event can be generated. * The pjproject transaction timer (timer_b) can expire. Normally when our pubsub_on_evsub_state is called with a terminate, we push a task to the serializer and return at which point the dialog is unlocked. This is usually not a problem because the task runs immediately and locks the dialog again. When the system is heavily loaded though, there may be a delay between the unlock and relock during which another event may occur such as the subscription timer or timer_b expiring, an extension state change, etc. These may also cause a terminate to be processed and if so, we could cause pjproject to try to destroy the evsub structure twice. There's no way for us to tell that the evsub was already destroyed and the evsub's group lock can't tolerate this and SEGVs. The remedy is twofold. * A patch has been submitted to Teluu and added to the bundled pjproject which adds add/decrement operations on evsub's group lock. * In res_pjsip_pubsub: * configure.ac and pjproject-bundled's configure.m4 were updated to check for the new evsub group lock APIs. * We now add a reference to the evsub group lock when we create the subscription and remove the reference when we clean up the subscription. This prevents evsub from being destroyed before we're done with it. * A state has been added to the subscription tree structure so termination progress can be tracked through the asyncronous tasks. * The pubsub_on_evsub_state callback has been split so it's not doing double duty. It now only handles the final cleanup of the subscription tree. pubsub_on_rx_refresh now handles both client refreshes and client terminates. It was always being called for both anyway. * The serialized_on_server_timeout task was removed since serialized_pubsub_on_rx_refresh was almost identical. * Missing state checks and ao2_cleanups were added. * Some debug levels were adjusted to make seeing only off-nominal things at level 1 and nominal or progress things at level 2+. ASTERISK-26099 #close Reported-by: Ross Beer. Change-Id: I779d11802cf672a51392e62a74a1216596075ba1	9 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	9 years ago
Richard Mudgett	dcfef53ee2	stasis: Add setting subscription congestion levels. ... Stasis subscriptions and message routers create taskprocessors to process the event messages. API calls are needed to be able to set the congestion levels of these taskprocessors for selected subscriptions and message routers. * Updated CDR, CEL, and manager's stasis subscription congestion levels based upon stress testing. Increased the congestion levels to reduce the potential for bursty call setup/teardown activity from triggering the taskprocessor overload alert. CDRs in particular need an extra high congestion level because they can take awhile to process the stasis messages. ASTERISK-26088 Reported by: Richard Mudgett Change-Id: Id0a716394b4eee746dd158acc63d703902450244	9 years ago
Richard Mudgett	4879cd875c	sorcery: Add setting object type congestion levels. ... Sorcery creates taskprocessors for object types to process object observer callbacks. An API call is needed to be able to set the congestion levels of these taskprocessors for selected object types. * Updated PJSIP's contact and contact_status sorcery object type observer default congestion levels based upon stress testing. Increased the congestion levels to reduce the potential for bursty register/unregister and subscribe/unsubscribe activity from triggering the taskprocessor overload alert. ASTERISK-26088 Reported by: Richard Mudgett Change-Id: I4542e83b556f0714009bfeff89505c801f1218c6	9 years ago
Richard Mudgett	2cd67d5b07	taskprocessors: Implement high/low water mark alerts. ... When taskprocessors get backed up, there is a good chance that we are being overloaded and need to defer adding new work to the system. * Implemented a high/low water alert mechanism for modules to check if the system is being overloaded and take appropriate action. When a taskprocessor is created it has default congestion levels set. A taskprocessor can later have those congestion levels altered for specific needs if stress testing shows that the taskprocessor is a symptom of overloading or needs to handle bursty activity without triggering an overload alert. * Add CLI "core show taskprocessor" low/high water columns. * Fixed __allocate_taskprocessor() to not use RAII_VAR(). RAII_VAR() was never a good thing to use when creating a taskprocessor because of the nature of how its references needed to be cleaned up on a partial creation. * Made res_pjsip's distributor check if the taskprocessor overload alert is active before placing a message representing brand new work onto a distributor serializer. ASTERISK-26088 Reported by: Richard Mudgett Change-Id: I182f1be603529cd665958661c4c05ff9901825fa	9 years ago
Richard Mudgett	c966a035e0	res_pjsip_session: Use distributor serializer for incoming calls. ... We must continue using the serializer that the original INVITE came in on for the dialog. There may be retransmissions already enqueued in the original serializer that can result in reentrancy and message sequencing problems. Outgoing call legs create the pjsip/outsess/<endpoint> serializers for their dialogs. ASTERISK-26088 Reported by: Richard Mudgett Change-Id: I24d7948749c582b8045d5389ba3f6588508adbbc	9 years ago
Richard Mudgett	2ff26e9746	pjsip_distributor.c: Consistently pick a serializer for messages. ... Incoming messages that are not part of a dialog or a recognized response to one of our requests need to be sent to a consistent serializer. Under load we may be queueing retransmissions before we can process the original message. We don't need to throw these messages onto random serializers and cause reentrancy and message sequencing problems. * Created a pool of pjsip/distributor serializers that get picked by hashing the call-id and remote tag strings of the received messages. * Made ast_sip_destroy_distributor() destroy items in the reverse order of creation. ASTERISK-26088 Reported by: Richard Mudgett Change-Id: I2ce769389fc060d9f379977f559026fbcb632407	9 years ago
Joshua Colp	6ef3094239	Merge "res_hep_{pjsip|rtcp}: Decline module loads if res_hep had not loaded"	9 years ago
Joshua Colp	7bcccd4db3	Merge "Fix #include poll.h and sys/cdefs.h"	9 years ago
Matt Jordan	7f5ca67e5f	res_hep_{pjsip|rtcp}: Decline module loads if res_hep had not loaded ... A crash can occur in res_hep_pjsip or res_hep_rtcp if res_hep has not loaded and does not have a configuration file. Previously when this occurred, checks were put in to see if the configuration was loaded successfully. While this is a good idea - and has been added to the offending function in res_hep - the reality is res_hep_pjsip and res_hep_rtcp have no business running if res_hep isn't also running. As such, this patch also adds a function to res_hep that returns whether or not it successfully loaded. Oddly enough, ast_module_check returns "everything is peachy" even if a module declined its load - so it cannot be solely relied on. res_hep_pjsip and res_hep_rtcp now also check this function to see if they should continue to load; if it fails, they decline their load as well. ASTERISK-26096 #close Change-Id: I007e535fcc2e51c2ca48534f48c5fc2ac38935ea	9 years ago
Joshua Colp	643dd462ee	Merge "ari/resource_channels: Add 'formats' to channel create/originate"	9 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	9 years ago
Timo Teräs	9c1d95e873	Fix #include poll.h and sys/cdefs.h ... POSIX defines poll.h, sys/poll.h should not be used at is c-library internal header which may or may not exist. Notable in musl it generates warning of being incorrect. And add explict include of sys/cdefs.h where needed. Change-Id: I142930df53fe7585a06b854b6faddc5301e024be	9 years ago
zuul	84a93b0d67	Merge "ARI: Re-implement the ARI dial command, allowing for early bridging."	9 years ago
Joshua Colp	11ea121cc8	Merge "res_pjsip: add "via_addr", "via_port", "call_id" to contact"	9 years ago
Mark Michelson	bb0f4a6310	multicast RTP: Add dialing options ... This adds a new parameter to the end of a multicast RTP dialing string. This parameter defines the following options: * i: Set the interface from which multicast RTP is sent * l: Set whether multicast packets are looped back to the sender * t: Set the TTL for multicast packets * c: Set the codec to use for RTP ASTERISK-26068 #close Reported by Mark Michelson Change-Id: I033b706b533f0aa635c342eb738e0bcefa07e219	9 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	9 years ago
Alexei Gradinari	31f17abe44	res_pjsip: add "via_addr", "via_port", "call_id" to contact ... As res_pjsip_nat rewrites contact's address, only the last Via header can contain the source address of registered endpoint. Also Call-Id header may contain the source address of registered endpoint. Added "via_addr", "via_port", "call_id" to contact. Added new fields ViaAddress, CallID to AMI event ContactStatus. ASTERISK-26011 Change-Id: I36bcc0bf422b3e0623680152d80486aeafe4c576	9 years ago
Joshua Colp	b0e4ea96de	Merge "Bridging: introduce "invisible" bridges."	9 years ago
Joshua Colp	cab97fd905	Merge "ARI: Add the ability to download the media associated with a stored recording"	9 years ago
Mark Michelson	f6c33771f6	Bridging: introduce "invisible" bridges. ... Invisible bridges function the same as normal bridges, but they have the following restrictions: * They never show up in CLI, AMI, or ARI queries. * They do not have Stasis messages published about them. Invisible bridges' main use is for when use of the bridging system is desired, but the bridge should not be known to users of the Asterisk system. ASTERISK-25925 Change-Id: I804a209d3181d7c54e3d61a60eb462e7ce0e3670	9 years ago
Joshua Colp	56d5af4584	Merge "func_curl: Don't trim response text on non-ASCII characters"	9 years ago
Ivan Poddubny	31897d2d99	func_curl: Don't trim response text on non-ASCII characters ... The characters 0x80-0xFF were trimmed as well as 0x00-0x20 because of a signed comparison. ASTERISK-25669 #close Reported by: Jesper patches: strings.curl.trim.patch submitted by Jesper (License 5518) Change-Id: Ia51e169f24e3252a7ebbaab3728630138ec6f60a	9 years ago
Richard Mudgett	ade5275a3e	parking.h: Update ast_parking_park_call() doxygen to reality. ... ASTERISK-26029 Change-Id: I2db14d102a48d3224010e6d1c69e856373cc1260	9 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	9 years ago
Joshua Colp	e205eb55a4	Merge "res_pjsip: Endpoint IP Access Controls"	9 years ago
Joshua Colp	5acb25722c	Merge "logger: Support JSON logging with Verbose messages"	9 years ago
Joshua Colp	b57032c364	Merge "res_hep: Provide an option to pick the UUID type"	9 years ago