3371 Commits (1ec4f8dd003b92c262a33f6e91c3dc538681bae5)

Author	SHA1	Message	Date
Matt Jordan	1ec4f8dd00	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	10 years ago
Joshua Colp	7ed680cf92	Merge "res_pjsip: improve realtime performance #2" into 13	10 years ago
zuul	999ce35bae	Merge "codecs: Fix ABI incompatibility created by adding format_name to ast_codec" into 13	10 years ago
George Joseph	d07c8a0504	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	10 years ago
Alexander Traud	5f0a098243	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	10 years ago
zuul	43612a84c8	Merge "Fix Alembic upgrades." into 13	10 years ago
Mark Michelson	1b79e2deff	Fix Alembic upgrades. ... A non-existent constraint was being referenced in the upgrade script. This patch corrects the problem by removing the reference. 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 updates Sorcery to use "yes" and "no" ASTERISK-26128 #close Change-Id: I366dbbf91418a9cb160b3ca74b0e59b5ac284bec	10 years ago
Alexei Gradinari	b3c787d1dd	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	10 years ago
zuul	d155d82747	Merge "res_pjsip_pubsub: Address SEGV when attempting to terminate a subscription" into 13	10 years ago
George Joseph	6a568bcc66	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	10 years ago
Mark Michelson	cfebe3b94a	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
Richard Mudgett	5823f279f3	chan_rtp: Backport changes from master. ... * Deprecate chan_multicast_rtp. Change-Id: Ib5a45e58c75ee8abd0b4f9575379b5321feb853e	10 years ago
Joshua Colp	ecc186a4cc	Merge "stasis: Add setting subscription congestion levels." into 13	10 years ago
Joshua Colp	e842a99e7c	Merge "sorcery: Add setting object type congestion levels." into 13	10 years ago
zuul	1ac47c5aae	Merge "taskprocessors: Implement high/low water mark alerts." into 13	10 years ago
zuul	2204a6e6f1	Merge "res_pjsip_session: Use distributor serializer for incoming calls." into 13	10 years ago
zuul	7e2dbcd771	Merge "pjsip_distributor.c: Consistently pick a serializer for messages." into 13	10 years ago
Joshua Colp	9d8a174092	Merge "res_hep_{pjsip|rtcp}: Decline module loads if res_hep had not loaded" into 13	10 years ago
Joshua Colp	f834852137	Merge "Fix #include poll.h and sys/cdefs.h" into 13	10 years ago
Matt Jordan	eabb398d71	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	10 years ago
Timo Teräs	9c5a0b814b	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	10 years ago
Richard Mudgett	557333ea4c	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	10 years ago
Richard Mudgett	110d772467	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	10 years ago
Richard Mudgett	610eee2a36	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	10 years ago
Richard Mudgett	26e3492246	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	10 years ago
Richard Mudgett	16b08444da	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	10 years ago
George Joseph	c27c232057	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
Joshua Colp	e3c9ad6382	Merge "res_pjsip: add "via_addr", "via_port", "call_id" to contact" into 13	10 years ago
Alexei Gradinari	230686f4ec	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	10 years ago
zuul	373e68539b	Merge "func_curl: Don't trim response text on non-ASCII characters" into 13	10 years ago
Ivan Poddubny	9ddaab789e	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	10 years ago
Richard Mudgett	9453d1187a	parking.h: Update ast_parking_park_call() doxygen to reality. ... ASTERISK-26029 Change-Id: I2db14d102a48d3224010e6d1c69e856373cc1260	10 years ago
Joshua Colp	4e4a991d90	Merge "res_pjsip: Endpoint IP Access Controls" into 13	10 years ago
Alexei Gradinari	524a302974	res_pjsip: Endpoint IP Access Controls ... With the old SIP module we can use IP access controls per peer. PJSIP module missing this feature. This patch added next configuration Endpoint options: "acl" - list of IP ACL section names in acl.conf "deny" - List of IP addresses to deny access from "permit" - List of IP addresses to permit access from "contact_acl" - List of Contact ACL section names in acl.conf "contact_deny" - List of Contact header addresses to deny "contact_permit" - List of Contact header addresses to permit This patch also better logging failed request: add custom message instead of "No matching endpoint found" add SIP method to logging ASTERISK-25900 Change-Id: I456dea3909d929d413864fb347d28578415ebf02	10 years ago
Matt Jordan	89ae4466ea	res_hep: Provide an option to pick the UUID type ... At one point in time, it seemed like a good idea to use the Asterisk channel name as the HEP correlation UUID. In particular, it felt like this would be a useful identifier to tie PJSIP messages and RTCP messages together, along with whatever other data we may eventually send to Homer. This also had the benefit of keeping the correlation UUID channel technology agnostic. In practice, it isn't as useful as hoped, for two reasons: 1) The first INVITE request received doesn't have a channel. As a result, there is always an 'odd message out', leading it to be potentially uncorrelated in Homer. 2) Other systems sending capture packets (Kamailio) use the SIP Call-ID. This causes RTCP information to be uncorrelated to the SIP message traffic seen by those capture nodes. In order to support both (in case someone is trying to use res_hep_rtcp with a non-PJSIP channel), this patch adds a new option, uuid_type, with two valid values - 'call-id' and 'channel'. The uuid_type option is used by a module to determine the preferred UUID type. When available, that source of a correlation UUID is used; when not, the more readily available source is used. For res_hep_pjsip: - uuid_type = call-id: the module uses the SIP Call-ID header value - uuid_type = channel: the module uses the channel name if available, falling back to SIP Call-ID if not For res_hep_rtcp: - uuid_type = call-id: the module uses the SIP Call-ID header if the channel type is PJSIP and we have a channel, falling back to the Stasis event provided channel name if not - uuid_type = channel: the module uses the channel name ASTERISK-25352 #close Change-Id: Ide67e59a52d9c806e3cc0a797ea1a4b88a00122c	10 years ago
George Joseph	2a7130b8b0	pjproject_bundled: Check for python-dev and TEST_FRAMEWORK ... The pjsua and pjsystest apps are now built only if TEST_FRAMEWORK is set. The python bindings are now built only if TEST_FRAMEWORK is set and a python development package is installed. libresample was also disabled. ASTERISK-25993 #close Reported-by: Joshua Colp Change-Id: If4e91c503a02f113d5b71bc8b972081fa3ff6f03	10 years ago
Alexei Gradinari	7a14e669f0	res_pjsip/AMI: add contact.updated event ... With the old SIP module AMI sends PeerStatus event on every successfully REGISTER requests, ie, on start registration, update registration and stop registration. With PJSIP AMI sends ContactStatus only when status is changed. Regarding registration: on start registration - Created on stop registration - Removed but on update registration nothing This patch added contact.updated event. ASTERISK-25904 Change-Id: I8fad8aae9305481469c38d2146e1ba3a56d3108f	10 years ago
Alexei Gradinari	3cb8934de0	pjsip: Added "reg_server" to contacts. ... If the Asterisk system name is set in asterisk.conf, it will be stored into the "reg_server" field in the ps_contacts table to facilitate multi-server setups. ASTERISK-25931 Change-Id: Ia8f6bd2267809c78753b52bcf21835b9b59f4cb8	10 years ago
zuul	2ae097f144	Merge "res_pjsip_pubsub.h: Fix doxygen association." into 13	10 years ago
Richard Mudgett	b1b2019046	res_pjsip_pubsub.h: Fix doxygen association. ... Change-Id: I110d3e3572598289fcd4215d966cf0c858f98632	10 years ago
George Joseph	38bed4515d	res_pjsip: Add ability to identify by Authorization username ... A feature of chan_sip that service providers relied upon was the ability to identify by the Authorization username. This is most often used when customers have a PBX that needs to register rather than identify by IP address. From my own experiance, this is pretty common with small businesses who otherwise don't need a static IP. In this scenario, a register from the customer's PBX may succeed because From will usually contain the PBXs account id but an INVITE will contain the caller id. With nothing recognizable in From, the service provider's Asterisk can never match to an endpoint and the INVITE just stays unauthorized. The fixes: A new value "auth_username" has been added to endpoint/identify_by that will use the username and digest fields in the Authorization header instead of username and domain in the the From header to match an endpoint, or the To header to match an aor. This code as added to res_pjsip_endpoint_identifier_user rather than creating a new module. Although identify_by was always a comma-separated list, there was only 1 choice so order wasn't preserved. So to keep the order, a vector was added to the end of ast_sip_endpoint. This is only used by res_pjsip_registrar to find the aor. The res_pjsip_endpoint_identifier_* modules are called in globals/endpoint_identifier_order. Along the way, the logic in res_pjsip_registrar was corrected to match most-specific to least-specific as res_pjsip_endpoint_identifier_user does. The order is: username@domain username@domain_alias username Auth by username does present 1 problem however, the first INVITE won't have an Authorization header so the distributor, not finding a match on anything, sends a securty_alert. It still sends a 401 with a challenge so the next INVITE will have the Authorization header and presumably succeed. As a result though, that first security alert is actually a false alarm. To address this, a new feature has been added to pjsip_distributor that keeps track of unidentified requests and only sends the security alert if a configurable number of unidentified requests come from the same IP in a configurable amout of time. Those configuration options have been added to the global config object. This feature is only used when auth_username is enabled. Finally, default_realm was added to the globals object to replace the hard coded "asterisk" used when an endpoint is not yet identified. The testsuite tests all pass but new tests are forthcoming for this new feature. ASTERISK-25835 #close Reported-by: Ross Beer Change-Id: I30ba62d208e6f63439600916fcd1c08a365ed69d	10 years ago
Joshua Colp	e0e03cd2c8	Merge "res_pjsip: disable multi domain to improve realtime performace" into 13	10 years ago
Alexei Gradinari	df3639700a	res_pjsip: disable multi domain to improve realtime performace ... This patch added new global pjsip option 'disable_multi_domain'. Disabling Multi Domain can improve Realtime performance by reducing number of database requests. ASTERISK-25930 #close Change-Id: I2e7160f3aae68475d52742107949a799aa2c7dc7	10 years ago
Joshua Colp	e5c5fac8f1	Merge "res_pjsip: Add serialized scheduler (res_pjsip/pjsip_scheduler.c)" into 13	10 years ago
Joshua Colp	9d8f59aaf2	Merge changes from topic 'system_stress_patches' into 13 ... * changes: test_message.c: Wait longer in case dialplan also processes the test message. Manager: Short circuit AMI message processing. manager.c: Eliminate most RAII_VAR usage. manager_channels.c: Fix allocation failure crash.	10 years ago
Joshua Colp	4efc6b4315	Merge changes from topic 'system_stress_patches' into 13 ... * changes: Bridge system: Fix memory leaks and double frees on impart failure. bridge_softmix.c: Fix crash if channel fails to join mixing tech.	10 years ago
Richard Mudgett	ba63aa7c9e	Manager: Short circuit AMI message processing. ... Improve AMI message processing performance if there are no consumers listening for the messages. We now skip creating the AMI event message text strings. Change-Id: I7b22fc5ec4e500d00635c1a467aa8ea68a1bb2b3	10 years ago
Richard Mudgett	1e93f3d723	Bridge system: Fix memory leaks and double frees on impart failure. ... You cannot reference the passed in features struct after calling ast_bridge_impart(). Even if the call fails. Change-Id: I902b88ba0d5d39520e670fb635078a367268ea21	10 years ago
Richard Mudgett	5e388d4188	bridge_softmix.c: Fix crash if channel fails to join mixing tech. ... softmix_bridge_join() failed because of an allocation failure. To address this, the softmix bridge technology now checks if the channel failed to join softmix successfully. In addition, the bridge now begins the process of kicking the channel out of the bridge so we don't have channels partially in the bridge for very long. * Fix the test_channel_feature_hooks.c unit tests. The test channel must have a valid codec to join the simple_bridge technology. This patch makes joining a bridge more strict by not allowing partially joined channels to remain in the bridge. Change-Id: I97e2ade6a2bcd1214f24fb839fda948825b61a2b	10 years ago
Richard Mudgett	9942d50aa5	bridge: Hold off more than one imparting channel at a time. ... An earlier patch blocked the ast_bridge_impart() call until the channel either entered the target bridge or it failed. Unfortuantely, if the target bridge is stasis and the imprted channel is not a stasis channel, stasis bounces the channel out of the bridge to come back into the bridge as a proper stasis channel. When the channel is bounced out, that released the block on ast_bridge_impart() to continue. If the impart was a result of a transfer, then it became a race to see if the swap channel would get hung up before the imparted channel could come back into the stasis bridge. If the imparted channel won then everything is fine. If the swap channel gets hung up first then the transfer will fail because the swap channel is leaving the bridge. * Allow a chain of ast_bridge_impart()'s to happen before any are unblocked to prevent the race condition described above. When the channel finally joins the bridge or completely fails to join the bridge then the ast_bridge_impart() instances are unblocked. ASTERISK-25947 Reported by: Richard Mudgett ASTERISK-24649 Reported by: John Bigelow ASTERISK-24782 Reported by: John Bigelow Change-Id: I8fef369171f295f580024ab4971e95c799d0dde1	10 years ago