28147 Commits (be36bd7ca565531c09ca444e59b6d057131a9ab0)

 

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Author	SHA1	Message	Date
Joshua Colp	e94aae00a7	res_pjsip_session: Handle race condition at shutdown with timer. ... When shutting down res_pjsip_session will get unloaded before res_pjsip. The act of unloading unregisters all the PJSIP services and sets their module IDs to -1. In some cases it is possible for a timer to occur after this happens which calls into res_pjsip_session. The res_pjsip_session module can then try to get the session from the INVITE session using the module ID. Since the module ID is now -1 this fails. This change stores a copy of the module ID and uses it for the timer callback scenario. If the module ID is -1 the callback immediately returns but if the module ID is valid then it continues as normal. This works as the original ID of the module is guaranteed to still be valid when used with the INVITE session. ASTERISK-26127 #close Change-Id: I88df72525c4e9ef9f19c13aedddd3ac4a335c573	9 years ago
zuul	ef1e0a65d4	Merge "http: leverage 'bindaddr' for TLS in http.conf"	9 years ago
Richard Mudgett	0a30008224	app_voicemail.c: Fix IMAP compile error. ... Fix compile error introduced by the patch for ASTERISK-26045 Change-Id: I5b02876266f2824f4cec2b54d6ff4db5de5778d3	9 years ago
Alexei Gradinari	820ed3d4b3	fix: memory leaks, resource leaks, out of bounds and bugs ... ASTERISK-26119 #close Change-Id: Iecbf7d0f360a021147344c4e83ab242fd1e7512c	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
Alexander Traud	f72ffc1ff9	http: leverage 'bindaddr' for TLS in http.conf ... The internal HTTP/WebSocket server supports both TCP and TLS, which can be activated separately via the file http.conf. The source code intends to re-use the TCP parameter 'bindaddr' for TLS, even if 'tlsbindaddr' is not specified explicitly. This did not work because of a typo. This change resolves this typo. ASTERISK-26126 #close Change-Id: I5efb0409ae12044dfb3495b6b97b6d40a8c9c51f	9 years ago
Joshua Colp	827c47b746	Merge "Add support for OGG/Speex file format"	9 years ago
zuul	947f76a971	Merge "chan_sip: bigger buffers for headers, better failure mode"	9 years ago
Richard Mudgett	3c80f84cd0	res_pjsip_transport_management.c: Misc cleanups to survive shutdown. ... * In unload_module(), reordered destroying things to minimize the window that the global transports container could be used by other threads on shutdown. When shutting down you need to stop things in the opposite order of creation. * Put the global transports container into an AO2_GLOBAL_OBJ_STATIC to eliminate the crash potential by other threads using the container on shutdown. * Made struct monitored_transport.sip_received not use ast_atomic_fetchadd_int() since it is used as a boolean value that is only set TRUE. It was previously incremented for every received SIP message and could theoretically overflow. * In monitored_transport_state_callback(), allocated the monitored transport object without a lock since the lock was unused. * In keepalive_global_loaded(), removed releasing the transports container if the keepalive_thread could not be started. I set it up to be tried again if the user reloads the configuration. Change-Id: I8d12d16ef564290fa6d25a32334bb5ce8fdf87ff	9 years ago
Richard Mudgett	7c59f2126f	res_pjsip.c: Add check that timer actually got scheduled. ... Change-Id: Iabaa2e5dccf0762c258101ea0eb1487cf6959ad1	9 years ago
zuul	181766748f	Merge "res_pjsip_session.c: Reorganize ast_sip_session_terminate()."	9 years ago
Richard Mudgett	51cc5c31c4	res_rtp_multicast.c: Fix warning message typo. ... Change-Id: Ic9928208b9957e09866abe3d9649030942ec52b3	9 years ago
Richard Mudgett	3d0632a9c2	res_pjsip_session.c: Reorganize ast_sip_session_terminate(). ... Change-Id: I68a2128bcba4830985d2d441e70dfd1ac5bd712b	9 years ago
zuul	d9b5aea9c3	Merge "core: Not the configured but granted number of possible file descriptors."	9 years ago
Alexander Traud	ac683f13c9	core: Not the configured but granted number of possible file descriptors. ... With CLI "core show settings", simply the parameter maxfiles of the file asterisk.conf was shown. If that parameter was not set, nothing was displayed although the environment might have set a default number itself. Or if maxfiles were not granted (completely), still maxfiles was shown. Now, the maximum number of possible file descriptors in the environment is shown. ASTERISK-26097 Change-Id: I2df5c58863b5007b34b77adbe28b885dfcdf7e0b	9 years ago
Joshua Colp	d1006faba9	Merge "astfd: With RLIMIT_NOFILE only the current value is sensible."	9 years ago
Joshua Colp	4eb8cf2684	translate: Enables native Packet-Loss Concealment (PLC) for supporting codecs. ... This reverts commit 5bfef2a8b4 as it caused fax test failures. ASTERISK-25629 Change-Id: I79de974dc4f63a1cafe0d2509169fd9a6b3cbaf4	9 years ago
Alexander Traud	0bf1a53db3	astfd: With RLIMIT_NOFILE only the current value is sensible. ... With menuselect "DEBUG_FD_LEAKS" and CLI "core show fd", both the maximum max and current max of possible file descriptors were shown. Both show the same value always. Not to confuse users, just the current maximum is shown now. ASTERISK-26097 Change-Id: I49cf7952d73aec9e3f6a88942842c39be18380fa	9 years ago
zuul	dcb6875428	Merge "cel: Ensure only one dial status per channel exists."	9 years ago
zuul	39e6d80937	Merge "ARI: Ensure proper channel state on operations."	9 years ago
zuul	850a09b099	Merge "test_http_media_cache: Fix failing test."	9 years ago
zuul	88dfcd21b2	Merge "chan_sip: Support auth username for callbackextension feature"	9 years ago
Joshua Colp	9e4efe9e16	Merge "res_pjsip_registrar.c: Eliminate rx REGISTER request race condition."	9 years ago
Joshua Colp	914a1502fa	Merge "stasis: Add setting subscription congestion levels."	9 years ago
Joshua Colp	67a45e0a38	Merge "sorcery: Add setting object type congestion levels."	9 years ago
Joshua Colp	48c2c3b8da	Merge "taskprocessors: Implement high/low water mark alerts."	9 years ago
Joshua Colp	2b840fbc3f	Merge "res_pjsip_session: Use distributor serializer for incoming calls."	9 years ago
Joshua Colp	66e1e0969b	Merge "res_pjsip_pubsub.c: Recreate subscriptions using distributor serializer."	9 years ago
Joshua Colp	2b2ca82e71	Merge "res_pjsip_pubsub.c: Use distributor serializer for incoming subscriptions."	9 years ago
Joshua Colp	9acb5e3084	Merge "pjsip_distributor.c: Consistently pick a serializer for messages."	9 years ago
zuul	f5ffcb1b72	Merge "pjsip_distributor.c: Ignore messages until fully booted."	9 years ago
Joshua Colp	d338343dac	cel: Ensure only one dial status per channel exists. ... CEL wrongly assumed that a channel would only have a single dial event on it. This is incorrect. Particularly in a queue each call attempt to a member will result in a dial event, adding a new dial status in CEL without removing the old one. This would cause the container to grow with only one dial status being removed when the channel went away. The other dial status entries would remain leaking memory. This change fixes the memory leak by ensuring that only one dial status will only ever exist for each channel. The behavior during the scenario where multiple events are received has also been improved. For failure cases the first failure will be the dial status. If an answer dial status is received, though, it will take priority and the dial status for the channel will be answer. Memory usage has also been decreased by storing the minimal amount of information and the code has been cleaned up slightly. ASTERISK-25262 #close Change-Id: I5944eb923db17b6a0faa7317ff6abc9307c009fe	9 years ago
Mark Michelson	1fd3a7849e	ARI: Ensure proper channel state on operations. ... ARI was recently outfitted with operations to create and dial channels. This leads to the ability to try funny stuff. You could create a channel and then immediately try to play back media on it. You could create a channel, dial it, and while it is ringing attempt to make it continue in the dialplan. This commit attempts to fix this by adding a channel state check to operations that should not be able to operate on outbound channels that have not yet answered. If a channel is in an invalid state, we will send a 412 response. ASTERISK-26047 #close Reported by Mark Michelson Change-Id: I2ca51bf9ef2b44a1dc5a73f2d2de35c62c37dfd8	9 years ago
Mark Michelson	10019dc70c	test_http_media_cache: Fix failing test. ... The retrieve_cache_control_directives test has been failing occasionally in Jenkins. The apparent failure occurs when attempting to validate the expiration of the retrieved file. After reproducing, the problem was pretty clear. At the beginning of the test, the current time is retrieved. The seconds value of this timestamp is X. When the file is retrieved, res_http_media_cache calculates the expiration and in doing so retrieves the current time. In most cases, since the test executes quickly, it will also retrieve a timestamp with X seconds. However, if the test starts very near to when the timestamp seconds are set to increment, res_http_media_cache may retrieve a timestamp with X+1 seconds instead. The test attempted to account for this by allowing a tolerance of 1 second when validating the expiration. However, the problem was that the comparisons being used in the validation used > and < operations. This meant that values that fell within the tolerance (because they equaled the upper bound of the tolerance) would fail. The solution is to use >= and <= operators in the expiration validation. However, I estimated that while the one second tolerance should be fine on most machines, it would still be possible on a very slow machine to end up falling outside the one second tolerance. So I have also relaxed the tolerance of expiration validation to be three seconds instead. The final change here is to add a debug message when validating expiration so that we can see what values are being compared. ASTERISK-25959 #close Reported by Joshua Colp Change-Id: Ic1a0e10722c1c5d276d5a4d6a67136d6ec26c247	9 years ago
Timo Teräs	56bdf048d2	Add support for OGG/Speex file format ... ASTERISK-18995 #close Change-Id: I98518bd28fc8f95668b3fe27d2cab45045ff3f7a	9 years ago
zuul	0388c40b8c	Merge "chan_pjsip: Lock channel when checking for RTP changes."	9 years ago
George Joseph	f0855358a6	cdr.c: Remove assert in base_process_dial_end ... Scenario: Caller blonde transfer Bob calls Charlie who answers. Bob puts Charlie on hold and calls Alice. Before Alice answers, Bob transfers Charlie to Alice. Charlie's channel triggers an assert because he gets an "ANSWERED" event even though he never dialed anything. With recent changes to dial events, this is now a valid scenario so the assert needed to be removed. ASTERISK-26103 #close Change-Id: I2679b517b696e7952ab7fb29403df9140e7d1de2	9 years ago
Mark Michelson	cdb7edbe7b	chan_pjsip: Lock channel when checking for RTP changes. ... bridge_native_rtp can call into an RTP-capable channel driver in order for the driver to update information about who the channel is communicating with. For SIP channel drivers, this means deactivating RTCP and sending a reinvite so that the endpoints can communicate directly. bridge_native_rtp does the right thing and has the channel locked when calling into the channel driver. chan_pjsip can't alter session properties in this thread, though. chan_pjsip queues a task on the session serializer in order to update properties there. The problem is that this queued task was not locking the channel. This meant that the queued task could attempt to deactivate RTCP at the same time that the channel thread was attempting to process an incoming RTCP packet. This could lead to a crash. This patch fixes the issue by locking the channel in the queued task when altering RTP properties. ASTERISK-26092 #close Reported by Niklas Larsson Change-Id: I3464e226a3c41f6b915f97891e07fa1599e2a159	9 years ago
Richard Mudgett	04ec9c745e	res_pjsip_registrar.c: Eliminate rx REGISTER request race condition. ... This patch fixes a race condition processing received REGISTER requests and their retransmissions caused by REGISTER requests being processed by two threads. The "sip_transaction Unable to register REGISTER transaction (key exists)" message is a notable symptom of this issue. This issue was more likely to happen before the pjsip/distributor serializers were created. Instead of steps one and two below placing the REGISTER messages into the same pjsip/distributor they were placed in random pjsip/default serializers. 1) REGISTER requests come in and get placed on the pjsip/distributor serializer. 2) Before the first request is processed a retransmission comes in and is placed on the same pjsip/distributor serializer. 3) The first request goes up the pjsip stack and is then shunted off to the pjsip/aor/<aor> serializer. 4) Before the first request is completed processing in the pjsip/aor/<aor> serializer, the second request goes up the pjsip stack and is also shunted off to the pjsip/aor/<aor> serializer. 5) The first request completes processing and sends out its response. 6) The second request completes processing and tries to send out its response but pjlib complains that the REGISTER transaction key already exists. 7) Sadness ensues. * The race is eliminated by removing the pjsip/aor/<aor> serializer and continuing the processing in the pjsip/distributor serializer. Now any retransmissions queued in the pjsip/distributor serializer will be processed after the first message is completely processed. ASTERISK-26088 #close Reported by: Richard Mudgett Change-Id: I842d714346088bf717ea27437f1dd85bff0bab5a	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	5b7b16a87f	res_pjsip_pubsub.c: Recreate subscriptions using distributor serializer. ... * Resolves potential reentrancy problems if system restarted in the middle of subscription message transactions. * Fixes memory leak recreating persistent subscriptions when the subscription resource tree could not be created. ASTERISK-26088 Reported by: Richard Mudgett Change-Id: I71e34d7ae8ed35a694f1030e820e2548c48697be	9 years ago
Richard Mudgett	c2ae49249c	res_pjsip_pubsub.c: Use distributor serializer for incoming subscriptions. ... We must continue using the serializer that the original SUBSCRIBE 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. The "sip_transaction Unable to register SUBSCRIBE transaction (key exists)" message is a notable symptom of this issue. Outgoing subscriptions still create the pjsip/pubsub/<endpoint> serializers for their dialogs. ASTERISK-26088 Reported by: Richard Mudgett Change-Id: I18b00bb74a56747b2c8c29543a82440b110bf0b0	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
Richard Mudgett	df2791da8f	pjsip_distributor.c: Ignore messages until fully booted. ... We should not be processing any incoming messages until we are fully booted. We may not have dialplan or other needed configuration loaded yet. ASTERISK-26089 #close Reported by: Scott Griepentrog ASTERISK-26088 Reported by: Richard Mudgett Change-Id: I584aefb4f34b885a8927e1f13a2c64babd606264	9 years ago
George Joseph	d21a77b325	build: Fix ast_sockaddr initialization to be more portable ... A change to glibc 2.22 changed the order of the sockadddr_storage members which caused the places where we do an initialization of ast_sockaddr with '{ { 0, 0, } }' to fail compilation. Those initializers (which we shouldn't have been using anyway) have been replaced with memsets. Change-Id: Idd1b3b320903d8771bfe221f0b015685de628fa4	9 years ago
Joshua Colp	fbece11a0c	Merge "translate: Enables native Packet-Loss Concealment (PLC) for supporting codecs."	9 years ago
Joshua Colp	2525563438	Merge "chan_sip: No rtpmap for static RTP payload IDs in SDP."	9 years ago