asterisk

Commit Graph

Author	SHA1	Message	Date
George Joseph	c7309a5254	chan_unistim: Fix memcpy in get_to_address A code block only enabled when HAVE_PKTINFO is not defined (FreeBSD) was using a pointer to a pointer as the destination of a memcpy and a '&' instead of '*' in the sizeof. ASTERISK-26138 #close Change-Id: Id4927ff256c0e470bdf7bcfc025146a2f656e708	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	3b4f5d1345	test_res_pjsip_scheduler: Add 'depends' on pjproject in MODULEINFO Since the file was missing the depends on pjproject, it wasn't picking up the pjproject related include path. If there was no system installed pjproject and pjproject-bundled was used, a compile would fail because pjsip.h wasn't found. ASTERISK-26139 #close Change-Id: I2ee64a999051452bc198c4e2c168c70769cd3757	9 years ago
Alexander Traud	5f23aacda4	BuildSystem: Avoid obsolete warning with AC_FUNC_SETVBUF_REVERSED on autoconf. Removed the obsolete macro AC_FUNC_SETVBUF_REVERSED because Asterisk does not support the platform SVR2 from the year 1987 anymore. ASTERISK-26046 Change-Id: I28161b037feb2d29ab46ed20e785928460226c22	9 years ago
Joshua Colp	aec09d9c09	Merge "res_rtp_asterisk: fix memory leak in dtls"	9 years ago
Joshua Colp	f88571822c	Merge "res_pjsip_pubsub: Address SEGV when attempting to terminate a subscription"	9 years ago
Torrey Searle	804005d251	res_rtp_asterisk: fix memory leak in dtls ensure that cert bios get freed after creating the fingerprint ASTERISK-26129 #close Change-Id: I44d23aea07dce80176ca1ff877c5ace9452ef451	9 years ago
Joshua Colp	eb08734a94	Merge "res_rtp_asterisk: Use latest DTLS version available by underlying platform."	9 years ago
Joshua Colp	eaaab8f55f	Merge "res_pjsip_session: Handle race condition at shutdown with timer."	9 years ago
Richard Mudgett	f572b26495	res_pjproject.c: Replace inlined DEBUG_ATLEAST() with macro. Change-Id: I8799fb0a347ad76e747dafd0eacf1ea1086b9a8c	9 years ago
zuul	9e0df5a1a7	Merge "PJSIP: provide transport type with received messages"	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
Alexander Traud	6eb0354f2d	res_rtp_asterisk: Use latest DTLS version available by underlying platform. Do not use DTLSv1_method() but DTLS_method() when available in OpenSSL of the underlying platform. This change enables DTLS 1.2 since OpenSSL 1.0.2, for WebRTC (DTLS-SRTP via SIP-over-WebSockets). This change enables AEAD-based cipher-suites. ASTERISK-26130 #close Change-Id: I41f24448d6d2953e8bdb97c9f4a6bc8a8f055fd0	9 years ago
Scott Griepentrog	596d0b0bc3	PJSIP: provide transport type with received messages The receipt of a SIP MESSAGE may occur over any transport including TCP and TLS. When the message is received, the original URI is added to the message in the field PJSIP_RECVADDR, but this is insufficient to ensure a reply message can reach the originating endpoint. This patch adds the PJSIP_TRANSPORT field populated with the transport type. ASTERISK-26132 #close Change-Id: I28c4b1e40d573a056c81deb213ecf53e968f725e	9 years ago
Alexander Traud	9e222efbf2	BuildSystem: Avoid obsolete warning with HELP_STRING on autoconf. Some configure scripts used both AC_HELP_STRING and its replacement AS_HELP_STRING. For consistency and to avoid obsolete warnings, those were changed to AS_HELP_STRING. ASTERISK-26046 Change-Id: I8aad4fd2bdee40aa2a31ce3339a1eb33ff4f5b0f	9 years ago
zuul	d22ce6fd3e	Merge "fix: memory leaks, resource leaks, out of bounds and bugs"	9 years ago
zuul	bbaa9c8139	Merge "app_voicemail.c: Fix IMAP compile error."	9 years ago
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

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