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Author	SHA1	Message	Date
zuul	dab39a6813	Merge "test_res_pjsip_scheduler: Add 'depends' on pjproject in MODULEINFO" into 13	9 years ago
George Joseph	77da168e58	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
Torrey Searle	dfcd466bf0	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
zuul	d155d82747	Merge "res_pjsip_pubsub: Address SEGV when attempting to terminate a subscription" into 13	9 years ago
Joshua Colp	6a2deb4d21	Merge "res_rtp_asterisk: Use latest DTLS version available by underlying platform." into 13	9 years ago
zuul	5280813846	Merge "res_pjsip_session: Handle race condition at shutdown with timer." into 13	9 years ago
Joshua Colp	d19169396c	Merge "PJSIP: provide transport type with received messages" into 13	9 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	9 years ago
Alexander Traud	ef97911a1c	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	69d58a1e37	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	cbfa9f771e	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	b0e71c6571	Merge "fix: memory leaks, resource leaks, out of bounds and bugs" into 13	9 years ago
zuul	9856b4afe0	Merge "app_voicemail.c: Fix IMAP compile error." into 13	9 years ago
Joshua Colp	ba0d9e7f7a	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
Richard Mudgett	c1512f4108	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	5134a8043a	fix: memory leaks, resource leaks, out of bounds and bugs ... ASTERISK-26119 #close Change-Id: Iecbf7d0f360a021147344c4e83ab242fd1e7512c	9 years ago
Joshua Colp	fe8aab6959	Merge "http: leverage 'bindaddr' for TLS in http.conf" into 13	9 years ago
zuul	db91fb74db	Merge "ARI: Ensure announcer channels are destroyed." into 13	9 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	9 years ago
Alexander Traud	76516bd79d	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
zuul	4d52b4c3e5	Merge "chan_sip: bigger buffers for headers, better failure mode" into 13	9 years ago
Vasil Kolev	89cc86fc38	chan_sip: bigger buffers for headers, better failure mode ... Currently chan_sip can give weird messages if the contacts don't fit in the From: or To: headers. This fix changes the from,to and invite variables to use ast_str, allocates and deallocates them and resizes them if needed. ASTERISK-26069 #close Change-Id: I1b68fcbddca6f6cc7d7a92fe1cb0d5430282b2b3	9 years ago
Richard Mudgett	d53a36ff33	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	03953d8034	res_pjsip.c: Add check that timer actually got scheduled. ... Change-Id: Iabaa2e5dccf0762c258101ea0eb1487cf6959ad1	9 years ago
zuul	97a8576d16	Merge "res_pjsip_session.c: Reorganize ast_sip_session_terminate()." into 13	9 years ago
Richard Mudgett	32ab98116e	res_rtp_multicast.c: Fix warning message typo. ... Change-Id: Ic9928208b9957e09866abe3d9649030942ec52b3	9 years ago
Joshua Colp	e80354caab	Merge "chan_rtp: Backport changes from master." into 13	9 years ago
Joshua Colp	9c2664d624	Merge "chan_rtp.c: Copy file from chan_multicast_rtp.c" into 13	9 years ago
Richard Mudgett	0429c53368	res_pjsip_session.c: Reorganize ast_sip_session_terminate(). ... Change-Id: I68a2128bcba4830985d2d441e70dfd1ac5bd712b	9 years ago
Richard Mudgett	5823f279f3	chan_rtp: Backport changes from master. ... * Deprecate chan_multicast_rtp. Change-Id: Ib5a45e58c75ee8abd0b4f9575379b5321feb853e	9 years ago
Richard Mudgett	dde58df318	chan_rtp.c: Copy file from chan_multicast_rtp.c ... Change-Id: I1119b53f2152ab1cbec74b5be7ea44844dbda8ef	9 years ago
Alexander Traud	ca38a3cbb4	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	c2f68787a3	Merge "cel: Ensure only one dial status per channel exists." into 13	9 years ago
Joshua Colp	ff018e28a0	Merge "res_pjsip_registrar.c: Eliminate rx REGISTER request race condition." into 13	9 years ago
Joshua Colp	ecc186a4cc	Merge "stasis: Add setting subscription congestion levels." into 13	9 years ago
Joshua Colp	e842a99e7c	Merge "sorcery: Add setting object type congestion levels." into 13	9 years ago
zuul	1ac47c5aae	Merge "taskprocessors: Implement high/low water mark alerts." into 13	9 years ago
zuul	2204a6e6f1	Merge "res_pjsip_session: Use distributor serializer for incoming calls." into 13	9 years ago
zuul	935d53fa08	Merge "res_pjsip_pubsub.c: Recreate subscriptions using distributor serializer." into 13	9 years ago
zuul	51ff79bdeb	Merge "res_pjsip_pubsub.c: Use distributor serializer for incoming subscriptions." into 13	9 years ago
zuul	7e2dbcd771	Merge "pjsip_distributor.c: Consistently pick a serializer for messages." into 13	9 years ago
zuul	54be8fd149	Merge "pjsip_distributor.c: Ignore messages until fully booted." into 13	9 years ago
Joshua Colp	caf6cccc5c	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
zuul	8438315428	Merge "chan_pjsip: Lock channel when checking for RTP changes." into 13	9 years ago
Mark Michelson	715ef071a1	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
George Joseph	a99ddc6a0d	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
zuul	bd3b6e9f19	Merge "astfd: Not maximum size of a single file but maximum file descriptors." into 13	9 years ago
zuul	c09af61c76	Merge "BuildSystem: Avoid 'ar cru' and use 'ar cr' instead." into 13	9 years ago
Joshua Colp	9d8a174092	Merge "res_hep_{pjsip|rtcp}: Decline module loads if res_hep had not loaded" into 13	9 years ago
Joshua Colp	f834852137	Merge "Fix #include poll.h and sys/cdefs.h" into 13	9 years ago