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Author	SHA1	Message	Date
Richard Mudgett	656ed73ac6	res_pjsip: Add missing NULL checks when using pjsip_inv_end_session(). ... pjsip_inv_end_session() is documented as being able to return the passed in tdata parameter set to NULL on success. Change-Id: I09d53725c49b7183c41bfa1be3ff225f3a8d3047	10 years ago
Richard Mudgett	4f7b859726	features: Fix channel datastore access. ... Found as a result of the testsuite tests/callparking test crashing. Several calls to ast_get_chan_featuremap_config() and ast_get_chan_features_xfer_config() did not lock the channel before calling so the channel's datastore list was accessed without the lock's protection. Apparently another thread deleted a datastore on the channel's list while the crashing thread was walking the list. Crash at 0xdeaddead due to MALLOC_DEBUG's memory filler value as a result. * Add missing channel locks to calls that were not already protected as the doxygen for those calls indicates. Change-Id: Id273b3d305cc616406c353cbc841b2b7655efaa1	10 years ago
George Joseph	5ad7e1c09a	configure: Fix HAVE_PJSIP_EVSUB_GRP_LOCK not set with external pjproject ... There was a typo in configure.ac preventing HAVE_PJSIP_EVSUB_GRP_LOCK from getting set when using an external pjproject. ASTERISK-26099 #close Reported-by: Ross Beer Change-Id: I709af70428e125fb5ccd44b171d25dd29141f0ae	10 years ago
Joshua Colp	5a1b3861ce	Merge "pjproject/patches/config_site: Increase the max number of ICE candidates"	10 years ago
Matt Jordan	dab2a6b689	hep.conf.sample: Default 'enabled' to 'no' ... Following the principle of least surprise, we should not be sending massive numbers of PJSIP and RTCP HEP packets out into the ether to some only-slightly-random IP address. Having 'enabled' set to 'no' in the sample configuration file should prevent this from happening for those who run 'make samples'. ASTERISK-26159 #close Change-Id: I1753a64ca83a3442a6ebdc31061f8185c062d9b1	10 years ago
Matt Jordan	9129ac8e73	pjproject/patches/config_site: Increase the max number of ICE candidates ... When negotiating ICE candidates with WebRTC capable endpoints, many networks will result in a browser offering ICE candidates that exceeds the default number of max candidates, 16. This patch bumps the max candidates to 32, with the max checks at twice the number of candidates. In practice, this has shown to be sufficient for browser/WebRTC negotiation. Change-Id: Ifd8da8b315f5ae14814d4ce20e10d2e6355020e5	10 years ago
zuul	fab67b8b4d	Merge "codecs: Fix ABI incompatibility created by adding format_name to ast_codec"	10 years ago
zuul	ba872766fa	Merge "siren: Add format attribute modules for Siren7 and Siren14."	10 years ago
zuul	6aaba96aca	Merge "BuildSystem: Avoid obsolete warning with AC_TYPE_SIGNAL on autoconf."	10 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	10 years ago
Joshua Colp	541f038694	Merge "BuildSystem: Fix a few issues hightlighted by gcc 6.x"	10 years ago
George Joseph	651290a809	BuildSystem: Fix a few issues hightlighted by gcc 6.x ... gcc 6.1.1 caught a few more issues. Made sure the unit tests still pass for the func_env and stdtime issues. ASTERISK-26157 #close Change-Id: I6664d8f34a45bc1481d2a854481c7878b0c1cf8e	10 years ago
Matt Jordan	83f2c2573b	configs/basic-pbx/modules.conf: Remove 'bad' modules ... This patch removes the following modules: - pbx_functions: It never existed. - res_pjsip_log_forwarder: It no longer exists. - res_hep_pjsip: The base HEP module wasn't loaded, and most basic PBXs aren't going to be installing HOMER - res_pjsip_phoneprov_provider: The basic res_phoneprov module isn't loaded, and we aren't configured to make use of the module Change-Id: Id91f68cae7c9c8c3d370029fe1268cb51e4ff5a5	10 years ago
Joshua Colp	75818b4084	siren: Add format attribute modules for Siren7 and Siren14. ... This change removes hardcoded SDP parsing and generation for Siren7 and Siren14 from chan_sip and moves it to format attribute modules so it can also be used by chan_pjsip. With this the fmtp lines for both are added with the bitrate information. ASTERISK-26021 Change-Id: Ibb004eda37a14c0a35ef0613f6237977fc800037	10 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	10 years ago
zuul	46cc7f114d	Merge "res_fax: Fix reference leak in fax_v21_session_new."	10 years ago
Joshua Colp	7a2daafa59	Merge "res_rtp_asterisk: Fix a self-comparison identified by gcc 6"	10 years ago
zuul	1afb0c98a9	Merge "chan_unistim: Fix memcpy in get_to_address"	10 years ago
zuul	0da2e6525b	Merge "BuildSystem: Avoid obsolete warning with AC_FUNC_SETVBUF_REVERSED on autoconf."	10 years ago
Joshua Colp	8b85b05092	Merge "Fix Alembic upgrades."	10 years ago
Corey Farrell	8c7017f76e	res_fax: Fix reference leak in fax_v21_session_new. ... fax_v21_session_new created a session details object but only released the allocation reference during error conditions. fax_session_new adds it's own reference to details if needed so the caller is always responsible for cleaning it's own reference. ASTERISK-26141 #close Change-Id: Ie7fc52a83b6596ce9ce2d5a2bd9f3e204f48fc88	10 years ago
zuul	df6f69ceb6	Merge "res_pjproject.c: Replace inlined DEBUG_ATLEAST() with macro."	10 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	10 years ago
George Joseph	d293ead077	res_rtp_asterisk: Fix a self-comparison identified by gcc 6 ... gcc 6 caught a previously unidentified self-comparison in ice_candidate_cmp. Fixed it and re-ordered the predicates for better short-circuiting. ASTERISK-26140 #close Change-Id: I3da713c568e24064430257b3502fbdafd35af7a7	10 years ago
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	10 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	10 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	10 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	10 years ago
Joshua Colp	aec09d9c09	Merge "res_rtp_asterisk: fix memory leak in dtls"	10 years ago
Joshua Colp	f88571822c	Merge "res_pjsip_pubsub: Address SEGV when attempting to terminate a subscription"	10 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	10 years ago
Joshua Colp	eb08734a94	Merge "res_rtp_asterisk: Use latest DTLS version available by underlying platform."	10 years ago
Joshua Colp	eaaab8f55f	Merge "res_pjsip_session: Handle race condition at shutdown with timer."	10 years ago
Richard Mudgett	f572b26495	res_pjproject.c: Replace inlined DEBUG_ATLEAST() with macro. ... Change-Id: I8799fb0a347ad76e747dafd0eacf1ea1086b9a8c	10 years ago
zuul	9e0df5a1a7	Merge "PJSIP: provide transport type with received messages"	10 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	10 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	10 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	10 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	10 years ago
zuul	d22ce6fd3e	Merge "fix: memory leaks, resource leaks, out of bounds and bugs"	10 years ago
zuul	bbaa9c8139	Merge "app_voicemail.c: Fix IMAP compile error."	10 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	10 years ago
zuul	ef1e0a65d4	Merge "http: leverage 'bindaddr' for TLS in http.conf"	10 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	10 years ago
Alexei Gradinari	820ed3d4b3	fix: memory leaks, resource leaks, out of bounds and bugs ... ASTERISK-26119 #close Change-Id: Iecbf7d0f360a021147344c4e83ab242fd1e7512c	10 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	10 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	10 years ago
Joshua Colp	827c47b746	Merge "Add support for OGG/Speex file format"	10 years ago
zuul	947f76a971	Merge "chan_sip: bigger buffers for headers, better failure mode"	10 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	10 years ago