asterisk

Commit Graph

Author	SHA1	Message	Date
Alexei Gradinari	1c949eea6c	res_pjsip: Added "subscribe_context" to endpoint If specified, incoming SUBSCRIBE requests will be searched for the matching extension in the indicated context. If no "subscribe_context" is specified, then the "context" setting is used. ASTERISK-25471 #close Change-Id: I3fb7a15f5bc154079bd348c08b7ad1cdd2d5e514	9 years ago
Joshua Colp	9e10aa8496	Merge "res_pjsip_session.c: Don't send extra BYE if SDP invalid."	9 years ago
Joshua Colp	764a009fbe	Merge "res_pjsip_session.c: End call on initial invalid SDP negotiation."	9 years ago
Joshua Colp	01a8d9844b	Merge "res_pjsip.c: Register PJMEDIA error code decoder."	9 years ago
Joshua Colp	4ad22164fe	Merge "res_pjsip_session.c: Remove unused parameter from handle_incoming()."	9 years ago
Joshua Colp	082f3d123c	Merge "res_pjsip: Add missing NULL checks when using pjsip_inv_end_session()."	9 years ago
zuul	0bfa3f0141	Merge "features: Fix channel datastore access."	9 years ago
Joshua Colp	040a11cecd	Merge "res_pjsip: improve realtime performance #2"	9 years ago
Richard Mudgett	9f2c007254	res_pjsip_session.c: Don't send extra BYE if SDP invalid. When an answer SDP is invalid we were disconnecting the outgoing call and sending two BYE requests. The first BYE was sent by PJPROJECT because of the invalid SDP answer. The second BYE was sent by Asterisk because it thought the canceled call was the result of the RFC5407 section 3.1.2 race condition. * Made not send the BYE on a canceled session if the SDP negotiation is incomplete because PJPROJECT has already sent a BYE for the failed negotiation. ASTERISK-25772 #close Reported by: Dmitriy Serov Change-Id: I44ad0bd0605e8eeb7035c890d6f97a1331f1a836	9 years ago
Richard Mudgett	08d3b9a89e	res_pjsip_session.c: End call on initial invalid SDP negotiation. When an incoming call defers SDP negotiation and then sends us an invalid SDP in the ACK, we need to send a BYE to disconnect the call. In this case SDP negotiation has failed and we don't have valid media streams negotiated. ASTERISK-25772 Change-Id: Ia358516b0fc1e6c4c139b78246f10b9da7a2dfb8	9 years ago
Richard Mudgett	e6e12c752c	res_pjsip.c: Register PJMEDIA error code decoder. Registering the PJMEDIA error codes allows errors found when parsing an incoming SDP to be easier to figure out. "Missing SDP rtpmap for dynamic payload type (PJMEDIA_SDP_EMISSINGRTPMAP)" is much easier to understand than "Unknown error 220030". ASTERISK-25772 Change-Id: I44b2dcea656fedd7593171be9e845880a2c70ca0	9 years ago
Richard Mudgett	5d2fc6bab7	res_pjsip_session.c: Remove unused parameter from handle_incoming(). Change-Id: Iedd182d189ec947c42edc2c66c4bda3c22060daa	9 years ago
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	9 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	9 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	9 years ago
Joshua Colp	5a1b3861ce	Merge "pjproject/patches/config_site: Increase the max number of ICE candidates"	9 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	9 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	9 years ago
zuul	fab67b8b4d	Merge "codecs: Fix ABI incompatibility created by adding format_name to ast_codec"	9 years ago
zuul	ba872766fa	Merge "siren: Add format attribute modules for Siren7 and Siren14."	9 years ago
zuul	6aaba96aca	Merge "BuildSystem: Avoid obsolete warning with AC_TYPE_SIGNAL on autoconf."	9 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	9 years ago
Joshua Colp	541f038694	Merge "BuildSystem: Fix a few issues hightlighted by gcc 6.x"	9 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	9 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	9 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	9 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	9 years ago
zuul	46cc7f114d	Merge "res_fax: Fix reference leak in fax_v21_session_new."	9 years ago
Joshua Colp	7a2daafa59	Merge "res_rtp_asterisk: Fix a self-comparison identified by gcc 6"	9 years ago
zuul	1afb0c98a9	Merge "chan_unistim: Fix memcpy in get_to_address"	9 years ago
zuul	0da2e6525b	Merge "BuildSystem: Avoid obsolete warning with AC_FUNC_SETVBUF_REVERSED on autoconf."	9 years ago
Joshua Colp	8b85b05092	Merge "Fix Alembic upgrades."	9 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	9 years ago
zuul	df6f69ceb6	Merge "res_pjproject.c: Replace inlined DEBUG_ATLEAST() with macro."	9 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	9 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	9 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	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

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