asterisk

Commit Graph

Author	SHA1	Message	Date
Alexander Traud	8476a9332f	install_prereq: Checkout of libSRTP 1.5.x. Since 5th November 2014, the master branch of libSRTP changed the prefix of several member names and is not compatible with the source code in Asterisk anymore. Therefore instead, this change checks out the latest version of the libSRTP 1.5.x branch. Furthermore now, libSRTP is compiled with OpenSSL as backend. This makes AES-GCM and AES-IN possible. ASTERISK-22131 #close Change-Id: I2e396cdc01da0ff610686e398ed210ca7408f7d6	10 years ago
Joshua Colp	e0f27ecabb	Merge "chan_sip/res_pjsip_t38: Handle a request to negotiate T.38 after it is enabled."	10 years ago
Joshua Colp	99cbecd270	Merge "REF_DEBUG: Prevent logging of container node objects."	10 years ago
Corey Farrell	7408c51a48	REF_DEBUG: Prevent logging of container node objects. Using AO2_CONTAINER_ALLOC_OPT_DUPS_REPLACE can result in an unref being recorded to the refs log for the node being replaced. This prevents logging of those unrefs since they would produce errors in refcounter.py. ASTERISK-26181 #close Change-Id: Ie4fded84e8a1a58b3a59ce59dfd7eb0da3ddc5d4	10 years ago
Joshua Colp	302be4809a	chan_sip/res_pjsip_t38: Handle a request to negotiate T.38 after it is enabled. Some T.38 implementations may send another re-invite after the initial one which adds additional negotiation details (such as the max bitrate). Currently this will fail when passthrough is being done in chan_sip as we do nothing if T.38 is already active. Other handlers of T.38 inside of Asterisk (such as res_fax) handle this scenario so this change adds support for it to chan_sip and res_pjsip_t38. If a request to negotiate is received while T.38 is already enabled a new re-INVITE is sent and negotiation is done again. ASTERISK-26179 #close Change-Id: I0298494d3da6df3219bbfa4be9aa04015043145c	10 years ago
Scott Griepentrog	fb96492ec4	PJSIP: provide valid tcp nodelay option for reuse When using TCP transport with chan_pjsip, the TCP_NODELAY option value was allocated on the stack, then passed as a pointer to the tcp transport configuration structure, and later re-used on subsequently created sockets when it was no longer valid. This patch changes the allocation to be a static. ASTERISK-26180 #close Reported by: Scott Griepentrog Change-Id: I3251164c7f710dbdab031282f00e30a9770626a0	10 years ago
Joshua Colp	9e10aa8496	Merge "res_pjsip_session.c: Don't send extra BYE if SDP invalid."	10 years ago
Joshua Colp	764a009fbe	Merge "res_pjsip_session.c: End call on initial invalid SDP negotiation."	10 years ago
Joshua Colp	01a8d9844b	Merge "res_pjsip.c: Register PJMEDIA error code decoder."	10 years ago
Joshua Colp	4ad22164fe	Merge "res_pjsip_session.c: Remove unused parameter from handle_incoming()."	10 years ago
Joshua Colp	082f3d123c	Merge "res_pjsip: Add missing NULL checks when using pjsip_inv_end_session()."	10 years ago
zuul	0bfa3f0141	Merge "features: Fix channel datastore access."	10 years ago
Joshua Colp	040a11cecd	Merge "res_pjsip: improve realtime performance #2"	10 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	10 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	10 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	10 years ago
Richard Mudgett	5d2fc6bab7	res_pjsip_session.c: Remove unused parameter from handle_incoming(). Change-Id: Iedd182d189ec947c42edc2c66c4bda3c22060daa	10 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	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

1 2 3 4 5 ...

28105 Commits (8476a9332fb0a1bff0b18a5ec30ec012be8a6798) All Branches Search

28105 Commits (8476a9332fb0a1bff0b18a5ec30ec012be8a6798)

All Branches