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2016-07-04BuildSystem: Avoid obsolete warning with libcurl.m4 on autoconf.Alexander Traud
Updated the macro-set autoconf/libcurl.m4 to its latest upstream version. This avoids a warning about an obsolete macro on AC_HELP_STRING, because Asterisk is using AS_HELP_STRING everywhere else already. ASTERISK-26046 Change-Id: I8299faf504ceaeee3e39930c59293809e116c631
2016-07-01Merge "res_pjsip_session.c: Don't send extra BYE if SDP invalid."Joshua Colp
2016-07-01Merge "res_pjsip_session.c: End call on initial invalid SDP negotiation."Joshua Colp
2016-07-01Merge "res_pjsip.c: Register PJMEDIA error code decoder."Joshua Colp
2016-07-01Merge "res_pjsip_session.c: Remove unused parameter from handle_incoming()."Joshua Colp
2016-07-01Merge "res_pjsip: Add missing NULL checks when using pjsip_inv_end_session()."Joshua Colp
2016-07-01Merge "features: Fix channel datastore access."zuul
2016-06-30Merge "res_pjsip: improve realtime performance #2"Joshua Colp
2016-06-30res_pjsip_session.c: Don't send extra BYE if SDP invalid.Richard Mudgett
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
2016-06-30res_pjsip_session.c: End call on initial invalid SDP negotiation.Richard Mudgett
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
2016-06-30res_pjsip.c: Register PJMEDIA error code decoder.Richard Mudgett
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
2016-06-30res_pjsip_session.c: Remove unused parameter from handle_incoming().Richard Mudgett
Change-Id: Iedd182d189ec947c42edc2c66c4bda3c22060daa
2016-06-30res_pjsip: Add missing NULL checks when using pjsip_inv_end_session().Richard Mudgett
pjsip_inv_end_session() is documented as being able to return the passed in tdata parameter set to NULL on success. Change-Id: I09d53725c49b7183c41bfa1be3ff225f3a8d3047
2016-06-30features: Fix channel datastore access.Richard Mudgett
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
2016-06-30configure: Fix HAVE_PJSIP_EVSUB_GRP_LOCK not set with external pjprojectGeorge Joseph
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
2016-06-29Merge "pjproject/patches/config_site: Increase the max number of ICE candidates"Joshua Colp
2016-06-29hep.conf.sample: Default 'enabled' to 'no'Matt Jordan
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
2016-06-29pjproject/patches/config_site: Increase the max number of ICE candidatesMatt Jordan
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
2016-06-29Merge "codecs: Fix ABI incompatibility created by adding format_name to ↵zuul
ast_codec"
2016-06-29Merge "siren: Add format attribute modules for Siren7 and Siren14."zuul
2016-06-29Merge "BuildSystem: Avoid obsolete warning with AC_TYPE_SIGNAL on autoconf."zuul
2016-06-29codecs: Fix ABI incompatibility created by adding format_name to ast_codecGeorge Joseph
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
2016-06-28Merge "BuildSystem: Fix a few issues hightlighted by gcc 6.x"Joshua Colp
2016-06-28BuildSystem: Fix a few issues hightlighted by gcc 6.xGeorge Joseph
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
2016-06-28configs/basic-pbx/modules.conf: Remove 'bad' modulesMatt Jordan
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
2016-06-23siren: Add format attribute modules for Siren7 and Siren14.Joshua Colp
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
2016-06-23BuildSystem: Avoid obsolete warning with AC_TYPE_SIGNAL on autoconf.Alexander Traud
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
2016-06-22Merge "res_fax: Fix reference leak in fax_v21_session_new."zuul
2016-06-22Merge "res_rtp_asterisk: Fix a self-comparison identified by gcc 6"Joshua Colp
2016-06-22Merge "chan_unistim: Fix memcpy in get_to_address"zuul
2016-06-22Merge "BuildSystem: Avoid obsolete warning with AC_FUNC_SETVBUF_REVERSED on ↵zuul
autoconf."
2016-06-22Merge "Fix Alembic upgrades."Joshua Colp
2016-06-22res_fax: Fix reference leak in fax_v21_session_new.Corey Farrell
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
2016-06-22Merge "res_pjproject.c: Replace inlined DEBUG_ATLEAST() with macro."zuul
2016-06-22res_pjsip: improve realtime performance #2Alexei Gradinari
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
2016-06-22res_rtp_asterisk: Fix a self-comparison identified by gcc 6George Joseph
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
2016-06-22chan_unistim: Fix memcpy in get_to_addressGeorge Joseph
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
2016-06-22Fix Alembic upgrades.Mark Michelson
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
2016-06-22test_res_pjsip_scheduler: Add 'depends' on pjproject in MODULEINFOGeorge Joseph
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
2016-06-22BuildSystem: Avoid obsolete warning with AC_FUNC_SETVBUF_REVERSED on autoconf.Alexander Traud
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
2016-06-22Merge "res_rtp_asterisk: fix memory leak in dtls"Joshua Colp
2016-06-22Merge "res_pjsip_pubsub: Address SEGV when attempting to terminate a ↵Joshua Colp
subscription"
2016-06-22res_rtp_asterisk: fix memory leak in dtlsTorrey Searle
ensure that cert bios get freed after creating the fingerprint ASTERISK-26129 #close Change-Id: I44d23aea07dce80176ca1ff877c5ace9452ef451
2016-06-21Merge "res_rtp_asterisk: Use latest DTLS version available by underlying ↵Joshua Colp
platform."
2016-06-21Merge "res_pjsip_session: Handle race condition at shutdown with timer."Joshua Colp
2016-06-21res_pjproject.c: Replace inlined DEBUG_ATLEAST() with macro.Richard Mudgett
Change-Id: I8799fb0a347ad76e747dafd0eacf1ea1086b9a8c
2016-06-21Merge "PJSIP: provide transport type with received messages"zuul
2016-06-21res_pjsip_pubsub: Address SEGV when attempting to terminate a subscriptionGeorge Joseph
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
2016-06-21res_rtp_asterisk: Use latest DTLS version available by underlying platform.Alexander Traud
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
2016-06-21PJSIP: provide transport type with received messagesScott Griepentrog
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
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