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2016-07-11ast_expr2: Fix off-nominal memory leak.Richard Mudgett
Thanks to ibercom for pointing out a memory leak that was missed in the earlier patch for the issue. ASTERISK-26119 Reported by: Alexei Gradinari Change-Id: I9a151f5c4725d97fb82a9e938bc73dc659532b71
2016-07-08Merge "chan_sip/res_pjsip_t38: Handle a request to negotiate T.38 after it ↵zuul
is enabled." into 13
2016-07-08Merge "REF_DEBUG: Prevent logging of container node objects." into 13zuul
2016-07-07REF_DEBUG: Prevent logging of container node objects.Corey Farrell
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
2016-07-07PJSIP: provide valid tcp nodelay option for reuseScott Griepentrog
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a pointer to the tcp transport configuration structure, and later re-used on subsequently created sockets when it was no longer valid.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
2016-07-07chan_sip/res_pjsip_t38: Handle a request to negotiate T.38 after it is enabled.Joshua Colp
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
2016-07-01Merge "res_pjsip_session.c: Don't send extra BYE if SDP invalid." into 13Joshua Colp
2016-07-01Merge "res_pjsip_session.c: End call on initial invalid SDP negotiation." ↵Joshua Colp
into 13
2016-07-01Merge "res_pjsip.c: Register PJMEDIA error code decoder." into 13Joshua Colp
2016-07-01Merge "res_pjsip_session.c: Remove unused parameter from handle_incoming()." ↵Joshua Colp
into 13
2016-07-01Merge "res_pjsip: Add missing NULL checks when using ↵Joshua Colp
pjsip_inv_end_session()." into 13
2016-07-01Merge "features: Fix channel datastore access." into 13Joshua Colp
2016-06-30Merge "res_pjsip: improve realtime performance #2" into 13Joshua Colp
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
2016-06-30res_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
2016-06-30configure: 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 ↵Joshua Colp
candidates" into 13
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" into 13
2016-06-29Merge "siren: Add format attribute modules for Siren7 and Siren14." into 13zuul
2016-06-29Merge "BuildSystem: Avoid obsolete warning with AC_TYPE_SIGNAL on autoconf." ↵Joshua Colp
into 13
2016-06-29Merge "BuildSystem: Fix a few issues hightlighted by gcc 6.x" into 13Joshua Colp
2016-06-28codecs: 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
2016-06-28BuildSystem: 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." into 13zuul
2016-06-22Merge "res_rtp_asterisk: Fix a self-comparison identified by gcc 6" into 13zuul
2016-06-22Merge "chan_unistim: Fix memcpy in get_to_address" into 13zuul
2016-06-22Merge "BuildSystem: Avoid obsolete warning with AC_FUNC_SETVBUF_REVERSED on ↵Joshua Colp
autoconf." into 13
2016-06-22Merge "Fix Alembic upgrades." into 13zuul
2016-06-22Merge "res_pjproject.c: Replace inlined DEBUG_ATLEAST() with macro." into 13zuul
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 "test_res_pjsip_scheduler: Add 'depends' on pjproject in MODULEINFO" ↵zuul
into 13
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. 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 updates Sorcery to use "yes" and "no" ASTERISK-26128 #close Change-Id: I366dbbf91418a9cb160b3ca74b0e59b5ac284bec
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
2016-06-22test_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
2016-06-22res_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 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_pjsip_pubsub: Address SEGV when attempting to terminate a ↵zuul
subscription" into 13
2016-06-21Merge "res_rtp_asterisk: Use latest DTLS version available by underlying ↵Joshua Colp
platform." into 13
2016-06-21Merge "res_pjsip_session: Handle race condition at shutdown with timer." into 13zuul
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