asterisk

Commit Graph

Author	SHA1	Message	Date
zuul	f5aa9f8034	Merge "install_prereq: Checkout of libSRTP 1.5.x." into 13	9 years ago
zuul	1cfe9514bd	Merge "chan_sip: Fix reference leaks in error paths." into 13	9 years ago
zuul	afd7aba585	Merge "res_sorcery_realtime: fix bug when successful UPDATE is treated as failed" into 13	9 years ago
zuul	daec52a8e6	Merge "res_pjsip: Added "subscribe_context" to endpoint" into 13	9 years ago
zuul	f0eb5a8b8b	Merge "BuildSystem: Avoid obsolete warning with libcurl.m4 on autoconf." into 13	9 years ago
zuul	f3b7339260	Merge "func_odbc: Fix connection deadlock." into 13	9 years ago
Richard Mudgett	5ee205d8bb	ast_expr2: Fix off-nominal memory leak. 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	9 years ago
Alexander Traud	f5e9872016	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	9 years ago
Joshua Colp	17efed6cf7	func_odbc: Fix connection deadlock. The func_odbc module was modified to ensure that the previous behavior of using a single database connection was maintained. This was done by getting a single database connection and holding on to it. With the new multiple connection support in res_odbc this will actually starve every other thread from getting access to the database as it also maintains the previous behavior of having only a single database connection. This change disables the func_odbc specific behavior if the res_odbc module is running with only a single database connection active. The connection is only kept for the duration of the request. ASTERISK-26177 #close Change-Id: I9bdbd8a300fb3233877735ad3fd07bce38115b7f	9 years ago
Corey Farrell	06ba533bc7	chan_sip: Fix reference leaks in error paths. * get_sip_pvt_from_replaces leaks sip_pvt_ptr on any error. * build_peer leaks peer on failure to allocate the endpoint. This patch fixes get_sip_pvt by using an RAII_VAR, build_peer is fixed with an unref in the appropriate place. ASTERISK-26184 #close Change-Id: I728b424648ad041409f7d90880f4c28b3ce2ca12	9 years ago
zuul	8019f32129	Merge "chan_sip/res_pjsip_t38: Handle a request to negotiate T.38 after it is enabled." into 13	9 years ago
zuul	f3fd648be6	Merge "REF_DEBUG: Prevent logging of container node objects." into 13	9 years ago
Corey Farrell	9d4e664f62	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	9 years ago
Scott Griepentrog	e26bd15e7a	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	9 years ago
Joshua Colp	77b0145a25	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	9 years ago
Alexei Gradinari	b4a9fa2c9e	res_sorcery_realtime: fix bug when successful UPDATE is treated as failed If the SQL UPDATE statement changes nothing then SQLRowCount returns 0. This value should be treated as success. But the function sorcery_realtime_update treats it as failed. This bug was found using stress tests on PJSIP. If there are 2 consecutive SIP REGISTER requests with the same contact data during 1 second then res_pjsip_registrar adds contact location on 1st request and tries to update contact location on 2nd. The update fails and res_pjsip_registrar even removes correct contact location. The test "object_update_uncreated" was removed from test_sorcery_realtime.c because it's now a valid situation. This patch also adds missing debug of extra SQL parameter. ASTERISK-26172 #close Change-Id: I05a7f3051455336c9dda29efc229decf86071303	9 years ago
Alexei Gradinari	2c16a81dd5	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
Alexander Traud	a1bd57884d	BuildSystem: Avoid obsolete warning with libcurl.m4 on autoconf. 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	9 years ago
Joshua Colp	ac6824e4c3	Merge "res_pjsip_session.c: Don't send extra BYE if SDP invalid." into 13	9 years ago
Joshua Colp	12b754ca36	Merge "res_pjsip_session.c: End call on initial invalid SDP negotiation." into 13	9 years ago
Joshua Colp	65d0c0beb1	Merge "res_pjsip.c: Register PJMEDIA error code decoder." into 13	9 years ago
Joshua Colp	643086dbc9	Merge "res_pjsip_session.c: Remove unused parameter from handle_incoming()." into 13	9 years ago
Joshua Colp	9e72f29ddd	Merge "res_pjsip: Add missing NULL checks when using pjsip_inv_end_session()." into 13	9 years ago
Joshua Colp	d8b8dca04b	Merge "features: Fix channel datastore access." into 13	9 years ago
Joshua Colp	7ed680cf92	Merge "res_pjsip: improve realtime performance #2" into 13	9 years ago
Richard Mudgett	640fbbbe28	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
Richard Mudgett	359134c8d3	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	5fabcf2ca1	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	38a4e983dc	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	1952434df5	res_pjsip_session.c: Remove unused parameter from handle_incoming(). Change-Id: Iedd182d189ec947c42edc2c66c4bda3c22060daa	9 years ago
Richard Mudgett	28928ba5c4	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
George Joseph	43a78100c0	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	77c49ac1a1	Merge "pjproject/patches/config_site: Increase the max number of ICE candidates" into 13	9 years ago
Matt Jordan	99eff80e76	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	78960975f2	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	999ce35bae	Merge "codecs: Fix ABI incompatibility created by adding format_name to ast_codec" into 13	9 years ago
zuul	272c02d4ed	Merge "siren: Add format attribute modules for Siren7 and Siren14." into 13	9 years ago
Joshua Colp	289d301c9f	Merge "BuildSystem: Avoid obsolete warning with AC_TYPE_SIGNAL on autoconf." into 13	9 years ago
Joshua Colp	c49b07aa63	Merge "BuildSystem: Fix a few issues hightlighted by gcc 6.x" into 13	9 years ago
George Joseph	d07c8a0504	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
George Joseph	f3d236ca7f	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	9d5b0934d9	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	1dfc286418	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	5f0a098243	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	73e2186195	Merge "res_fax: Fix reference leak in fax_v21_session_new." into 13	9 years ago
zuul	5a568df73d	Merge "res_rtp_asterisk: Fix a self-comparison identified by gcc 6" into 13	9 years ago
zuul	b044195e31	Merge "chan_unistim: Fix memcpy in get_to_address" into 13	9 years ago
Joshua Colp	5a73c115c8	Merge "BuildSystem: Avoid obsolete warning with AC_FUNC_SETVBUF_REVERSED on autoconf." into 13	9 years ago
zuul	43612a84c8	Merge "Fix Alembic upgrades." into 13	9 years ago
zuul	08a4699367	Merge "res_pjproject.c: Replace inlined DEBUG_ATLEAST() with macro." into 13	9 years ago

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