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2016-07-21Merge "Add conditional support for noreturn functions." into 13zuul
2016-07-20Merge "Makefile: Retain XML Declaration and DTD in docs." into 13zuul
2016-07-20Merge "Unit tests: Use AST_TEST_DEFINE in conditional code only." into 13zuul
2016-07-20Merge "res_rtp_asterisk: Count a roll-over of the sequence number even on ↵zuul
lost packets." into 13
2016-07-20Merge "res_pjsip_mwi: remove unneeded check on endpoint's contacts." into 13zuul
2016-07-19Add conditional support for noreturn functions.Corey Farrell
This adds support for tagging functions with the noreturn attribute. If DO_CRASH is enabled then ast_do_crash never returns. If AST_DEVMODE and DO_CRASH are enabled then failed assertions never return. This can resolve a large number of false positives with static analyzers. ASTERISK-26220 #close Change-Id: Icfb61e5fe54574eced4c3e88b317244f467ec753
2016-07-19Makefile: Retain XML Declaration and DTD in docs.Alexander Traud
Since Asterisk 12, the documentation got an XML Stylesheet. Because of a typo, the XML Declaration and DTD were overwritten by this. ASTERISK-26212 #close Change-Id: If5ee4625068042e98ab3fcb22a25e2f15d0c68bd
2016-07-18Unit tests: Use AST_TEST_DEFINE in conditional code only.Corey Farrell
If AST_TEST_DEFINE is not conditional to TEST_FRAMEWORK it produces dead code. This places all existing unit tests into a conditional block if they weren't already. ASTERISK-26211 #close Change-Id: I8ef83ee11cbc991b07b7a37ecb41433e8c734686
2016-07-18res_rtp_asterisk: Count a roll-over of the sequence number even on lost packets.Alexander Traud
With this change, the initial RTP sequence number is randomly chosen not between 0 and 65535 (0xffff) but 0 and 32767 (0x7fff). This assures, the roll-over counter (ROC) synchronization is not lost for sRTP, when the very first RTP packets get lost; see http://srtp.sourceforge.net/faq.html#Q6 ASTERISK-26207 #close Change-Id: I9a527e3aa3ce8f3becc5131d7ba32b57b5845464
2016-07-18Makefile: Suppress echoing of target 'config' again.Alexander Traud
ASTERISK-26038 #close Change-Id: I5746cf639f3fdc6332e8a97cf01f979e30bf403f
2016-07-15Merge "app_queue: Only remove queue member from pending when state changes." ↵zuul
into 13
2016-07-14features.c: Remove unneeded adsi.h include.Corey Farrell
adsi.h is no longer used by features.c since parking was moved to a module. Change-Id: I2248b8a455225a17cb6ddaafd6c20c511a1eaf59
2016-07-14Merge "Update support for SILK format." into 13zuul
2016-07-14res_pjsip_mwi: remove unneeded check on endpoint's contacts.Alexei Gradinari
The function create_mwi_subscriptions_for_endpoint checks if there is active contacts by retrieving aors and contacts. This function is used to create all unsolicited mwi subscriptions on startup and is used when contact added. In both cases it's not necessary to check if there are contacts. The contacts are needed when asterisk sends mwi. ASTERISK-26200 #close Change-Id: I98e43bdc97f3c0829951cd9bf5f3c6348c6ac1fa
2016-07-14Update support for SILK format.Mark Michelson
This commit adds scaffolding in order to support the SILK audio format on calls. Roughly, this is what is added: * Cached silk formats. One for each possible sample rate. * ast_codec structures for each possible sample rate. * RTP payload mappings for "SILK". In addition, this change overhauls the res_format_attr_silk file in the following ways: * The "samplerate" attribute is scrapped. That's native to the format. * There are far more checks to ensure that attributes have been allocated before attempting to reference them. * We do not SDP fmtp lines for attributes set to 0. These changes make way to be able to install a codec_silk module and have it actually work. It also should allow for passthrough silk calls in Asterisk. Change-Id: Ieeb39c95a9fecc9246bcfd3c45a6c9b51c59380e
2016-07-14Merge "translate: explicit format destination not properly set" into 13zuul
2016-07-14Merge "threadpool: Fix leak in ast_threadpool_serializer_group error path." ↵Joshua Colp
into 13
2016-07-14Merge "pbx: Fix leak of timezone for time based includes." into 13zuul
2016-07-14Merge "BuildSystem: Avoid obsolete warning with pthread.m4 on autoconf." into 13zuul
2016-07-14Merge "res_rtp_asterisk: Enable Forward Secrecy (PFS) for DTLS." into 13zuul
2016-07-14Merge "stasis_endpoint.c: Fix contactstatus_to_json()." into 13zuul
2016-07-14app_queue: Only remove queue member from pending when state changes.Joshua Colp
It is possible for a not in use state change to occur multiple times causing a queue member to be removed from the pending call container prematurely. The first not in use state change will remove the queue member from the container. At this moment the member may be called and placed in the pending container. After this another not in use state change can be received which will remove it from the container. Despite being called at this point the code will incorrectly see that there are no pending calls to it. This change only removes it from the pending container if the state has actually changed. ASTERISK-26133 #close patches: app_queue.diff submitted by Richard Miller (license 5685) Change-Id: Ie5a7f17a44f98e9159e9b85009ce3f8393aa78c0
2016-07-14Merge "pjsip_options.c: Fix container operation." into 13zuul
2016-07-14Merge "pjsip_configuration.c: Misc cleanups." into 13zuul
2016-07-14pbx: Fix leak of timezone for time based includes.Corey Farrell
Create include_free to run ast_destroy_timing and ast_free, use that in all places that freed an ast_include structure. This fixes a couple of paths that previously did not run ast_destroy_timing. ASTERISK-26196 #close Change-Id: I1671bd111bef0dc113e8bf8f77f89fcfc395d838
2016-07-13Merge "chan_sip: Fix reference leak in mwi_event_cb" into 13zuul
2016-07-13Merge "res/res_pjsip_session: Check for presence of an active negotiator" ↵zuul
into 13
2016-07-13Merge "res/res_pjsip_pubsub: Add additional debug statements" into 13Joshua Colp
2016-07-13Merge "res/res_corosync: Raise a Stasis message on node join/leave events" ↵Joshua Colp
into 13
2016-07-13translate: explicit format destination not properly setKevin Harwell
If the destination format's name differed from the codec name then the translator's explict_dst field would be improperly set. In some circumstances it would end up setting it to a newly created format that has the same name as the codec when it actually needed to be the given destination codec. This could cause the translation path to use the wrong format. For instance, if an endpoint had specified 'myulaw' as a format the translator could end up using a 'ulaw' format (with whatever/default settings) instead. If the format attribute settings differed between the two then there may unexpected results during processing. This patch removes the name check when building the translation path. This should make it always set the translator's explicit_dst to the given destination format as long as the sample rate and types match. Change-Id: Iaf8a03831d68e657d89569d54b505074efbefab5
2016-07-13stasis_endpoint.c: Fix contactstatus_to_json().Richard Mudgett
The roundtrip_usec json member is optional. If it isn't present then don't put it into the converted json structure where ast_json_pack() will choke on it. Change-Id: I39bb2f86154ef54591270c58bfda8635070f9ea0
2016-07-13chan_sip: Fix reference leak in mwi_event_cbCorey Farrell
Cleanup the peer reference when stasis_subscription_final_message is true. Also free peer_name even if peer exists, after reload a new peer_name will be allocated. ASTERISK-26193 #close Change-Id: If7ecd52facdc5c227f701c760841e3f6ca53cc69
2016-07-13res_rtp_asterisk: Enable Forward Secrecy (PFS) for DTLS.Alexander Traud
Since July 2014, TLS based protocols (SIP over TLS, Secure WebSockets, HTTPS) support PFS thanks to ASTERISK-23905. In July 2015, the same feature was added for DTLS. The source code from main/tcptls.c should have been re-used to ease security audits. Therefore, this change rolls back the change from July 2015 and re-uses the code from July 2014. This has the additional benefits to work under CentOS 7 and enabling not just ECDHE but DHE based cipher suites as well. ASTERISK-25659 #close Reported by: StefanEng86, urbaniak, pay123 Tested by: sarumjanuch, traud patches: res_rtp_asterisk.patch submitted by sarumjanuch dtls_centos_step_1.patch submitted by traud dtls_centos_step_2.patch submitted by traud Change-Id: I537cadf4421f092a613146b230f2c0ee1be28d5c
2016-07-13threadpool: Fix leak in ast_threadpool_serializer_group error path.Corey Farrell
ast_threadpool_serializer_group leaks a reference to ser when listener is allocated but tps is not. Although listener takes the reference to ser cleanup functions are not run without tps. ASTERISK-26191 #close Change-Id: Ie3ccf69a3f1e676c2ef62a77067c0cb57dc9a585
2016-07-13pjsip_options.c: Fix container operation.Richard Mudgett
aor_observer_deleted() needs to operate on all contacts found for the deleted AOR instead of only the first one found. This is really only a problem if there is more than one contact for the AOR. Change-Id: Id24ac0d5e8c931330231fb45dd2a331a84339dc1
2016-07-13pjsip_configuration.c: Misc cleanups.Richard Mudgett
* Fix some whitespace in various routines. * Rename i to iter in persistent_endpoint_update_state(). * Fix off-nominal copy/paste message wording in persistent_endpoint_contact_deleted_observer() Change-Id: Id8e34f5d09e7eebac3af22501c44c1110a3e29d8
2016-07-13BuildSystem: Avoid obsolete warning with pthread.m4 on autoconf.Alexander Traud
Updated the macro-set autoconf/ax_pthread.m4 to its latest upstream version. ASTERISK-26046 #close Change-Id: I11abc11d17acd2b6a8a5a5be8ae8e0949dab9cc7
2016-07-13Merge "res_pjsip: Fix statsd regression." into 13zuul
2016-07-13Merge "BuildSystem: Allow own CFLAGS on ./configure." into 13zuul
2016-07-12Merge "install_prereq: Checkout of libSRTP 1.5.x." into 13zuul
2016-07-12Merge "chan_sip: Fix reference leaks in error paths." into 13zuul
2016-07-12Merge "res_sorcery_realtime: fix bug when successful UPDATE is treated as ↵zuul
failed" into 13
2016-07-12Merge "res_pjsip: Added "subscribe_context" to endpoint" into 13zuul
2016-07-12Merge "BuildSystem: Avoid obsolete warning with libcurl.m4 on autoconf." into 13zuul
2016-07-12Merge "func_odbc: Fix connection deadlock." into 13zuul
2016-07-12res_pjsip: Fix statsd regression.Richard Mudgett
The ASTERISK-25904 change-id I8fad8aae9305481469c38d2146e1ba3a56d3108f patch introduced several regressions when the newly created "Updated" state goes out for each endpoint registration refresh. 1) It restarted any OPTIONS RTT ping cycle. 2) It would interfere with a currently active ping and throw off that ping's resulting RTT calculation. 3) It cleared the RTT time each time the endpoint was refreshed. 4) The cleared RTT time was sent out as a statsd update each time. 5) It created two AMI events for each update. * Revert the original patch and reimplement it. Now the current contact status state is re-sent instead of the state being momentarily toggled every time the endpoint refreshes its registration. The statsd events are not created for the re-sent refresh because they are sent after every OPTIONS ping. ASTERISK-26160 #close Reported by: Matt Jordan Change-Id: Ie072be790fbb2a8f5c1c874266e4143fa31f66d1
2016-07-12BuildSystem: Allow own CFLAGS on ./configure.Alexander Traud
Before this change, make failed with the error Unknown value '' found in build_tools/menuselect-deps for NATIVE_ARCH when CFLAGS were supplied to the configure script. This was introduced with <https://reviewboard.asterisk.org/r/1852/> which disabled BUILD_NATIVE when CFLAGS were supplied. Those who need different -march= values, please, go for ./configure make menuselect.makeopts or make menuselect ./menuselect/menuselect --disable BUILD_NATIVE ASTERISK-25289 #close Change-Id: Ic6365d5a97bb9b3556858f06432a8d1cfa83eebc
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
2016-07-11install_prereq: Checkout of libSRTP 1.5.x.Alexander Traud
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
2016-07-10func_odbc: Fix connection deadlock.Joshua Colp
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
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