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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
2016-07-09chan_sip: Fix reference leaks in error paths.Corey Farrell
* 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
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
2016-07-07res_sorcery_realtime: fix bug when successful UPDATE is treated as failedAlexei Gradinari
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
2016-07-05res_pjsip: Added "subscribe_context" to endpointAlexei Gradinari
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
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." 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
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