mirror of http://gerrit.asterisk.org/asterisk
master
20
18
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28147 Commits (be36bd7ca565531c09ca444e59b6d057131a9ab0)
Author | SHA1 | Message | Date |
---|---|---|---|
|
be36bd7ca5 |
pbx: Create pbx_include.c for management of 'struct ast_include'.
This changes context includes from a linked list to a vector, makes 'struct ast_include' opaque to pbx.c. Although ast_walk_context_includes is maintained the procedure is no longer efficient except for the first call (inc==NULL). This functionality is replaced by two new functions implemented by vector macros. * ast_context_includes_count (AST_VECTOR_SIZE) * ast_context_includes_get (AST_VECTOR_GET) As with ast_walk_context_includes callers of these functions are expected to have locked contexts. Only a few places in Asterisk walked the includes, they have been converted to use the new functions. const have been applied where possible to parameters for ast_include functions. Change-Id: Ib5c882e27cf96fb2aec67a39c18b4c71c9c83b60 |
9 years ago |
|
273052f404 |
Update support for SILK format.
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 |
9 years ago |
|
3cf33dd4e7 |
Merge "translate: explicit format destination not properly set"
|
9 years ago |
|
afcc519dff |
Merge "threadpool: Fix leak in ast_threadpool_serializer_group error path."
|
9 years ago |
|
707dbcbcd1 |
Merge "pbx: Fix leak of timezone for time based includes."
|
9 years ago |
|
bea3e9b6fb |
Merge "BuildSystem: Avoid obsolete warning with pthread.m4 on autoconf."
|
9 years ago |
|
89f0a7d3f4 |
Merge "res_rtp_asterisk: Enable Forward Secrecy (PFS) for DTLS."
|
9 years ago |
|
f0f137e5bd |
Merge "stasis_endpoint.c: Fix contactstatus_to_json()."
|
9 years ago |
|
153875be24 |
Merge "pjsip_options.c: Fix container operation."
|
9 years ago |
|
43596895f9 |
Merge "pjsip_configuration.c: Misc cleanups."
|
9 years ago |
|
5703374cc5 |
Merge "chan_sip: Fix reference leak in mwi_event_cb"
|
9 years ago |
|
f3608b50d7 |
pbx: Fix leak of timezone for time based includes.
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 |
9 years ago |
|
2567e57624 |
Merge "res/res_corosync: Raise a Stasis message on node join/leave events"
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9 years ago |
|
3849f23bff |
Merge "res/res_pjsip_session: Check for presence of an active negotiator"
|
9 years ago |
|
63ac4c9487 |
translate: explicit format destination not properly set
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 |
9 years ago |
|
2f26512fd8 |
stasis_endpoint.c: Fix contactstatus_to_json().
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 |
9 years ago |
|
bc1ff41be7 |
pjsip_options.c: Fix container operation.
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 |
9 years ago |
|
eabcfeeaa3 |
pjsip_configuration.c: Misc cleanups.
* 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 |
9 years ago |
|
f73ddde7d4 |
chan_sip: Fix reference leak in mwi_event_cb
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 |
9 years ago |
|
fd54d69feb |
threadpool: Fix leak in ast_threadpool_serializer_group error path.
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 |
9 years ago |
|
85212f2799 |
res_rtp_asterisk: Enable Forward Secrecy (PFS) for DTLS.
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 |
9 years ago |
|
0d487b53b1 |
res/res_pjsip_session: Check for presence of an active negotiator
It is possible in a hypothetical situation for a session refresh to be invoked on a PJSIP when the negotiatior on the INVITE session has not yet been established. While this shouldn't occur with existing uses of ast_sip_session_refresh, the crashes that occur due to improperly calling PJSIP functions that expect a non-NULL negotiatior are avoidable. PJSIP will create the negotiator in pjsip_inv_reinvite; this means that simply checking for the presence of the negotiator before passing it to other PJSIP functions that use it is allowable. As such, this patch adds checks for the presence of the negotiator before calling PJSIP functions that assume it is non-NULL. Change-Id: I1028323e7e01b0a531865e5412a71b6f6ec4276d |
9 years ago |
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c49833653b |
res/res_pjsip_pubsub: Add additional debug statements
When something very sad and wrong occurs, it's challenging sometimes to figure out why. This patch adds some additional debug statements on off-nominal paths to try and make debugging easier. Change-Id: I7bffb73cc733b6f80193a23340881db4a102b640 |
9 years ago |
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f12311ee69 |
res/res_corosync: Raise a Stasis message on node join/leave events
When res_corosync detects that a node leaves or joins, it currently is informed of this via Corosync callbacks. However, there are a few limitations with the information presented: (1) While we have information that Corosync is aware of - such as the Corosync nodeid - that information is really only useful inside of Corosync or res_corosync. There's no way to translate a Corosync nodeid to some other internally useful unique identifier for the Asterisk instance that just joined or left the cluster. (2) While res_corosync is notified of the instance joining or leaving the cluster, it has no mechanism to inform the Asterisk core or other modules of this event. This limits the usefulness of res_corosync as a heartbeat mechanism for other modules. This patch addresses both issues. First, it adds the notion of a cluster discovery message both within the Stasis message bus, as well as the binary event messages that res_corosync uses to transmit data back and forth within the cluster. When Asterisk joins the cluster, it sends a discovery message to the other nodes in the cluster, which correlates the Corosync nodeid along with the Asterisk EID. res_corosync now maintains a hash of Corosync nodeids to Asterisk EIDs, such that it can map changes in cluster state with the Asterisk instance that has that nodeid. Likewise, when an Asterisk instance receives a discovery message from a node in the cluster, it now sends its own discovery message back to the originating node with the local Asterisk EID. This lets Asterisk instances within the cluster build a complete picture of the other Asterisk instances within the cluster. Second, it publishes the discovery messages onto the Stasis message bus. Said messages are published whenever a node joins or leaves the cluster. Interested modules can subscribe for the ast_cluster_discovery_type() message under the ast_system_topic() and be notified when changes in cluster state occur. Change-Id: I9015f418d6ae7f47e4994e04e18948df4d49b465 |
9 years ago |
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a3f4141f6f |
BuildSystem: Avoid obsolete warning with pthread.m4 on autoconf.
Updated the macro-set autoconf/ax_pthread.m4 to its latest upstream version. ASTERISK-26046 #close Change-Id: I11abc11d17acd2b6a8a5a5be8ae8e0949dab9cc7 |
9 years ago |
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73d8cb587d |
Merge "rest_api/channels: Fix multiple issues with create and dial"
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9 years ago |
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e049248161 |
Merge "res_pjsip: Fix statsd regression."
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9 years ago |
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c48016e2f2 |
Merge "BuildSystem: Allow own CFLAGS on ./configure."
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9 years ago |
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c2a72e6aa6 |
Merge "install_prereq: Checkout of libSRTP 1.5.x."
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9 years ago |
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260cd7c2cd |
Merge "chan_sip: Fix reference leaks in error paths."
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9 years ago |
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69796bf5fe |
Merge "res_sorcery_realtime: fix bug when successful UPDATE is treated as failed"
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9 years ago |
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90d4ebbb40 |
Merge "res_pjsip: Added "subscribe_context" to endpoint"
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9 years ago |
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8654727eb7 |
Merge "BuildSystem: Avoid obsolete warning with libcurl.m4 on autoconf."
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9 years ago |
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886f2cab23 |
rest_api/channels: Fix multiple issues with create and dial
* We weren't properly subscribing to the channel and it's originator on create. * We weren't doing a publish_dial after calling ast_call on dial. * We weren't calling depart_bridge when a channel left the dial bridge. The first 2 issues were causing events to not be generated and the third was actually causing channels to not get properly destroyed when hung up. Together these 3 issues were causing the new rest_apichannels/create_dial_bridge tests to fail. As a result of the fixes, the cdr state machine had to be slightly tweaked to allow bridge leave events without asserting and the tests themselves had to be updated to account for the channels now cleaning themselves up. Change-Id: Ibf23abf5a62de76e82afb4461af5099c961b97d8 |
9 years ago |
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b85446d039 |
res_pjsip: Fix statsd regression.
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 |
9 years ago |
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4ad333bb0e |
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 |
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110b01a0bc |
BuildSystem: Allow own CFLAGS on ./configure.
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 |
9 years ago |
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44f16af7cc |
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 |
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8476a9332f |
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 |
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ad30d60c69 |
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 |
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e0f27ecabb |
Merge "chan_sip/res_pjsip_t38: Handle a request to negotiate T.38 after it is enabled."
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9 years ago |
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99cbecd270 |
Merge "REF_DEBUG: Prevent logging of container node objects."
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9 years ago |
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7408c51a48 |
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 |
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c832f100d9 |
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 |
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302be4809a |
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 |
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fb96492ec4 |
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 |
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1c949eea6c |
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 |
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32cb981d04 |
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 |
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9e10aa8496 |
Merge "res_pjsip_session.c: Don't send extra BYE if SDP invalid."
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9 years ago |
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764a009fbe |
Merge "res_pjsip_session.c: End call on initial invalid SDP negotiation."
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9 years ago |