asterisk

Commit Graph

Author	SHA1	Message	Date
zuul	c2aed1c6e0	Merge "pbx.c: Allow dangerous functions when adding a hint to dialplan." into 13	10 years ago
Joshua Colp	cc5660406c	Merge "dsp.c: Add fax and DTMF detection unit tests." into 13	10 years ago
Joshua Colp	cc825e54b0	Merge "dsp.c: Added descriptive comments to Goertzel calculations." into 13	10 years ago
Joshua Colp	323042d99d	Merge "dsp.c: Fix incorrect format reference typo." into 13	10 years ago
Richard Mudgett	873fc0fda5	pbx.c: Allow dangerous functions when adding a hint to dialplan. We can allow dangerous functions when adding a hint since altering dialplan is itself a privileged activity. Otherwise, we could never execute dangerous functions. ASTERISK-25996 #close Reported by: Andrew Nagy Change-Id: I4929ff100ad1200a0198262d069a34f2296e77ba	10 years ago
Kevin Harwell	852e763571	rtp_engine: Failed assertion and wrong name given for codec Fixed an assert check that would trigger when the passed in value was negative. The negative value was being cast to an unsigned value. This resulted in the check failing. Also fixed another problem when loading formats in the engine. When setting the mime type the format's name was being passed in instead of the codec's name. Change-Id: I1a201cd419ba4d8e9a40d337e36b6fbe1737192c	10 years ago
Richard Mudgett	e8c34680ca	dsp.c: Add fax and DTMF detection unit tests. * Add fax amplitude and frequency sweep tests. * Add DTMF amplitude and twist unit tests. Change-Id: I8d77c9a1eec89e440d715f998c928687e870c3f7	10 years ago
Richard Mudgett	c1f240b818	dsp.c: Added descriptive comments to Goertzel calculations. * Added doxygen to describe some struct members and what is going on in the code. Change-Id: I2ec706a33b52aee42b16dcc356c2bd916a45190d	10 years ago
Richard Mudgett	003a52fd62	dsp.c: Fix incorrect format reference typo. Change-Id: Ia131da3ec29acf385cb43a586a29ecc975eb3896	10 years ago
Richard Mudgett	159e437e5a	dsp.c: Fix erroneous fax tone detection. The Goertzel calculations get less accurate the lower the signal level being worked with becomes because there is less resolution remaining. If it is too low we can erroneously detect a tone where none really exists. The searched for fax frequencies not only need to be so much stronger than the background noise they must also be a minimum strength. * Add needed minimum threshold test to tone_detect(). * Set TONE_THRESHOLD to allow low volume frequency spread detection. ASTERISK-26237 #close Reported by: Richard Mudgett Change-Id: I84dbba7f7628fa13720add6a88eae3b129e066fc	10 years ago
Joshua Colp	efebb1b9e0	Merge "res_pjsip: Add fax_detect_timeout endpoint option." into 13	10 years ago
Corey Farrell	7fdf7c3d4c	Add conditional support for noreturn functions. 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	10 years ago
Richard Mudgett	851b1c3a17	res_pjsip: Add fax_detect_timeout endpoint option. The new endpoint option allows the PJSIP channel driver's fax_detect endpoint option to timeout on a call after the specified number of seconds into a call. The new feature is disabled if the timeout is set to zero. The option is disabled by default. ASTERISK-26214 Reported by: Richard Mudgett Change-Id: Id5a87375fb2c4f9dc1d4b44c78ec8735ba65453d	10 years ago
Corey Farrell	76d4983c15	features.c: Remove unneeded adsi.h include. adsi.h is no longer used by features.c since parking was moved to a module. Change-Id: I2248b8a455225a17cb6ddaafd6c20c511a1eaf59	10 years ago
zuul	4b2031226d	Merge "Update support for SILK format." into 13	10 years ago
Mark Michelson	28501051b4	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	10 years ago
zuul	714ceb88df	Merge "translate: explicit format destination not properly set" into 13	10 years ago
Joshua Colp	8892029a03	Merge "threadpool: Fix leak in ast_threadpool_serializer_group error path." into 13	10 years ago
zuul	b4976b9038	Merge "pbx: Fix leak of timezone for time based includes." into 13	10 years ago
zuul	3c1a40eb8d	Merge "stasis_endpoint.c: Fix contactstatus_to_json()." into 13	10 years ago
Corey Farrell	a17b071e36	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	10 years ago
Joshua Colp	355fc081e6	Merge "res/res_corosync: Raise a Stasis message on node join/leave events" into 13	10 years ago
Kevin Harwell	8cef8f35e7	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	10 years ago
Richard Mudgett	afbd10b0c5	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	10 years ago
Corey Farrell	672a64bda3	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	10 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	10 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	10 years ago
Matt Jordan	1ec4f8dd00	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	10 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	10 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	10 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	10 years ago
zuul	43612a84c8	Merge "Fix Alembic upgrades." into 13	10 years ago
Mark Michelson	1b79e2deff	Fix Alembic upgrades. 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	10 years ago
zuul	b0e71c6571	Merge "fix: memory leaks, resource leaks, out of bounds and bugs" into 13	10 years ago
Alexei Gradinari	5134a8043a	fix: memory leaks, resource leaks, out of bounds and bugs ASTERISK-26119 #close Change-Id: Iecbf7d0f360a021147344c4e83ab242fd1e7512c	10 years ago
Alexander Traud	76516bd79d	http: leverage 'bindaddr' for TLS in http.conf The internal HTTP/WebSocket server supports both TCP and TLS, which can be activated separately via the file http.conf. The source code intends to re-use the TCP parameter 'bindaddr' for TLS, even if 'tlsbindaddr' is not specified explicitly. This did not work because of a typo. This change resolves this typo. ASTERISK-26126 #close Change-Id: I5efb0409ae12044dfb3495b6b97b6d40a8c9c51f	10 years ago
Alexander Traud	ca38a3cbb4	core: Not the configured but granted number of possible file descriptors. With CLI "core show settings", simply the parameter maxfiles of the file asterisk.conf was shown. If that parameter was not set, nothing was displayed although the environment might have set a default number itself. Or if maxfiles were not granted (completely), still maxfiles was shown. Now, the maximum number of possible file descriptors in the environment is shown. ASTERISK-26097 Change-Id: I2df5c58863b5007b34b77adbe28b885dfcdf7e0b	10 years ago
Joshua Colp	c2f68787a3	Merge "cel: Ensure only one dial status per channel exists." into 13	10 years ago
Joshua Colp	ecc186a4cc	Merge "stasis: Add setting subscription congestion levels." into 13	10 years ago
Joshua Colp	e842a99e7c	Merge "sorcery: Add setting object type congestion levels." into 13	10 years ago
zuul	1ac47c5aae	Merge "taskprocessors: Implement high/low water mark alerts." into 13	10 years ago
Joshua Colp	caf6cccc5c	cel: Ensure only one dial status per channel exists. CEL wrongly assumed that a channel would only have a single dial event on it. This is incorrect. Particularly in a queue each call attempt to a member will result in a dial event, adding a new dial status in CEL without removing the old one. This would cause the container to grow with only one dial status being removed when the channel went away. The other dial status entries would remain leaking memory. This change fixes the memory leak by ensuring that only one dial status will only ever exist for each channel. The behavior during the scenario where multiple events are received has also been improved. For failure cases the first failure will be the dial status. If an answer dial status is received, though, it will take priority and the dial status for the channel will be answer. Memory usage has also been decreased by storing the minimal amount of information and the code has been cleaned up slightly. ASTERISK-25262 #close Change-Id: I5944eb923db17b6a0faa7317ff6abc9307c009fe	10 years ago
zuul	bd3b6e9f19	Merge "astfd: Not maximum size of a single file but maximum file descriptors." into 13	10 years ago
zuul	c09af61c76	Merge "BuildSystem: Avoid 'ar cru' and use 'ar cr' instead." into 13	10 years ago
Joshua Colp	f834852137	Merge "Fix #include poll.h and sys/cdefs.h" into 13	10 years ago
Alexander Traud	0d84421f93	astfd: Not maximum size of a single file but maximum file descriptors. With menuselect "DEBUG_FD_LEAKS" and CLI "core show fd", the maximum size of a single file was shown. Now, the maximum number of possible file descriptors is shown. ASTERISK-26097 Change-Id: Icf98d145774b38cac144ca76d19eaef42ce659a3	10 years ago
zuul	c9b873add8	Merge "ari/resource_channels: Add 'formats' to channel create/originate" into 13	10 years ago
Timo Teräs	9c5a0b814b	Fix #include poll.h and sys/cdefs.h POSIX defines poll.h, sys/poll.h should not be used at is c-library internal header which may or may not exist. Notable in musl it generates warning of being incorrect. And add explict include of sys/cdefs.h where needed. Change-Id: I142930df53fe7585a06b854b6faddc5301e024be	10 years ago
Richard Mudgett	557333ea4c	stasis: Add setting subscription congestion levels. Stasis subscriptions and message routers create taskprocessors to process the event messages. API calls are needed to be able to set the congestion levels of these taskprocessors for selected subscriptions and message routers. * Updated CDR, CEL, and manager's stasis subscription congestion levels based upon stress testing. Increased the congestion levels to reduce the potential for bursty call setup/teardown activity from triggering the taskprocessor overload alert. CDRs in particular need an extra high congestion level because they can take awhile to process the stasis messages. ASTERISK-26088 Reported by: Richard Mudgett Change-Id: Id0a716394b4eee746dd158acc63d703902450244	10 years ago
Richard Mudgett	110d772467	sorcery: Add setting object type congestion levels. Sorcery creates taskprocessors for object types to process object observer callbacks. An API call is needed to be able to set the congestion levels of these taskprocessors for selected object types. * Updated PJSIP's contact and contact_status sorcery object type observer default congestion levels based upon stress testing. Increased the congestion levels to reduce the potential for bursty register/unregister and subscribe/unsubscribe activity from triggering the taskprocessor overload alert. ASTERISK-26088 Reported by: Richard Mudgett Change-Id: I4542e83b556f0714009bfeff89505c801f1218c6	10 years ago

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5224 Commits (c2aed1c6e0522d9d193db8aeeb30b8c3dfeeaeb0)