asterisk

Commit Graph

Author	SHA1	Message	Date
George Joseph	346cf5fb14	Merge "loader: Fix result of module reload error." into 15	7 years ago
George Joseph	33e1481694	Merge "core: Disable astobj2 locking for some common objects." into 15	7 years ago
George Joseph	8700c3d810	Merge "astobj2: Record lock usage to refs log when DEBUG_THREADS is enabled." into 15	7 years ago
Corey Farrell	fa08aed784	core: Disable astobj2 locking for some common objects. * ACO options * Indications * Module loader ref_debug object * Media index info and variants * xmldoc items These allocation locations were identified using reflocks.py on the master branch. Change-Id: Ie999b9941760be3d1946cdb6e30cb85fd97504d8	7 years ago
Corey Farrell	f8b71fadab	Resolve warning about duplicate 'dialplan' CLI. Change-Id: I029db1b4a32ccfb38374d6fe944dc430866f4b30	7 years ago
Corey Farrell	2c43e5bad4	loader: Fix result of module reload error. When a module reload fails we never set AST_MODULE_RELOAD_ERROR. This caused reload failures to incorrectly report 'No module found'. Change-Id: I5f3953e0f7d135e53ec797f24c97ee3f73f232e7	7 years ago
Corey Farrell	078459b3f2	astobj2: Record lock usage to refs log when DEBUG_THREADS is enabled. When DEBUG_THREADS is enabled we can know if the astobj2 mutex / rwlock was ever used, so it can be recorded in the REF_DEBUG destructor entry. Create contrib/scripts/reflocks.py to process locking used by allocator. This can be used to identify places where AO2_ALLOC_OPT_LOCK_NOLOCK should be used to reduce memory usage. Change-Id: I2e3cd23336a97df2692b545f548fd79b14b53bf4	7 years ago
Richard Mudgett	308c3f9cf6	app_queue.c: Fix json ref leak Declining the queue_member_status_type stasis message in stasis.conf causes these messages to leak json objects. * Add missing ast_json_unref() if the type is NULL in queue_publish_member_blob(). ASTERISK-28084 Change-Id: I691ecf49bd1f7d9c29182e1eee8c4bb7103be9fc	7 years ago
Jenkins2	9cedc7bc81	Merge "app_confbridge: Use bridge join hook to send join and leave events" into 15	7 years ago
Joshua Colp	9f9772ee9e	Merge "astobj2: Reduce memory overhead." into 15	7 years ago
Joshua Colp	5e22118a91	Merge "lock: Improve performance of DEBUG_THREADS." into 15	7 years ago
Joshua Colp	3be86d3f55	Merge "configure.ac: Check for unbound version >= 1.5" into 15	7 years ago
Joshua Colp	370abe7aac	Merge "res_pjsip: improve realtime performance on CLI 'pjsip show contacts'" into 15	7 years ago
Joshua Colp	8baa117ea8	Merge "jansson-bundled: Add patches to improve json_pack error reporting." into 15	7 years ago
Joshua Colp	d2805ca533	Merge "res_stasis: Fix stale data in ARI bridges" into 15	7 years ago
Alexei Gradinari	625b329f4e	res_pjsip: improve realtime performance on CLI 'pjsip show contacts' CLI command 'pjsip show contacts' inefficiently make a lot of DB requests. For example if there are 10k aors then asterisk requests these 10k records of aor and then does 10k requests of contact - one request per aor. Even if use 'like <pattern>' the asterisk requests all aor's and contact's records and then filters them by itself. This patch gathers contact's container by - retrieving all dynamic contacts by regex (filtered by reg_server) - retrieving all aors with permanent contacts - finally filters container by regex ASTERISK-28077 #close Change-Id: Id0ad65d14952a02fb213273a90f3f680a8149618	7 years ago
Corey Farrell	288bf43eed	jansson-bundled: Add patches to improve json_pack error reporting. Change-Id: I045e420d5e73e60639079246e810da6ae21ae22b	7 years ago
Corey Farrell	1ffbec4672	lock: Improve performance of DEBUG_THREADS. Add a volatile flag to lock tracking structures so we only need to use the global lock when first initializing tracking. Additionally add support for DEBUG_THREADS_LOOSE_ABI. This is used by astobj2.c to eliminate storage for tracking fields when DEBUG_THREADS is not defined. Change-Id: Iabd650908901843e9fff47ef1c539f0e1b8cb13b	7 years ago
George Joseph	160ab08840	Merge "config.c: Cleanup AST_INCLUDE_GLOB" into 15	7 years ago
Kevin Harwell	d89812e356	Merge "res_odbc: fix missing SQL error diagnostic" into 15	7 years ago
George Joseph	0298f4eb0c	Merge "astobj2: Fix shutdown order." into 15	7 years ago
George Joseph	386bda4ab0	Merge "app_queue: Fix Attended transfer hangup with removing pending member." into 15	7 years ago
George Joseph	48cf548654	app_confbridge: Use bridge join hook to send join and leave events The first attempt at publishing confbridge events to participants involved publishing them at the same time stasis events were created. This caused issues with bridge and channel locks. The second attempt involved publishing them when the stasis events were received by the code that published the confbridge AMI events. This caused timing issues because, depending on resources available, the event could be received before channels actually joined the bridge and would therefore fail to send messages to the participant. This attempt reverts to the original mechanism with one exception. The join and leave events are published via bridge join and leave hooks. This guarantees the states of the channels and bridge and provides deterministic timing for event publishing. Change-Id: I2660074f8a30a5224cb953d5e047ee84484a9036	7 years ago
George Joseph	b8a74b4f7d	Merge "res_rtp_asterisk.c: Add "seqno" strictrtp option" into 15	7 years ago
Corey Farrell	4afcd37d15	astobj2: Reduce memory overhead. Reduce options to 2-bit field, magic to 30 bit field. Move ref_counter next to options the fields will pack. This reduces memory overhead for every ao2 object by 8 bytes on x86_64. Change-Id: Idc1baabb35ec3b3d8de463c4fa3011eaf7fcafb5	7 years ago
Sean Bright	066cade82b	config.c: Cleanup AST_INCLUDE_GLOB * In main/config.c, AST_INCLUDE_GLOB is fixed to '1' making the #ifdefs pointless. * In utils/extconf.c, AST_INCLUDE_GLOB is never defined so there is a lot of dead code. Change-Id: I1bad1a46d7466ddf90d52cc724e997195495226c	7 years ago
George Joseph	34c2baf88f	Merge "res_rtp_asterisk: Raise event when RTP port is allocated" into 15	7 years ago
Joshua Colp	6ebf4ba433	Merge "CI: Add --test-timeout option to runTestsuite.sh" into 15	7 years ago
Corey Farrell	427c3e3152	astobj2: Fix shutdown order. When REF_DEBUG and AO2_DEBUG are both enabled we closed the refs log before we shutdown astobj2_container. This caused the AO2_DEBUG container registration container to be reported as a leak. Change-Id: If9111c4c21c68064b22c546d5d7a41fac430430e	7 years ago
Cao Minh Hiep	fc842169d4	app_queue: Fix Attended transfer hangup with removing pending member. This issue related to setting of holdtime, announcements, member delays. It works well if we set the member delays to "0" and no announcements and no holdtime.This issue will happen if we set member delays to "1", "2"... or announcements or holdtime and hangs up the call during processing it. And here is the reason: (At the step of answering a phone.) It takes care any holdtime, announcements, member delays, or other options after a call has been answered if it exists. Normally, After the call has been aswered, and we wait for the processing one of the cases of the member delays or hold time or announcements finished, "if (ast_check_hangup(peer))" will be not executed, then queue will be updated at update_queue(). Here, pending member will be removed. However, after the call has been aswered, if we hangs up the call during one of the cases of the member delays or hold time or announcements, "if (ast_check_hangup(peer))" will be executed. outgoing = NULL and at hangupcalls, pending members will not be removed. * This fixed patch will remove the pending member from container before hanging up the call with outgoing is NULL. ASTERISK-27920 Reported by: Cao Minh Hiep Tested by: Cao Minh Hiep Change-Id: Ib780fbf48ace9d2d8eaa1270b9d530a4fc14c855	7 years ago
Moritz Fain	dfa926337c	res_stasis: Fix stale data in ARI bridges Fixed an issue that resulted in "Allocation failed" each time an ARI request was made to start playing MOH on a bridge. In bridge_moh_create() we were attaching the after bridge callbacks to chan which is the ;1 channel of the unreal channel pair. We should have attached them to the ;2 channel which is pushed into the bridge by ast_unreal_channel_push_to_bridge(). The callbacks are called when the specific channel leaves the bridging system. Since the ;1 channel is never put into a bridge the callbacks never get called. The callbacks then never remove the moh_wrapper from the app_bridges_moh container. As a result we cannot find the channel associated with the wrapper to start MOH because it has hungup. This is the reason causing the reported issue. * Rather than using after bridge callbacks to cleanup, we now have moh_channel_thread() doing the cleanup when the channel hangs up. * Fixed moh_channel_thread() accumulating control frames on the stasis bridge MOH channel until MOH is stopped. Control frames are no longer accumulated while MOH is playing. * Fixed channel ref counting issue. stasis_app_bridge_moh_channel() may or may not return a channel ref. As a result ast_ari_bridges_start_moh() wouldn't know it may have a channel ref to release. stasis_app_bridge_moh_channel() will now return a ref with the channel it returns. * Eliminated RAII_VAR in bridge_moh_create(). ASTERISK-26094 #close Change-Id: Ibff479e167b3320c68aaabfada7e1d0ef7bd548c	7 years ago
Ben Ford	166c130eac	res_rtp_asterisk.c: Add "seqno" strictrtp option When networks experience disruptions, there can be large gaps of time between receiving packets. When strictrtp is enabled, this created issues where a flood of packets could come in and be seen as an attack. Another option - seqno - has been added to the strictrtp option that ignores the time interval and goes strictly by sequence number for validity. Change-Id: I8a42b8d193673899c8fc22fe7f98ea87df89be71	7 years ago
George Joseph	a0d5f45baa	Merge "jansson: Backport fixes to bundled, use json_vsprintf if available." into 15	7 years ago
George Joseph	d95515449c	Merge "chan_sip: SipNotify on Chan_Sip vi AMI behave different to CLI" into 15	7 years ago
Alexei Gradinari	d03661bcaf	res_odbc: fix missing SQL error diagnostic On SQL error there is not diagnostic information about this error. There is only WARNING res_odbc.c: SQL Execute error -1! The function ast_odbc_print_errors calls a SQLGetDiagField to get the number of available diagnostic records, but the SQLGetDiagField returns 0. However SQLGetDiagRec could return one diagnostic records in this case. Looking at many example of getting diagnostics error information I found out that the best way it's to use only SQLGetDiagRec while it returns SQL_SUCCESS. Also this patch adds calls of ast_odbc_print_errors on SQL_ERROR to res_config_odbc. ASTERISK-28065 #close Change-Id: Iba5ae5470ac49ecd911dd084effbe9efac68ccc1	7 years ago
George Joseph	4927be3523	CI: Add --test-timeout option to runTestsuite.sh The default is 600 seconds. Also added timeouts to the *TestGroups.json files. Change-Id: I8ab6a69e704b6a10f06a0e52ede02312a2b72fe0	7 years ago
George Joseph	ff8f46bf75	Merge "rtp_engine: rtcp_report_to_json can overflow the ssrc integer value" into 15	7 years ago
pk16208	f3539d1d07	chan_sip: SipNotify on Chan_Sip vi AMI behave different to CLI With tls and udp enabled asterisk generates a warning about sending message via udp instead of tls. sip notify command via cli works as expected and without warning. asterisk has to set the connection information accordingly to connection and not on presumption ASTERISK-28057 #close Change-Id: Ib43315aba1f2c14ba077b52d8c5b00be0006656e	7 years ago
George Joseph	4c636f5377	configure.ac: Check for unbound version >= 1.5 In order to do this and provide good feedback, a new macro was created (AST_EXT_LIB_EXTRA_CHECK) which does the normal check and path setups for the library then compiles, links and runs a supplied code fragment to do the final determination. In this case, the final code fragment compares UNBOUND_VERSION_MAJOR and UNBOUND_VERSION_MINOR to determine if they're greater than or equal to 1.5. Since we require version 1.5, some code in res_resolver_unbound was also simplified. ASTERISK-28045 Reported by: Samuel Galarneau Change-Id: Iee94ad543cd6f8b118df8c4c7afd9c4e2ca1fa72	7 years ago
Corey Farrell	86ae100016	jansson: Backport fixes to bundled, use json_vsprintf if available. Use json_vsprintf from versions which contain fix for va_copy leak. Apply fixes from jansson master: * va_copy leak fix. * Avoid potential invalid memory read in json_pack. * Rename variable that shadowed another. Change-Id: I7522e462d2a52f53010ffa1e7d705c666ec35539	7 years ago
George Joseph	412a23c521	Merge "app_voicemail: Fix stack overrun in append_mailbox" into 15	7 years ago
Joshua Colp	af090e4fad	res_rtp_asterisk: Raise event when RTP port is allocated This change raises a testsuite event to provide what port Asterisk has actually allocated for RTP. This ensures that testsuite tests can remove any assumption of ports and instead use the actual port in use. ASTERISK-28070 Change-Id: I91bd45782e84284e01c89acf4b2da352e14ae044	7 years ago
George Joseph	4d7558f7d8	Merge "chan_sip.c: chan_sip unstable with TLS after asterisk start or reloads" into 15	7 years ago
George Joseph	2784fe688a	Merge "res_remb_modifier: Add module for controlling REMB from CLI." into 15	7 years ago
George Joseph	fd532f2d61	Merge "app_voicemail: Cleanup mailbox topic and cache" into 15	7 years ago
George Joseph	b5159c6a50	Merge "stasis: Add function to delete topic from pool" into 15	7 years ago
George Joseph	71f1e23173	Merge "res_rtp_asterisk: Fix crash on ast_rtp_new failure." into 15	7 years ago
George Joseph	34402b0f3c	Merge "channel.c: Address stack overflow in does_id_conflict()" into 15	7 years ago
Kevin Harwell	720f76def0	rtp_engine: rtcp_report_to_json can overflow the ssrc integer value When writing an RTCP report to json the code attempts to pack the "ssrc" and "source_ssrc" unsigned integer values as a signed int value type. This of course means if the ssrc's unsigned value is greater than that which can fit into a signed integer value it gets converted to a negative number. Subsequently, the negative value goes out in the json report. This patch now packs the value as a json_int_t, which is the widest integer type available on a given system. This should make it so the value no longer overflows. Note, this was caught by two failing tests hep/rtcp-receiver/ and hep/rtcp-sender. Change-Id: I2af275286ee5e795b79f0c3d450d9e4b28e958b0	7 years ago
George Joseph	47214d4de1	app_voicemail: Fix stack overrun in append_mailbox The append_mailbox function wasn't calculating the correct length to pass to ast_alloca and it wasn't handling the case where context might be empty. Found by the Address Sanitizer. Change-Id: I7eb51c7bd18a7a8dbdba261462a95cc69e84f161	7 years ago

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31418 Commits (346cf5fb147a6eafcd83fd655e512031fc9418c8) All Branches Search

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