31751 Commits (4cbe04de77790c4c2950f1b65518be9a1c605627)

 

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Author	SHA1	Message	Date
Alexei Gradinari	ab90479642	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	3337fe85c5	jansson-bundled: Add patches to improve json_pack error reporting. ... Change-Id: I045e420d5e73e60639079246e810da6ae21ae22b	7 years ago
Corey Farrell	b3b95c8d36	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	0ad987168e	Merge "config.c: Cleanup AST_INCLUDE_GLOB" into 16	7 years ago
Kevin Harwell	2ee1c0cec8	Merge "res_odbc: fix missing SQL error diagnostic" into 16	7 years ago
George Joseph	60754fb806	Merge "astobj2: Fix shutdown order." into 16	7 years ago
George Joseph	15f91a4789	Merge "app_queue: Fix Attended transfer hangup with removing pending member." into 16	7 years ago
George Joseph	b4ab26e72f	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	de43d25716	Merge "res_rtp_asterisk.c: Add "seqno" strictrtp option" into 16	7 years ago
Corey Farrell	33455f8d6f	astobj2: Reduce memory overhead. ... Reduce options to 2-bit field, magic to 30 bit field. Move ref_counter next to options and explicitly use int32_t so 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	d1698b4e0b	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	06fef54b5a	Merge "res_rtp_asterisk: Raise event when RTP port is allocated" into 16	7 years ago
Joshua Colp	2e307ee0b7	Merge "CI: Add --test-timeout option to runTestsuite.sh" into 16	7 years ago
Corey Farrell	0fe08c5e79	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	958727cece	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	1f68614f3c	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	67e1e49e08	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	94ef074cc7	Merge "jansson: Backport fixes to bundled, use json_vsprintf if available." into 16	7 years ago
George Joseph	7638366c2f	Merge "chan_sip: SipNotify on Chan_Sip vi AMI behave different to CLI" into 16	7 years ago
Alexei Gradinari	e04c49b517	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	227d9719da	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	a6c3984175	Merge "rtp_engine: rtcp_report_to_json can overflow the ssrc integer value" into 16	7 years ago
pk16208	681e115ff1	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	cc8e0dfabd	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
Joshua Colp	7c49ea7cff	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
Corey Farrell	ede59966d9	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
Corey Farrell	7b0dbda936	json: Take advantage of new API's. ... * Use "o*" format specifier for optional fields in ast_json_party_id. * Stop using ast_json_deep_copy on immutable objects, it is now thread safe to just use ast_json_ref. Additional changes to ast_json_pack calls in the vicinity: * Use "O" when an object needs to be bumped. This was previously avoided as it was not thread safe. * Use "o?" and "O?" to replace NULL with ast_json_null(). The "?" is a new feature of ast_json_pack starting with Asterisk 16. Change-Id: I8382d28d7d83ee0ce13334e51ae45dbc0bdaef48	7 years ago
George Joseph	3ddfeccf35	Merge "app_voicemail: Fix stack overrun in append_mailbox" into 16	7 years ago
George Joseph	685abce1d0	Merge "chan_sip.c: chan_sip unstable with TLS after asterisk start or reloads" into 16	7 years ago
George Joseph	6a0dc87b97	Merge "res_remb_modifier: Add module for controlling REMB from CLI." into 16	7 years ago
George Joseph	82f28106df	Merge "app_voicemail: Cleanup mailbox topic and cache" into 16	7 years ago
George Joseph	64e3ed75ab	Merge "stasis: Add function to delete topic from pool" into 16	7 years ago
George Joseph	a389b020de	Merge "res_rtp_asterisk: Fix crash on ast_rtp_new failure." into 16	7 years ago
George Joseph	361295f192	Merge "channel.c: Address stack overflow in does_id_conflict()" into 16	7 years ago
Kevin Harwell	755febe497	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	ba91c73f8e	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
George Joseph	0a1a96d331	channel.c: Address stack overflow in does_id_conflict() ... does_id_conflict() was passing a pointer to an int to a callback that expected a pointer to a size_t. Found by the Address Sanitizer. Change-Id: I0ff542067eef63a14a60301654d65d34fe2ad503	7 years ago
Corey Farrell	7a7b21f3a0	res_rtp_asterisk: Fix crash on ast_rtp_new failure. ... ast_rtp_new free'd rtp upon failure, but rtp_engine.c would also call the destroy callback. Remove call to ast_free from ast_rtp_new, leave it to rtp_engine.c to initiate the full cleanup. Add error detection for the ssrc_mapping vector initialization. In rtp_allocate_transport set rtp->s = -1 in the failure path where we close that FD to ensure we don't try closing it twice. ASTERISK-27854 #close Change-Id: Ie02aecbb46228ca804e24b19cec2bb6f7b94e451	7 years ago
Sean Bright	880905e7eb	res_rtp_asterisk: Reset all settings on module reload ... 'rtpchecksums' and 'rtcpinterval' are not being reset to their defaults if they are not present in the updated configuration file. Change-Id: I1162e40199314d46cf3225d5e1271c4c81176670	7 years ago
George Joseph	fa1b836374	app_voicemail: Cleanup mailbox topic and cache ... app_voicemail wasn't properly cleaning up the stasis cache or the mwi topic pool when the module was unloaded or when a user was deleted as a result of a reload. This resulted in leaks in both areas. * app_voicemail now calls ast_delete_mwi_state_full when it frees a user structure and ast_delete_mwi_state_full in turn now calls the new stasis_topic_pool_delete_topic function to clear the topic from the pool. Change-Id: Ide23144a4a810e7e0faad5a8e988d15947965df8	7 years ago
George Joseph	dec6ebd9e1	Merge "stasis: No need to keep a stasis type ref in a stasis msg or cache object." into 16	7 years ago
Sean Bright	339bf0cf7b	AST-2018-009: Fix crash processing websocket HTTP Upgrade requests ... The HTTP request processing in res_http_websocket allocates additional space on the stack for various headers received during an Upgrade request. An attacker could send a specially crafted request that causes this code to overflow the stack, resulting in a crash. * No longer allocate memory from the stack in a loop to parse the header values. NOTE: There is a slight API change when using the passed in strings as is. We now require the passed in strings to no longer have leading or trailing whitespace. This isn't a problem as the only callers have already done this before passing the strings to the affected function. ASTERISK-28013 #close Change-Id: Ia564825a8a95e085fd17e658cb777fe1afa8091a	7 years ago
George Joseph	1a9c69d729	stasis: Add function to delete topic from pool ... There's been a long standing leak when using topic pools. The topics in the pool get cleaned up when the last pool reference is released but you can't remove a topic specifically. If you reloaded app_voicemail for instance, and mailboxes went away, their topics were left in the pool. * Added stasis_topic_pool_delete_topic() so modules can clean up topics from pools. * Registered the topic pool containers so it can be examined from the CLI when AO2_DEBUG is enabled. They'll be named "<topic_pool_name>-pool". Change-Id: Ib7957951ee5c9b9b4482af7b9b4349112d62bc25	7 years ago
Joshua Colp	84b60aeb70	Merge "stasis_cache: Stop caching stasis subscription change messages" into 16	7 years ago
hajekd	8811ab1803	chan_sip.c: chan_sip unstable with TLS after asterisk start or reloads ... Fixes random asterisk crash on start or reload with TLS phones. ASTERISK-28034 #close Reported-by: David Hajek Change-Id: I2a859f97dc80c348e2fa56e918214ee29521c4ac	7 years ago
Joshua Colp	36898351e5	Merge "pjproject: Update initial 2.8 patches to apply cleanly." into 16	7 years ago
Joshua Colp	2f38bcdfc0	res_remb_modifier: Add module for controlling REMB from CLI. ... This adds a module which registers a CLI command that can set the REMB bitrate value for REMB as it enters or exits Asterisk. This allows you to ignore what Asterisk or a client produces and is useful for demonstrations. This does not generate REMB frames, however, but just modifies them as they flow to or from a channel. Change-Id: Ib089427c46a4a36d645cecfe02406adb38c17bec	7 years ago
Joshua Colp	e313ec3a7e	Merge "alembic: fix suppress_q850_reason_headers column name" into 16	7 years ago
Joshua Colp	3c1a8be11e	Merge "app_voicemail: Remove need to subscribe to stasis" into 16	7 years ago
Richard Mudgett	f6695249a5	stasis: No need to keep a stasis type ref in a stasis msg or cache object. ... Stasis message types are global ao2 objects and we make stasis messages and cache entries hold references to them. Since there are currently situations where cache objects are never deleted, the reference count on the types can exceed 100000 and generate a FRACK assertion message. The stasis message cache could conceivably also have that many messages legitimately on large systems. The only down side to not holding the message type ref in the stasis message is it only makes a crash either at shutdown or when manually unloading a busy module slightly more likely. However, this is more exposing a pre-existing stasis shutdown ordering issue than a problem with not holding a message type ref in stasis messages. * Made stasis messages and cache entries no longer hold a ref to the message type. Change-Id: Ibaa28efa8d8ad3836f0c65957192424c7f561707	7 years ago