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Author	SHA1	Message	Date
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
George Joseph	670fbbd702	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	21f472355e	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	65f63bd678	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	4181fa6772	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	51656adf35	Merge "stasis: No need to keep a stasis type ref in a stasis msg or cache object." into 15	7 years ago
Sean Bright	fd87b8e49c	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	80cc691252	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	66b2c87689	Merge "stasis_cache: Stop caching stasis subscription change messages" into 15	7 years ago
Joshua Colp	1d32a02c61	Merge "pjproject: Update initial 2.8 patches to apply cleanly." into 15	7 years ago
Joshua Colp	42bf0fd892	Merge "app_voicemail: Remove need to subscribe to stasis" into 15	7 years ago
Joshua Colp	87bac2057e	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
Richard Mudgett	f43463d0d3	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
Richard Mudgett	03c759d91a	pjproject: Update initial 2.8 patches to apply cleanly. ... ASTERISK-28059 Change-Id: I027472f2753391646dde594a709a75f14422db93	7 years ago
Joshua Colp	70af90e2f5	Merge "alembic: fix suppress_q850_reason_headers column name" into 15	7 years ago
Joshua Colp	e7bc861f88	Merge "res_pjsip_session: Don't add declined stream if one does not exist." into 15	7 years ago
George Joseph	98c4f7df7c	Merge "pjproject: Upgrade to 2.8." into 15	7 years ago
Richard Mudgett	a966a23023	stasis_message.c: Don't create immutable stasis objects with locks. ... * Create the stasis message object without a lock as it is immutable. * Create the stasis message type object without a lock as it is immutable. * Creating the stasis message type could crash if the passed in type name is NULL and REF_DEBUG is enabled. Added missing NULL check when passing the ao2 object tag string. Change-Id: I28763c58bb9f0b427c11971d0103bf94055e7b32	7 years ago
Joshua Colp	b5959113f6	pjproject: Upgrade to 2.8. ... This change brings in PJSIP 2.8, removes all the patches that were merged upstream, and makes a minor change to support a breaking change that was done. ASTERISK-28059 Change-Id: I5097772b11b0f95c3c1f52df6400158666f0a189	7 years ago
Florian Floimair	36ce08fa7d	alembic: fix suppress_q850_reason_headers column name ... In the original commit introducing the feature the column in the alembic script was called 'suppress_q850_reason_header'. In the code however the option is called 'suppress_q850_reason_headers' (trailing 's'). This leads to errors when ARI push configuration is used. Change-Id: Ie84808adbca6fcc9136556e4f5d741adbef5d14f	7 years ago
George Joseph	82f9f2db6b	app_voicemail: Remove need to subscribe to stasis ... app_voicemail was using the stasis cache to build and maintain a list of mailboxes that had subscribers. It then used this list to determine if a mailbox should be polled for new messages if polling was enabled. For this to work, stasis had to cache every subscription and unsubscription to the mailbox which caused a lot of overhead, both cpu and memory related. Since polling is only required when changes are being made to mailboxes outside of app_voicemail and since the number of mailboxes that don't have any subscribers is likely to be very low, all mailboxes are now polled instead of just the ones with subscribers. This paves the way for disabling the caching of stasis subscription change messages. Also fixed cleanup in some of the unit tests that not only left test users in the users list but also caused segfaults if the tests were run more than once. ASTERISK-27121 Change-Id: I5cceb737246949f9782955c64425b8bd25a9e9ee	7 years ago
Joshua Colp	5626b8ecae	res_pjsip_session: Don't add declined stream if one does not exist. ... Given a scenario where a session refresh was done with a removed stream we would always add a removed stream to the outgoing SDP even if one did not already exist. This change makes it so that a removed stream is only placed into the SDP if one already exists. ASTERISK-28047 Change-Id: Ibb97d21cdeb87a8acae0c720861b0ff255708442	7 years ago
Sean Bright	5713fc0b49	autoconf: Check for srtp_get_version_string() before using it ... Change-Id: Id2a916ff9448706090e72ff2c7fb3f5ba24a05df	7 years ago
George Joseph	16ec2f2645	Merge "res_srtp.c: Show linked version of libsrtp on module init" into 15	7 years ago
George Joseph	f83321d89f	Merge "res_pjsip: Log IPv6 addresses correctly" into 15	7 years ago
George Joseph	cb19ca981d	CI: Fix typo in testsuite git checkout ... Change-Id: I30024515e5b00a5044fd39fbff27d818f016b719	7 years ago
Sean Bright	644f7ef413	res_srtp.c: Show linked version of libsrtp on module init ... Change-Id: Ib0a645d6985de5757cc4399ed2524b2d02c4f342	7 years ago
Sean Bright	5f6bef4e5b	res_pjsip: Log IPv6 addresses correctly ... Both pjsip_tx_data.tp_info.dst_name and pjsip_rx_data.pkt_info.src_name store IPv6 addresses without enclosing brackets. This causes some log output to be confusing because it is difficult to separate the IPv6 address from a port specification. * Use pj_sockaddr_print() along with pjsip_tx_data.tp_info.dst_addr and pjsip_rx_data.pkt_info.src_addr where possible for consistent IPv6 output. * When a pj_sockaddr is not available, explicitly wrap IPv6 addresses in brackets. * When assigning pjsip_rx_data.pkt_info.src_name ourselves, make sure to also set pjsip_rx_data.pkt_info.src_addr. Change-Id: I5cfe997ced7883862a12b9c7d8551d76ae02fcf8	7 years ago
George Joseph	07d6e2bf16	CI: Use proper credentials for Security testsuite checkout ... Can't do anonymous http checkout from Security-testsuite. Need to use same credentials as the gerrit review checkout. Change-Id: I87af68c995cb8926f5e87f9af245600d76984f05	7 years ago