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Author	SHA1	Message	Date
Sean Bright	acfc06cfc0	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
Alexei Gradinari	7d79805d54	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
George Joseph	efadf89131	Merge "res_rtp_asterisk: Raise event when RTP port is allocated" into 13	7 years ago
Joshua Colp	49116b55d9	Merge "CI: Add --test-timeout option to runTestsuite.sh" into 13	7 years ago
Corey Farrell	1f5c2a2d0d	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
Moritz Fain	27c46b8ff3	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
Cao Minh Hiep	74c5c1cd1b	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
George Joseph	4ddca53164	Merge "jansson: Backport fixes to bundled, use json_vsprintf if available." into 13	7 years ago
George Joseph	5b8b18cbd2	Merge "chan_sip: SipNotify on Chan_Sip vi AMI behave different to CLI" into 13	7 years ago
George Joseph	97f4290421	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	63c02eeba4	Merge "rtp_engine: rtcp_report_to_json can overflow the ssrc integer value" into 13	7 years ago
Joshua Colp	155ff8e174	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	9c35f80adf	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	594bbbea57	Merge "app_voicemail: Fix stack overrun in append_mailbox" into 13	7 years ago
George Joseph	a509bb729a	Merge "app_voicemail: Cleanup mailbox topic and cache" into 13	7 years ago
George Joseph	984af7a1e8	Merge "stasis: Add function to delete topic from pool" into 13	7 years ago
George Joseph	eda4e86df2	Merge "channel.c: Address stack overflow in does_id_conflict()" into 13	7 years ago
Kevin Harwell	286cf80bec	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	656b3e85cf	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	1948fbe439	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
Alexei Gradinari	37c1f35922	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
Sean Bright	3033242a7b	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	a741cd4f11	Merge "stasis: No need to keep a stasis type ref in a stasis msg or cache object." into 13	7 years ago
George Joseph	7bdf1d3c67	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
Sean Bright	33ca3664ca	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	1d141112b5	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
George Joseph	94a4eea7f6	stasis_cache: Stop caching stasis subscription change messages ... Since app_voicemail no longer uses the cache to maintain its state there is no longer a need to cache these messages. ASTERISK-27121 Change-Id: I321c708505f5ad8d00e1b0afc4c27dc2ac12ecb4	7 years ago
Joshua Colp	bce2a09793	Merge "pjproject: Update initial 2.8 patches to apply cleanly." into 13	7 years ago
Joshua Colp	b2310c5434	Merge "app_voicemail: Remove need to subscribe to stasis" into 13	7 years ago
Richard Mudgett	ac18bb23a9	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	418eb22ba6	pjproject: Update initial 2.8 patches to apply cleanly. ... ASTERISK-28059 Change-Id: I027472f2753391646dde594a709a75f14422db93	7 years ago
Joshua Colp	f4294baf21	Merge "alembic: fix suppress_q850_reason_headers column name" into 13	7 years ago
Richard Mudgett	2f84ff9728	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
George Joseph	c4f1adf78e	Merge "pjproject: Upgrade to 2.8." into 13	7 years ago
Florian Floimair	8539f6a657	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
pk16208	84c574bb8b	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	1843b0e2b5	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	cd2deadb69	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
Sean Bright	99b2e0c2ff	autoconf: Check for srtp_get_version_string() before using it ... Change-Id: Id2a916ff9448706090e72ff2c7fb3f5ba24a05df	7 years ago
George Joseph	926ac196af	Merge "res_srtp.c: Show linked version of libsrtp on module init" into 13	7 years ago
George Joseph	3a09d9c74c	Merge "res_pjsip: Log IPv6 addresses correctly" into 13	7 years ago
George Joseph	0b88512a4d	CI: Fix typo in testsuite git checkout ... Change-Id: I30024515e5b00a5044fd39fbff27d818f016b719	7 years ago
Sean Bright	44375c0616	res_srtp.c: Show linked version of libsrtp on module init ... Change-Id: Ib0a645d6985de5757cc4399ed2524b2d02c4f342	7 years ago
Sean Bright	d3c869c736	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	52324ef1a1	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
Jenkins2	d4becda8ec	Merge "res_musiconhold.c: Restart MOH if previous hold just reached end-of-file" into 13	7 years ago
Jenkins2	641e5b7e63	Merge "optional_api: Remove unused nonoptreq fields" into 13	7 years ago
George Joseph	dd429cd4d8	Merge "CI: Use .gitreview to default BRANCH_NAME." into 13	7 years ago
Joshua Colp	55a306deef	Merge "res_pjproject: Fix sockaddr conversion routines for non-bundled PJSIP" into 13	7 years ago
Corey Farrell	d1f6a323a0	CI: Use .gitreview to default BRANCH_NAME. ... This ensures that binary modules are avoided in the master branch even if BRANCH_NAME is not set. Change-Id: I79162d2063f22fa9d6b31fde4827ace2dd5bf0da	7 years ago