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Author	SHA1	Message	Date
Joshua Colp	249d80f120	Merge "res_pjsip: Fix infinite recursion when loading transports from realtime" into 13	9 years ago
Joshua Colp	0e25e4a924	Merge "Build: Fix menuselect USAN conflicts" into 13	9 years ago
Badalyan Vyacheslav	bd07b6f0dd	Build: Added testing compiler to support the system sanitizes ... In older versions of the compiler was not sanitizes. Compilers other than GCC can not support the Usan and TSAN or have other options for *FLAGS. ASTERISK-25767 #close Reported by: Badalyan Vyacheslav Tested by: Badalyan Vyacheslav Change-Id: Iefce6608221fa87884b82ae3cb5649b7b1804916	9 years ago
Badalyan Vyacheslav	e9e896abd1	Build: Fix menuselect USAN conflicts ... USAN can be used together with other sanitizers. Reported by: Badalyan Vyacheslav Tested by: Badalyan Vyacheslav Change-Id: I3bffa350d70965c3026651dba3a12414d0aaa45f	9 years ago
Corey Farrell	93e8ed0154	Simplify and fix conditional in FD_SET. ... FD_SET contains a conditional statement to protect against buffer overruns. The statement was overly complicated and prevented use of the last array element of ast_fdset. We now just verify the fd is less than ast_FDMAX. Change-Id: I41895c0b497b052aef5bf49d75c817c48b326f40	9 years ago
Joshua Colp	a7c8d4cd6b	tests/test_sorcery_memory_cache_thrash: Improve termination process. ... When terminating the threads thrashing a sorcery memory cache each would be told to stop and then we would wait on them. During at least one thrashing test this was problematic due to the specific usage pattern in use. It would take some time for termination of the thread to occur. This would occur due to contention between the threads retrieving and the threads updating the cache. As the retrieving threads are given priority it may be some time before the updating threads are able to proceed. This change makes it so all threads are told to stop and then each are joined to ensure they stop. This way all the threads should stop at around the same time instead of waiting for one to stop, the next to stop, then the next, and so on. As a result of this the execution time for each thrash test is much closer to their expected value than previously seen as well. Change-Id: I04a53470b0ea4170b8819180b0bd7475f3642827	9 years ago
George Joseph	2451d4e455	res_pjsip: Fix infinite recursion when loading transports from realtime ... Attempting to load a transport from realtime was forcing asterisk into an infinite recursion loop. The first thing transport_apply did was to do a sorcery retrieve by id for an existing transport of the same name. For files, this just returns the previous object from res_sorcery_config's internal container, if any. For realtime, the res_sourcery_realtime driver looks in the database and finds the existing row but now it has to rehydrate it into a sorcery object which means calling... transport_apply. And so it goes. The main issue with loading from realtime (apart from the loop) was that transport stores structures and pointers directly in the ast_sip_transport structure instead of the separate ast_transport_state structure. This patch separates those items into the ast_sip_transport_state structure. The pattern is roughly the same as res_pjsip_outbound_registration. Although all current usages of ast_sip_transport and ast_sip_transport_state were modified to use the new ast_sip_get_transport_state API, the original items are left in ast_sip_transport and kept updated to maintain ABI compatability for third-party modules. They are marked as deprecated and noted that they're now in ast_sip_transport_state. ASTERISK-25606 #close Reported-by: Martin Moučka Change-Id: Ic7a836ea8e786e8def51fe3f8cce855ea54f5f19	9 years ago
Joshua Colp	78fa818c1b	Merge "chan_misdn: Fix a few issues causing compile errors" into 13	9 years ago
Richard Mudgett	6f978fbfe5	app_confbridge: Only use b_profile options from the conference. ... A user cannot set new bridge options after the conference is created by the first user. Attempting to do so is documented as undefined behavior. This patch ensures that the bridge profile options used are from the conference and not what a subsequent user may have tried to set. Change-Id: I1b6383eba654679e5739d5a8de98199cf074a266	9 years ago
George Joseph	ec8fd6714d	chan_misdn: Fix a few issues causing compile errors ... Change-Id: I54b48c24d7ca88ed80496fdfd142d08772a7ab98	9 years ago
Joshua Colp	1fca224533	Merge "pjsip/alembic: Add missing columns to system and registration" into 13	9 years ago
Joshua Colp	c9e1c203b5	Merge "app_confbridge.c: Replace inlined code with existing function." into 13	9 years ago
Joshua Colp	2177dbea50	Merge topic 'ASTERISK-20987' into 13 ... * changes: app_confbridge: Add ability to get the muted conference state. app_confbridge.c: Update CONFBRIDGE and CONFBRIDGE_INFO documentation. app_confbridge: Make non-admin users join a muted conference muted.	9 years ago
Mark Michelson	6a799cd78f	Check for OpenSSL defines before trying to use them. ... The SSL_OP_NO_TLSv1_1 and SSL_OP_NO_TLSv1_2 defines did not exist prior to OpenSSL version 1.0.1. A recent commit attempts to, by default, set these options, which can cause problems on systems with older OpenSSL installations. This commit adds a configure script check for those defines and will not attempt to make use of those if they do not exist. We will print a warning urging the user to upgrade their OpenSSL installation if those defines are not present. Change-Id: I6a2eb9a43fd0738b404d8f6f2cf4b5c22d9d752d	9 years ago
George Joseph	953d1cc11a	pjsip/alembic: Add missing columns to system and registration ... ps_systems needed disable_tcp_switch ps_registrations needed line and endpoint ASTERISK-25737 #close Change-Id: Iaf9c2d69e62243d9fa53104c28c5339c47d4ac19	9 years ago
Mark Michelson	c5e7d5c105	Merge "logging: Remove/fix some message annoyances" into 13	9 years ago
Joshua Colp	6cac364284	Merge "res_stasis_device_state: Fix refcounting error." into 13	9 years ago
Joshua Colp	014fc9ef65	Merge "res_xmpp: Does not connect in component mode" into 13	9 years ago
Mark Michelson	23829b3253	res_stasis_device_state: Fix refcounting error. ... Device state subscription lifetimes were governed by when the subscription was established and unsubscribed from. However, it is possible that at the time of unsubscription, there could be device state events still in flight. When those device state events occur, the device state callback could attempt to dereference a freed pointer. Crash. This change ensures that the lifetime of the device state subscription does not end until the underlying stasis subscription has confirmed that its final message has been sent. Change-Id: I25a0f1472894c1a562252fb7129671478e25e9b2	9 years ago
Sean Bright	4e8e6d3922	res_rtp_asterisk: Allow ICE host candidates to be overriden ... During ICE negotiation the IPs of the local interfaces are sent to the remote peer as host candidates. In many cases Asterisk is behind a static one-to-one NAT, so these host addresses will be internal IP addresses. To help in hiding the topology of the internal network, this patch adds the ability to override the host candidates by matching them against a user-defined list of replacements. Change-Id: I1c9541af97b83a4c690c8150d19bf7202c8bff1f	9 years ago
Kevin Harwell	f55f79edba	Merge "AST-2016-003 udptl.c: Fix uninitialized values." into 13	9 years ago
Kevin Harwell	4834d460d6	Merge "AST-2016-002 chan_sip.c: Fix retransmission timeout integer overflow." into 13	9 years ago
Richard Mudgett	c6b1b2b1c8	AST-2016-003 udptl.c: Fix uninitialized values. ... Sending UDPTL packets to Asterisk with the right amount of missing sequence numbers and enough redundant 0-length IFP packets, can make Asterisk crash. ASTERISK-25603 #close Reported by: Walter Doekes ASTERISK-25742 #close Reported by: Torrey Searle Change-Id: I97df8375041be986f3f266ac1946a538023a5255	9 years ago
Joshua Colp	f8acadde2c	AST-2016-001 http: Provide greater control of TLS and set modern defaults. ... This change exposes the configuration of various aspects of the TLS support and sets the default to the modern standards. The TLS cipher is now set to the best values according to the Mozilla OpSec team, different TLS versions can now be disabled, and the cipher order can be forced to be that of the server instead of the client. ASTERISK-24972 #close Change-Id: I0a10f2883f7559af5e48dee0901251dbf30d45b8	9 years ago
Richard Mudgett	3c81a052c8	AST-2016-002 chan_sip.c: Fix retransmission timeout integer overflow. ... Setting the sip.conf timert1 value to a value higher than 1245 can cause an integer overflow and result in large retransmit timeout times. These large timeout times hold system file descriptors hostage and can cause the system to run out of file descriptors. NOTE: The default sip.conf timert1 value is 500 which does not expose the vulnerability. * The overflow is now detected and the previous timeout time is calculated. ASTERISK-25397 #close Reported by: Alexander Traud Change-Id: Ia7231f2f415af1cbf90b923e001b9219cff46290	9 years ago
George Joseph	2a6ee8caeb	logging: Remove/fix some message annoyances ... test_dlinklists doesn't need to NOTICE everyone that every macro worked. res_phoneprov doesn't need to VERBOSE everyone that a phoneprov extension or provider was registered. res_odbc was missing a newline at the end of one message. Change-Id: I6c06361518ef3711821795e535acd439782a995e	9 years ago
Joshua Colp	73159cb45f	Merge "res_sorcery_realtime: Fix regex regression." into 13	9 years ago
Mark Michelson	389a94eda2	Merge "chan_sip.c: AMI & CLI notify methods get different values of asterisk's own ip." into 13	9 years ago
Mark Michelson	32fc784284	res_sorcery_realtime: Fix regex regression. ... A regression was introduced where searching for realtime PJSIP objects by regex by starting the regex with a leading "^" would cause no items to be returned. This was due to a change which attempted to drop the requirement for a leading "^" to be present due to how some CLI commands formulate their regexes. However, the change, rather than simply eliminating the requirement, caused any regexes that did begin with "^" to end up not returning the expected results. This change fixes the problem by inspecting the regex and formulating the realtime query differently depending on if it begins with "^". ASTERISK-25702 #close Reported by Nic Colledge Patches: realtime_retrieve_regex.patch submitted by Alexei Gradinari License #5691 Change-Id: I055df608a6e6a10732044fa737a9fe8dca602693	9 years ago
Karsten Wemheuer	0405c31756	res_xmpp: Does not connect in component mode ... The module res_xmpp does not accept usernames in the form used in component mode (XEP-0114). In component mode there is no @something in the name. In component mode the connection is now not dropped anymore. If the xmpp server sends out a "stream" tag before handshake is finished, the connection gets dropped in res_xmpp. Now this tag will be ignored and the connection will be established. After connecting there will be an exchange of presence states. This does not work as expected in component mode. The responsible function "xmpp_pak_presence" is left before the states get sent out. Sending presence states in component mode is now moved to the top of the function. ASTERISK-25735 #close Change-Id: I70e036f931c3124ebb2ad1e56f93ed35cfdd9d5c	9 years ago
Joshua Colp	f6551868be	Merge "res_odbc: Remove connection management" into 13	9 years ago
George Joseph	8804d0973c	build_system: Fix some warnings highlighted by clang ... Fix some warnings found with clang. Change-Id: I5195b6189b148c2ee3ed4a19d015a6d4ef3e77bd	9 years ago
George Joseph	109b0aff6b	res/Makefile: Fix bug in "clean" target for ari ... The "clean" target was attempting to clean res/ari from inside the res directory which doesn't remove anything. Removed the res/ prefix. Change-Id: Ib1a518d54efa81b9fd5a42742d43cc3767435bf6	9 years ago
George Joseph	a85fab7c44	pjsip/alembic: Fix definition of qualify_timeout ... A recent commit set qualify_timeout to Decimal which isn't supported. This path corrects it to Float. Change-Id: I038f5274ba8cb60f8518a5845ce448d49306aadf	9 years ago
StefanEng86	aa9348ab9a	chan_sip.c: AMI & CLI notify methods get different values of asterisk's own ip. ... When I ask asterisk to send a SIP NOTIFY message to a sip peer using either a) AMI action: SIPnotify or b) cli command: sip notify <cmd> <peer>, I expect asterisk to include the same value for its own ip in both cases a) and b), but it seems a) produces a contact header like Contact: <sip:asterisk@192.168.1.227:8060> whereas b) produces a contact header like <sip:asterisk@127.0.0.1:8060>. 0.0.0.0:8060 is my udpbindaddr in sip.conf My guess is that manager_sipnotify should call ast_sip_ouraddrfor(&p->sa, &p->ourip, p) the same way sip_cli_notify does, because after applying this patch, both cases a) and b) produce the contact header that I expect: <sip:asterisk@192.168.1.227:8060> Reported by: Stefan Engström Tested by: Stefan Engström Change-Id: I86af5e209db64aab82c25417de6c768fb645f476	9 years ago
Mark Michelson	65bd4fcc3f	res_odbc: Remove connection management ... Asterisk by default will create a single database connection and share it among all threads that attempt to access the database. In previous versions of Asterisk, this was tolerable, because the most used channel driver, chan_sip, mostly accessed the database from a single thread. With PJSIP, however, many threads may be attempting to perform database operations, and there is the potential for many more database accesses, meaning the concurrency is a horrible bottleneck if only one connection is shared. Asterisk has a connection pooling facility built into it, but the implementation has flaws. For one, there is a strict limit on the number of simultaneous connections that could be made to the database. Anything beyond the maximum would result in a failed operation. Attempting to predict what the maximum should be is nearly impossible even for someone intimately familiar with Asterisk's threading model. In addition, use of transactions in the dialplan can cause some severe bugs if connection pooling is enabled. This commit seeks to fix the concurrency problem by removing all connection management code from Asterisk and leaving that to the underlying unixODBC code instead. Now, Asterisk does not share a single connection, nor does it try to maintain a connection pool. Instead, all Asterisk ever does is request a connection from unixODBC and allow unixODBC to either allocate those connections or retrieve them from a pool. Doing this has a bit of a ripple effect. For one, since connections are not long-lived objects, several of the safeguards that previously existed have been removed. We don't have to worry about trying to use a connection that has gone stale. In every case, when we request a connection, it has just been made and we don't need to perform any sanity checks to be sure it's still active. Another major player affected by this change is transactions. Transactions and their respective connections were so tightly coupled that it was almost pornographic. This code change moves transaction-related code to its own file separate from the core ODBC functionality. This way, the core of ODBC does not even have to know that transactions exist. In making this large change, I had to look at a lot of code and understand it. When making this change, I discovered several places where the behavior is definitely not ideal, but it seemed outside the scope of this change to be fixing it. Instead, any place where I saw some sort of room for improvement has had a XXX comment added explaining what could be altered to improve it. Change-Id: I37a84def5ea4ddf93868ce8105f39de078297fbf	9 years ago
Joshua Colp	9711ad0de6	Merge "build_system: Prevent goals needing makeopts from running when it's missing" into 13	9 years ago
Mark Michelson	afdf8e5d5c	Merge "config: Allow options to register when documentation is unavailable." into 13	9 years ago
Richard Mudgett	2a9e623ff9	config_options.c: Fix warning message wording. ... Change-Id: I915ea437936320393afde0e7552cf0a980a6b2e4	9 years ago
Richard Mudgett	ed3c9c1512	app_confbridge.c: Replace inlined code with existing function. ... Change-Id: Ida5594e9f8d7c1fc18eeb733a11f8fb96326da51	9 years ago
Richard Mudgett	1d0abf86e7	app_confbridge: Add ability to get the muted conference state. ... * Added CONFBRIDGE_INFO(muted,) for querying the muted conference state. * Added Muted header to AMI ConfbridgeListRooms action response list events to indicate the muted conference state. * Added Muted column to CLI "confbridge list" output to indicate the muted conference state and made the locked column a yes/no value instead of a locked/unlocked value. ASTERISK-20987 Reported by: hristo Change-Id: I4076bd8ea1c23a3afd4f5833e9291b49a0c448b1	9 years ago
Richard Mudgett	f0d40afa69	app_confbridge.c: Update CONFBRIDGE and CONFBRIDGE_INFO documentation. ... Change-Id: Ic1f9e22ba1f2ff3b3f5cb017c5ddcd9bd48eccc7	9 years ago
Richard Mudgett	3e51e5c7fd	app_confbridge: Make non-admin users join a muted conference muted. ... ASTERISK-20987 #close Reported by: hristo Change-Id: Ic61a2b524ab3a4cfadf227fc6b3506527bc03f38	9 years ago
George Joseph	9da18af992	res_pjsip: Add res_pjproject dependency to UPGRADE.txt and samples ... Since res_pjsip now depends on res_pjproject, this is now mentioned in UPGRADE.txt and the basic-pbx modules.conf has been updated. Change-Id: I42826597d5e10f08e518208860c44c96e52f1b2d	9 years ago
George Joseph	aee8448bc2	build_system: Prevent goals needing makeopts from running when it's missing ... The Makefile only optionally includes makeopts so when goals like uninstall that dont depend on anything else are run after a distclean, rules like 'rm -f "$(DESTDIR)$(ASTMODDIR)/"*' get run as 'rm -f ""/*' which attempts to remove everything in the root directory. Although there's a rule defined for makeopts which prints a message and does an 'exit 1', since '-include makepopts' was specified (with the -), the exit was ignored letting the rest of the rules run. This patch makes makeopts required unless the goal has the string 'clean' in it. ASTERISK-25730 #close Reported-by: George Joseph Change-Id: I1bce59a7ea4f48e7a468e22b2abbb13c63417ac7	9 years ago
Joshua Colp	4cc784eb04	Merge "Stasis: Use custom structure when setting variables." into 13	9 years ago
Joshua Colp	f22074e5d9	config: Allow options to register when documentation is unavailable. ... The config options framework is strict in that configuration options must be documented unless XML documentation support is not available. In practice this is useful as it ensures documentation exists however in off-nominal cases this can cause strange problems. If it is expected that a config option has a non-zero or non-empty default value but the config option documentation is unavailable this reasonable expectation will not be met. This can cause obscure crashes and weirdness depending on how the code handles it. This change tweaks the behavior to ensure that the config option is still allowed to register, apply default values, and be set when devmode is not enabled. If devmode is enabled then the option can NOT be set. This also does not remove the initial documentation error message that is output on load when registering the configuration option. ASTERISK-25725 #close Change-Id: Iec42fca6b35f31326c33fcdc25473f6fd7bc8af8	9 years ago
Mark Michelson	4a3275abb9	Stasis: Use custom structure when setting variables. ... A recent change to queue channel variable setting to the Stasis control queue caused a regression. When setting channel variables, it is possible to give a NULL channel variable value in order to unset the variable (i.e. remove it from the channel variable list). The change introduced a call to ast_variable_new(), which is not tolerant of NULL channel variable values. This new change switches from using ast_variable to using a custom channel variable struct that is lighter weight and NULL value-tolerant. Change-Id: I784d7beaaa3c036ea936d103e7caf0bb1562162d	9 years ago
Matt Jordan	c9a0f4f8ff	Merge "res_pjsip_pubsub: Prevent crash from AMI command on freed subscription." into 13	9 years ago
Rusty Newton	b2c8a99f9e	sounds/Makefile: Incremented core and extra sounds versions to 1.5 ... Core and extra sounds 1.5 was recently released! The tarballs contain change descriptions however I figure more people will see this one so I'll try to be a bit detailed. Approximately 60 sounds were moved from Extra to Core for en, en_GB, fr and added for languages that didn't already have Extra sound sets (it,ja,ru). In addition all of the English and Russian sounds have been completely re-recorded. Sounds moved and added: activated,added,all-circuits-busy-now,astcc-followed-by-pound at-tone-time-exactly,call-forwarding,call-fwd-no-ans,call-fwd-on-busy ,call-fwd-unconditional,calling,call-waiting,cancelled, cannot-complete-as-dialed,check-number-dial-again,conf-full,de-activated ,disabled,do-not-disturb,enabled,enter-num-blacklist,entr-num-rmv-blklist ,extension,feature-not-avail-line,for,from-unknown-caller,goodbye,hello ,if-correct-press,im-sorry,info-about-last-call,is,is-in-use,is-set-to ,location,number,number-not-answering,num-was-successfully,one-moment-please ,please-try-again,pls-hold-while-try,pls-try-call-later,pm-invalid-option ,privacy-to-blacklist-last-caller,removed,simul-call-limit-reached ,something-terribly-wrong,sorry,sorry-youre-having-problems,speed-dial ,speed-dial-empty,telephone-number,time,to-call-this-number,to-extension ,to-listen-to-it,to-rerecord-it,unidentified-no-callback,with,you-entered ,your There were also a few random fixes here and there to file names for a few of the languages. ASTERISK-25068 #close Change-Id: I2b594344ec585d7dfd922b40c1af43b1508828b3	9 years ago