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Author	SHA1	Message	Date
Joshua Colp	c9e1c203b5	Merge "app_confbridge.c: Replace inlined code with existing function." into 13	10 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.	10 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	10 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	10 years ago
Mark Michelson	c5e7d5c105	Merge "logging: Remove/fix some message annoyances" into 13	10 years ago
Joshua Colp	6cac364284	Merge "res_stasis_device_state: Fix refcounting error." into 13	10 years ago
Joshua Colp	014fc9ef65	Merge "res_xmpp: Does not connect in component mode" into 13	10 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	10 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	10 years ago
Kevin Harwell	f55f79edba	Merge "AST-2016-003 udptl.c: Fix uninitialized values." into 13	10 years ago
Kevin Harwell	4834d460d6	Merge "AST-2016-002 chan_sip.c: Fix retransmission timeout integer overflow." into 13	10 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	10 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	10 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	10 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	10 years ago
Joshua Colp	73159cb45f	Merge "res_sorcery_realtime: Fix regex regression." into 13	10 years ago
Mark Michelson	389a94eda2	Merge "chan_sip.c: AMI & CLI notify methods get different values of asterisk's own ip." into 13	10 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	10 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	10 years ago
Joshua Colp	f6551868be	Merge "res_odbc: Remove connection management" into 13	10 years ago
George Joseph	8804d0973c	build_system: Fix some warnings highlighted by clang ... Fix some warnings found with clang. Change-Id: I5195b6189b148c2ee3ed4a19d015a6d4ef3e77bd	10 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	10 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	10 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	10 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	10 years ago
Joshua Colp	9711ad0de6	Merge "build_system: Prevent goals needing makeopts from running when it's missing" into 13	10 years ago
Mark Michelson	afdf8e5d5c	Merge "config: Allow options to register when documentation is unavailable." into 13	10 years ago
Richard Mudgett	2a9e623ff9	config_options.c: Fix warning message wording. ... Change-Id: I915ea437936320393afde0e7552cf0a980a6b2e4	10 years ago
Richard Mudgett	ed3c9c1512	app_confbridge.c: Replace inlined code with existing function. ... Change-Id: Ida5594e9f8d7c1fc18eeb733a11f8fb96326da51	10 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	10 years ago
Richard Mudgett	f0d40afa69	app_confbridge.c: Update CONFBRIDGE and CONFBRIDGE_INFO documentation. ... Change-Id: Ic1f9e22ba1f2ff3b3f5cb017c5ddcd9bd48eccc7	10 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	10 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	10 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	10 years ago
Joshua Colp	4cc784eb04	Merge "Stasis: Use custom structure when setting variables." into 13	10 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	10 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	10 years ago
Matt Jordan	c9a0f4f8ff	Merge "res_pjsip_pubsub: Prevent crash from AMI command on freed subscription." into 13	10 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	10 years ago
Mark Michelson	8261bda1bf	res_pjsip_pubsub: Prevent crash from AMI command on freed subscription. ... A test recently uncovered that running an ill-timed AMI command to show inbound subscriptions could cause a crash since Asterisk will try to operate on a freed subscription. The fix for this is to remove the subscription tree from the list of subscriptions at the time that we are sending our final NOTIFY request out. This way, as the subscription is in the process of dying, it is inaccessible from AMI. Change-Id: Ic0239003d8d73e04c47c12dd2a7e23867e5b5b23	10 years ago
Corey Farrell	a6823bb0c4	chan_sip: Fix buffer overrun in sip_sipredirect. ... sip_sipredirect uses sscanf to copy up to 256 characters to a stacked buffer of 256 characters. This patch reduces the copy to 255 characters to leave room for the string null terminator. ASTERISK-25722 #close Change-Id: Id6c3a629a609e94153287512c59aa1923e8a03ab	10 years ago
Joshua Colp	fcb6c1737d	Merge "Stasis: Use control queue to prevent crash." into 13	10 years ago
Mark Michelson	1003c2eb05	Stasis: Fix potential memory leak of control data. ... When queuing tasks onto the Stasis control queue, you can pass an arbitrary data pointer and a function to free that data. All ARI commands that use the Stasis control queue made the assumption that the destructor function would be called in all paths, whether the task was queued successfully or not. However, this was not correct. If a task was queued onto a control structure that was already completed, the allocated data would not be freed properly. This patch corrects this by making sure that all return paths call the data destructor. Change-Id: Ibf06522094f8e5c4cce652537dc5d7222b1c4fcb	10 years ago
Mark Michelson	eedd77fda0	Stasis: Use control queue to prevent crash. ... A crash occurred when attempting to set a channel variable on a channel that had already been hung up. This is because there is a small window between when a control is grabbed and when the channel variable is set that the channel can be hung up. The fix here is to queue the setting of the channel variable onto the control queue. This way, the manipulation of the channel happens in a thread where it is safe to be done. In this change, I also noticed that the setting of bridge roles on channels was being done outside of the control queue, so I also changed those operations to be done in the control queue. ASTERISK-25709 #close Reported by Mark Michelson Change-Id: I2a0a4d51bce6fba6f1d9954e40935e42f366ea78	10 years ago
Richard Mudgett	1c95b211a0	logger.c: Fix buffer overrun found by address sanitizer. ... The null terminator of the tail struct member was not being allocated when no logger.conf config file is installed. ASTERISK-25714 #close Reported by: Badalian Vyacheslav Change-Id: I45770fdd08af39506a3bc33ba279c4f16e047a30	10 years ago
Matt Jordan	a0d3932e2a	Merge "Build System: Add support for checking alembic branches." into 13	10 years ago
Matt Jordan	c3f4afe40c	Merge "res/res_pjsip/presence_xml.c: Add missing 2nd call presence state case." into 13	10 years ago
Corey Farrell	6ff945ab87	Build System: Add support for checking alembic branches. ... * Add 'check-alembic' target to root Makefile. * Create build_tools/make_check_alembic to do the actual checks. ASTERISK-25685 Change-Id: Ibb3cae7d1202ac23dc70b0f3b5801571ad46b004	10 years ago
Richard Mudgett	02035212de	res/res_pjsip/presence_xml.c: Add missing 2nd call presence state case. ... ASTERISK-25712 #close Reported by: Richard Mudgett Change-Id: I70634df24f8c6c3a2c66c45af61d021e4999253f	10 years ago
Diederik de Groot	c68c66c61f	main/asterisk.c: ast_el_read_char ... Make sure buf[res] is not accessed at res=-1 (buffer underrun). Address Sanitizer will complain about this quite loudly. ASTERISK-24801 #close Change-Id: Ifcd7f691310815a31756b76067c56fba299d3ae9	10 years ago