asterisk

Commit Graph

Author	SHA1	Message	Date
Matthew Jordan	1a934e007f	res/res_monitor: Reset in/out sample counts on Monitor start When repeatedly starting/stopping a Monitor on a channel, the accumulated in/out sample counts are never reset to 0. This can cause inadvertent jumps in the recordings, as the code in the channel core will determine incorrectly that a jump in the recorded file position should occur. Setting the sample counts to 0 simply reflects the initial state a Monitor should be in when it is started, as this is the initial count that would be on the channels at that time. ASTERISK-24573 #close Reported by: Nuno Borges patches: 24573.patch uploaded by Nuno Borges (License 6116) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@429031 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	c02beb1097	apps/app_meetme: Apply default values on initial load with no config file When the app_meetme module is loaded without its configuration file, the module settings aren't initialized. In particular, this impacts the use of logging realtime members. This patch guarantees that we always set the default module settings on initial load. Review: https://reviewboard.asterisk.org/r/4242/ ASTERISK-24572 #close Reported by: Nuno Borges patches: 24572.patch uploaded by Nuno Borges (License 6116) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@429027 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	4a054379de	apps/app_voicemail: Fix crash with IMAP when streams are opened simultaneously The UW IMAP library is instrinsically not thread-safe, and relies upon higher level applications to guarantee thread safety. For the most part, this is provided by the vms object, which provides locking for individual streams. Unfortunately, this is not sufficient for calls to mail_open which create the IMAP stream. mail_open can, on some systems, call into a UW IMAP specific function for determining the address of a system based on a hostname, ip_nametoaddr. In the ip6_unix implementation of this function, static variables are used to hold parsing buffers. This can cause a crash if multiple threads attempt to convert a hostname to an address at the same time. Locking on a single mail stream is not sufficient to prevent simultaneous access to these static variables. In the IMAP library, this function can be called from the mail_open and imap_status functions. As the imap_status function is not used by app_voicemail, locking on access to mail_open is sufficient to prevent any mangling of the buffers. Review: https://reviewboard.asterisk.org/r/4188/ ASTERISK-24516 #close Reported by: David Duncan Ross Palmer Tested by: David Duncan Ross Palmer patches: ASTERISK-24516.diff uploaded by David Duncan Ross Palmer (License 6660) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@428863 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	a6ed286a9a	pbx/pbx_loopback: Speed up switches by avoiding unneeded lookups This patch makes a small rearrangement to only do dialplan lookups during loopback switches if the pattern matches. Prior to this patch, the dialplan lookups were always performed, even when the result would be discarded. Dialplan lookups can be very costly if remote switches - like DUNDi - are present. In those cases extension matching is sped up considerably, making the issue of lost digits more manageable. As collateral damage, 6 trailing spaces were killed. Review: https://reviewboard.asterisk.org/r/4211 ASTERISK-24577 #close Reported by: Birger Harzenetter patches: ast-loopback.patch uploaded by Birger Harzenetter (License 5870) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@428787 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Joshua Colp	d2d6a36bc8	app_record: Fix bug where using the 'k' option and hanging up would trim 1/4 of a second of the recording. The Record dialplan function trims 1/4 of a second from the end of recordings in case they are terminated because of DTMF. When hanging up, however, you don't want this to happen. This change makes it so on hangup this does not occur. ASTERISK-24530 #close Reported by: Ben Smithurst patches: app_record_v2.diff submitted by Ben Smithurst (license 6529) Review: https://reviewboard.asterisk.org/r/4201/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@428653 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	14d534a73a	manager: Fix could not extend string messages. When shutting down Asterisk that has an active AMI connection, you get several "failed to extend from %d to %d" messages because use of the EVENT_FLAG_SHUTDOWN attempts to add all AMI permission strings to the event. * Created MAX_AUTH_PERM_STRING to use when creating stack based struct ast_str variables used with the authority_to_str() and user_authority_to_str() functions instead of a variety of magic numbers that could be too small. * Added a special check for EVENT_FLAG_SHUTDOWN to authority_to_str() so it will not attempt to add all permission level strings. Review: https://reviewboard.asterisk.org/r/4200/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@428570 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Mark Michelson	4035e91fb6	Fix error with mixed address family ACLs. Prior to this commit, the address family of the first item in an ACL was used to compare all incoming traffic. This could lead to traffic of other IP address families bypassing ACLs. ASTERISK-24469 #close Reported by Matt Jordan Patches: ASTERISK-24469-11.diff uploaded by Matt Jordan (License #6283) AST-2014-012 ........ Merged revisions 428402 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@428417 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kevin Harwell	31d0fc0ecc	AST-2014-018 - func_db: DB Dialplan function permission escalation via AMI. The DB dialplan function when executed from an external protocol (for instance AMI), could result in a privilege escalation. Asterisk now inhibits the DB function from being executed from an external interface if the live_dangerously option is set to no. ASTERISK-24534 Reported by: Gareth Palmer patches: submitted by Gareth Palmer (license 5169) ........ Merged revisions 428331 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@428363 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kevin Harwell	060ced4b54	AST-2014-017 - app_confbridge: permission escalation/ class authorization. Confbridge dialplan function permission escalation via AMI and inappropriate class authorization on the ConfbridgeStartRecord action. The CONFBRIDGE dialplan function when executed from an external protocol (for instance AMI), could result in a privilege escalation. Also, the AMI action “ConfbridgeStartRecord” could also be used to execute arbitrary system commands without first checking for system access. Asterisk now inhibits the CONFBRIDGE function from being executed from an external interface if the live_dangerously option is set to no. Also, the “ConfbridgeStartRecord” AMI action is now only allowed to execute under a user with system level access. ASTERISK-24490 Reported by: Gareth Palmer git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@428332 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Joshua Colp	ce20eaa076	AST-2014-014: Fix race condition where channels may get stuck in ConfBridge under load. Under load it was possible for the bridging API, and thus ConfBridge, to get channels that may have hung up stuck in it. This is because handling of state transitions for a bridged channel within a bridge was not protected and simply set the new state without regard to the existing state. If the existing state had been hung up this would get overwritten. This change adds locking to protect changing of the state and also takes into consideration the existing state. ASTERISK-24440 #close Reported by: Ben Klang Review: https://reviewboard.asterisk.org/r/4173/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@428299 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	094eeade6e	ast_str: Fix improper member access to struct ast_str members. Accessing members of struct ast_str outside of the string manipulation API routines is invalid since struct ast_str is supposed to be treated as opaque. Review: https://reviewboard.asterisk.org/r/4194/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@428244 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Corey Farrell	b3e0d05aad	chan_sip: Fix theoretical leak of p->refer. If transmit_refer is called when p->refer is already allocated, it leaks the previous allocation. Updated code to always free previous allocation during a new allocation. Also instead of checking if we have a previous allocation, always create a clean record. ASTERISK-15242 #close Reported by: David Woolley Review: https://reviewboard.asterisk.org/r/4160/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@428117 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	f20ddb1285	apps/app_confbridge: Ensure 'normal' users hear message when last marked leaves When r428077 was made for ASTERISK-24522, it failed to take into account users who are neither wait_marked nor end_marked. These users are also supposed to hear the 'leader has left the conference' message. Granted, this behaviour is a bit odd; however, that is how it used to work... and behaviour changes are not good. This patch ensures that if there are any 'normal' users present when the last marked user leaves the conference, the message will still be played to them. Note that this regression was caught by the Asterisk Test Suite's confbridge_nominal test, which has a quirky combination of users. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@428113 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	10d242b728	app_confbridge: Don't play leader leaving prompt if no one will hear it Consider the following: - A marked user in a conference - One or more end_marked only users in the conference When the marked users leaves, we will be in the conf_state_multi_marked state. This currently will traverse the users, kicking out any who have the end_marked flags. When they are kicked, a full ast_bridge_remove is immediately called on the channels. At this time, we also unilaterally set the need_prompt flag. When the need_prompt flag is set, we then playback a sound to the bridge informing everyone that the leader has left; however, no one is left in the bridge. This causes some odd behaviour for the end_marked users - they are stuck waiting for the bridge to be unlocked. This results in them waiting for 5 or 6 seconds of dead air before hearing that they've been kicked. Unfortunately, we do have to keep the bridge locked while we're playing back the 'leader-has-left' prompt. If there are any wait_marked users in the conference, this behaviour can't be easily changed - but we do make the case of the end_marked users better with this patch. Review: https://reviewboard.asterisk.org/r/4184/ ASTERISK-24522 #close Reported by: Matt Jordan git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@428077 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	2f27faa037	cel/cel_odbc: Provide microsecond precision in 'eventtime' column when possible This patch adds microsecond precision when inserting a CEL record into a table with an "eventtime" column of type timestamp, instead of second precision. The documentation (configs/cel_odbc.conf.sample) was already saying that the eventtime column included microseconds precision, but that was not the case. Also, without this patch, if you had a table with an "eventtime" column of type varchar, you had millisecond precision. With this patch, you also get microsecond precision in this case. Review: https://reviewboard.asterisk.org/r/3980 ASTERISK-24283 #close Reported by: Etienne Lessard patches: cel_odbc_time_precision.patch uploaded by Etienne Lessard (License 6394) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@427952 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Scott Griepentrog	d0495f4139	stun: correct attribute string padding to match rfc When sending the USERNAME attribute in an RTP STUN response, the implementation in append_attr_string passed the actual length, instead of padding it up to a multiple of four bytes as required by the RFC 3489. This change adds separate variables for the string and padded attributed lengths, and performs padding correctly. Reported by: Thomas Arimont Review: https://reviewboard.asterisk.org/r/4139/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@427874 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Joshua Colp	093db340b1	app_confbridge: Play "leader has left" sound even when musiconhold is enabled. Currently if the leader of a conference bridge leaves any participant that has musiconhold enabled will not hear the "leader has left" sound. This is because musiconhold is started and THEN the sound is played. This change makes it so that the sound is played and THEN musiconhold is started. This provides a better experience for users as they may not have known previously why they went back to musiconhold. Review: https://reviewboard.asterisk.org/r/4177/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@427844 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Joshua Colp	329d09913e	pbx: Fix off-nominal case where a freed extension may still be used. If during the operation of adding an extension a priority is added but fails it is possible for the extension to be freed but still exist in the PBX core. If this occurs subsequent lookups may try to access the extension and end up in freed memory. This change removes the extension from the PBX core when the priority addition fails and then frees the extension. ASTERISK-24444 #close Reported by: Leandro Dardini Review: https://reviewboard.asterisk.org/r/4162/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@427709 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Corey Farrell	e55b96ad57	Fix compiler error when using ./configure --enable-dev-mode --enable-coverage When DONT_OPTIMIZE is enabled with dev-mode, it causes a shadow compilation to be done with output to /dev/null. This can cause errors with coverage when GCC attempts to write to /dev/null.gcno. This change disables coverage for the shadow compilation. ASTERISK-24502 #close Reported by: Corey Farrell Review: https://reviewboard.asterisk.org/r/4151/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@427682 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Corey Farrell	0e37018d93	manager: Fix HTTP connection reference leaks. Fix reference leak that happens if (session && !blastaway). ASTERISK-24505 #close Reported by: Corey Farrell Review: https://reviewboard.asterisk.org/r/4153/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@427641 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	cc46ce38bc	configs/features.conf: Add documentation noting potential chan_agent conflict In chan_agent, a '' is used by default to terminate a bridge with a caller. This can lead to all sorts of problems if '' is used by a feature in features.conf, as the chan_agent disconnect '*' may be detected first. This patch adds a documentation snippet to features.conf so that users who attempt to use features with agents know of the potential conflict. ASTERISK-20402 #close Reported by: Matt Riddell patches: features.conf.diff uploaded by Matt Riddell (License 5023) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@427617 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	938f6d3bc0	channels/chan_mgcp: Fix regression which causes gateways to be skipped In r227276, a while loop was turned into a for loop. Unfortunately, a portion of the while loop was left in the code such that, when a static gateway is encountered in the list of MGCP gateways, the next gateway would be skipped. At best, we would simply flip past a gateway; at worst, this could lead to a crash. ASTERISK-24500 #close Reported by: Xavier Hienne patches: chan_mgcp.patch uploaded by Xavier Hienne (License 6657) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@427613 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	80439dac04	addons/chan_mobile: Increase buffer size of UCS2 encoded SMS messages When UCS2 character encoding is used, one symbol in national language can be expanded to 4 bytes. The current buffer used for receiving message in do_monitor_phone is 256 bytes, which is not large enough for incoming messages. For example: * AT+CMGR phone response prefix '+CMGR: "REC UNREAD","+7*********",,"14/10/29,13:31:39+12"\r\n' - 60 bytes SMS body with UCS2 encoding (max) - 280 bytes * AT+CMGR phone response suffix '\r\n\r\nOK\r\n' - 8 bytes * Terminating null character - 1 byte This results in a needed buffer size of 349 bytes. Hence, this patch opts for a 350 byte buffer. ASTERISK-24468 #close Reported by: Dmitriy Bubnov patches: chan_mobile-1_8.diff uploaded by Dmitriy Bubnov (License 6651) chan_mobile-trunk.diff uploaded by Dmitry Bubnov (License 6651) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@427607 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Corey Farrell	ed7dabef46	chan_console: Fix reference leaks to pvt. Fix a bunch of calls to get_active_pvt where the reference is never released. ASTERISK-24504 #close Reported by: Corey Farrell Review: https://reviewboard.asterisk.org/r/4152/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@427554 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Corey Farrell	8745e12323	main/file.c: fix possible extra ast_module_unref to format modules. fn_wrapper only adds a reference to the format's module if the file was able to be opened. If not this causes an unmatched ast_module_unref in filestream_destructor. Move ast_module_ref to get_stream. ASTERISK-24492 #close Reported by: Corey Farrell Review: https://reviewboard.asterisk.org/r/4149/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@427464 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Corey Farrell	e9f3480121	Fix unintential memory retention in stringfields. * Fix missing / unreachable calls to __ast_string_field_release_active. * Reset pool->used to zero when the current pool->active reaches zero. ASTERISK-24307 #close Reported by: Etienne Lessard Tested by: ibercom, Etienne Lessard Review: https://reviewboard.asterisk.org/r/4114/ ........ Merged revisions 427380 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@427381 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
George Joseph	48f329bfe8	test_strings: Remove string tests that exercise asserts. Since unit tests are run with DO_CRASH, those tests were causing the test to fail. Tested-by: George Joseph git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@427354 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
George Joseph	03fcc1ad72	config: Make text_file_save and 'dialplan save' escape semicolons in values. When a config file is read, an unescaped semicolon signals comments which are stripped from the value before it's stored. Escaped semicolons are then unescaped and become part of the value. Both of these behaviors are normal and expected. When the config is serialized either by 'dialplan save' or AMI/UpdateConfig however, the now unescaped semicolons are written as-is. If you actually reload the file just saved, the unescaped semicolons are now treated as start of comments. Since true comments are stripped on read, any semicolons in ast_variable.value must have been escaped originally. This patch re-escapes semicolons in ast_variable.values before they're written to file either by 'dialplan save' or config/ast_config_text_file_save which is called by AMI/UpdateConfig. I also fixed a few pre-existing formatting issues nearby in pbx_config.c Tested-by: George Joseph ASTERISK-20127 #close Review: https://reviewboard.asterisk.org/r/4132/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@427328 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Corey Farrell	47ee18acc1	Fix compile error caused by review 4138 There is no procedure called ast_closeframe, fix code to use ast_closestream. Reported By: Matt Jordan git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@427087 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Corey Farrell	9dc2f92921	Fix ast_writestream leaks Fix cleanup in __ast_play_and_record where others[x] may be leaked. This was caught where prepend != NULL && outmsg != NULL, once realfile[x] == NULL any further others[x] would be leaked. A cleanup block was also added for prepend != NULL && outmsg == NULL. 11+: Fix leak of ast_writestream recording_fs in app_voicemail:leave_voicemail. ASTERISK-24476 #close Reported by: Corey Farrell Review: https://reviewboard.asterisk.org/r/4138/ ........ Merged revisions 427023 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@427024 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Corey Farrell	0b55748232	func_jitterbuffer: fix frame leaks. Fix code paths where it is possible for frames to leak. Fix uninitialized variable in jb_get_fixed and jb_get_adaptive. ASTERISK-22409 #related Reported by: Corey Farrell Review: https://reviewboard.asterisk.org/r/4128/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@427019 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Tzafrir Cohen	72bf6d5052	Fix syntax from commit r426927 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@426931 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Tzafrir Cohen	4a313981f1	install init.d files on GNU/kFreeBSD Review: https://reviewboard.asterisk.org/r/4118/ ........ Merged revisions 426926 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@426927 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	9da14f75e0	channels/sip/reqresp_parser: Fix unit tests for r426594 When r426594 was made, it did not take into account a unit test that verified that the function properly populated the unsupported buffer. The function would previously memset the buffer if it detected it had any contents; since this function can now be called iteratively on successive headers, the unit tests would now fail. This patch updates the unit tests to reset the buffer themselves between successive calls, and updates the documentation of the function to note that this is now required. ........ Merged revisions 426858 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@426860 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Corey Farrell	2716e17f51	REF_DEBUG: Install refcounter.py to $(ASTDATADIR)/scripts This change ensures refcounter.py is installed to a place where it can be found by the Asterisk testsuite if REF_DEBUG is enabled. ASTERISK-24432 #close Reported by: Corey Farrell Review: https://reviewboard.asterisk.org/r/4094/ ........ Merged revisions 426830 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@426831 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Corey Farrell	a3ec9d8f1b	app_queue: fix a couple leaks to struct call_queue in set_member_value set_member_value has a couple leaks to references in the variable q found through testsuite tests/queues/set_penalty. Also remove the REF_DEBUG_ONLY_QUEUES compiler declaration, this is no longer possible with the updated REF_DEBUG code. ASTERISK-24466 #close Reported by: Corey Farrell Review: https://reviewboard.asterisk.org/r/4125/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@426805 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Walter Doekes	15f16e3187	app_voicemail: Fix unchecked bounds of myArray in IMAP_STORAGE. In update_messages_by_imapuser(), messages were appended to a finite array which resulted in a crash when an IMAP mailbox contained more than 256 entries. This memory is now dynamically increased as needed. Observe that this patch adds a bunch of XXX's to questionable code. See the review (url below) for more information. ASTERISK-24190 #close Reported by: Nick Adams Tested by: Nick Adams Review: https://reviewboard.asterisk.org/r/4126/ ........ Merged revisions 426691 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@426692 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Igor Goncharovskiy	865aa54aac	Add additional checks for NULL pointers to fix several crashes reported.	12 years ago
Matthew Jordan	cfa7763f85	channels/chan_sip: Add improved support for 4xx error codes This patch adds support for 414, 493, 479, and a stray 400 response in REGISTER response handling. This helps interoperability in a number of scenarios. Review: https://reviewboard.asterisk.org/r/3437 patches: rb3437.patch uploaded by oej (License 5267) ........ Merged revisions 426599 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@426600 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	bcd3f49994	channels/chan_sip: Support mutltiple Supported and Required headers A SIP request may contain multiple Supported: and Required: headers. Currently, chan_sip only parses the first Supported/Required header it finds. This patch adds support for multiple Supported/Required headers for INVITE requests. Review: https://reviewboard.asterisk.org/r/2478 ASTERISK-21721 #close Reported by: Olle Johansson patches: rb2478.patch uploaded by oej (License 5267) ........ Merged revisions 426594 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@426595 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Corey Farrell	5b69b095d1	res_fax: Resolve T38 gateway frame leak. When frames are translated by a fax gateway they need to be freed. The existing call to ast_frfree was unreachable. This change reorganizes fax_gateway_framehook to ensure that ast_frfree is called when needed. ASTERISK-24457 #close Reported by: Corey Farrell Review: https://reviewboard.asterisk.org/r/4115/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@426527 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Malcolm Davenport	a70300f7ac	ASTERISK-23512, correct inaccurate comment in manager.conf.sample git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@426456 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	0a2400ab6e	main/manager: Fix typo in AMI event documentation of "OriginateResponse" The parameter name is "Response", not "Resonse". ASTERISK-24430 #close Reported by: Dafi Ni git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@426366 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Malcolm Davenport	8c40b67c50	ASTERISK-24323, fix bug in documentation of AGI STREAM FILE CONTROL git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@426359 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Malcolm Davenport	ab4b71abe0	ASTERISK-24419, fix incorrect syntax for setting language in extensions.conf.sample git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@426291 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Corey Farrell	37d9bfdd05	app_queue: Cleanup ao2_iterator Clean ao2_iterator, resolving reference leak to queue members. ASTERISK-24454 #close Reported by: Corey Farrell Review: https://reviewboard.asterisk.org/r/4111/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@426255 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	7ba7792027	res/res_http_websocket: Fix minor nits found by wdoekes on r409681 When Moises committed the fixes for WSS (which was a great patch), wdoekes had a few style nits that were on the review that got missed. This patch resolves what I think were all of the ones that were still on the review. Thanks to both moy for the patch, and wdoekes for the reviews. Review: https://reviewboard.asterisk.org/r/3248/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@426209 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	f1d0cb5daa	res/res_srtp: Fix include issue for libsrtp 1.5.0 In libsrtp 1.5.0, crypto_get_random is no longer resolved simply by including srtp.h. Now, one must include crypto_kernel.h as well. As it turns out, this header file has been provided by the library since 2006, so this is a relatively benign change. ASTERISK-24436 #close Reported by: Patrick Laimbock ........ Merged revisions 426140 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@426141 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	8c96fa291c	AST-2014-011: Fix POODLE security issues There are two aspects to the vulnerability: (1) res_jabber/res_xmpp use SSLv3 only. This patch updates the module to use TLSv1+. At this time, it does not refactor res_jabber/res_xmpp to use the TCP/TLS core, which should be done as an improvement at a latter date. (2) The TCP/TLS core, when tlsclientmethod/sslclientmethod is left unspecified, will default to the OpenSSL SSLv23_method. This method allows for all encryption methods, including SSLv2/SSLv3. A MITM can exploit this by forcing a fallback to SSLv3, which leaves the server vulnerable to POODLE. This patch adds WARNINGS if a user uses SSLv2/SSLv3 in their configuration, and explicitly disables SSLv2/SSLv3 if using SSLv23_method. For TLS clients, Asterisk will default to TLSv1+ and WARN if SSLv2 or SSLv3 is explicitly chosen. For TLS servers, Asterisk will no longer support SSLv2 or SSLv3. Much thanks to abelbeck for reporting the vulnerability and providing a patch for the res_jabber/res_xmpp modules. Review: https://reviewboard.asterisk.org/r/4096/ ASTERISK-24425 #close Reported by: abelbeck Tested by: abelbeck, opsmonitor, gtjoseph patches: asterisk-1.8-jabber-tls.patch uploaded by abelbeck (License 5903) asterisk-11-jabber-xmpp-tls.patch uploaded by abelbeck (License 5903) AST-2014-011-1.8.diff uploaded by mjordan (License 6283) AST-2014-011-11.diff uploaded by mjordan (License 6283) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@425986 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	ebb3d8db82	channels/chan_sip: Respect outboundproxy setting when sending qualify requests The outboundproxy setting is currently ignored when sending OPTIONS requests as a result of the qualify setting. This means that if an Asterisk server is unable to send the packet directly to a peer, it is unable to qualify any non-inbound registered peer (e.g. a peer SIP Trunk). This patch grabs the outboundproxy information for a peer when a qualify attempt is being constructed and, if it finds the information, uses it when sending the OPTIONS request. Review: https://reviewboard.asterisk.org/r/3948 ASTERISK-24063 #close Reported by: Damian Ivereigh patches: outboundproxy-dai.patch uploaded by Damian Ivereigh (License 6632) ........ Merged revisions 425818 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@425819 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago

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