asterisk

Commit Graph

Author	SHA1	Message	Date
Kinsey Moore	457d5c39dc	Use srtp_shutdown when available This allows the SRTP library to be shut down properly when the functionality is offered by libsrtp. Review: https://reviewboard.asterisk.org/r/2538/ (closes issue ASTERISK-21719) ........ Merged revisions 388768 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@388769 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	a58b7639dd	Make ao2 global objects not always use the debug version of the ao2_ref() calls. The debug versions of ao2_ref() should only be used if REF_DEBUG is enabled so nothing is written to /tmp/refs unexpectedly. (closes issue ASTERISK-21785) Reported by: abelbeck Patches: jira_asterisk_21785_v11.patch (license #5621) patch uploaded by rmudgett Tested by: abelbeck git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@388700 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Sean Bright	f03ff24bac	Use the proper lower bound when doing saturation arithmetic. 16 bit signed integers have a range of [-32768, 32768). The existing code was using the interval (-32768, 32768) instead. This patch fixes that. Review: https://reviewboard.asterisk.org/r/2479/ ........ Merged revisions 386929 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@386930 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Jason Parker	d8216bd9ee	Add dependency on libuuid, for res_rtp_asterisk pjproject is what actually requires libuuid. (closes issue ASTERISK-21125) reported by Private Name (Ed. note: Really? Private Name? I am rolling my eyes so hard right now.) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@385356 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Joshua Colp	2d95e2884e	Regenerate the configure script. The one in the tree was not working for me at all. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@382151 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Kevin Harwell	1595101bb6	Stopped spamming of debug messages during attended transfer. While autoservice is running and servicing a channel the callid is being stored and removed in the thread's local storage for each iteration of the thread loop. If debug was set to a sufficient level the log file would be spammed with callid thread local storage debug messages. Added a new function that checks to see if the callid to be stored is different than what is already contained (if anything). If it is different then store/replace and log, otherwise just leave as is. Also made it so all logging of debug messages pertaining to the callid thread storage outputs only when TEST_FRAMEWORK is defined. (issue ASTERISK-21014) (closes issue ASTERISK-21014) Report by: Rusty Newton Review: https://reviewboard.asterisk.org/r/2324/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@381554 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Richard Mudgett	2d1b7d4732	Make CHECK_BLOCKING() debug message more useful. Change the displayed pthread value to hex format so it can be easily matched with CLI core show threads or gdb. ........ Merged revisions 380611 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@380612 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	49959b511f	Support building Asterisk for Raspberry Pi/Raspbian with hard-float support Building Asterisk on Raspbian with hard-float support fails as it uses the string 'linux-gnueabihf' for host os, as opposed to 'linux-gnueabi'. This patch modifies the configure script for Asterisk such that it will match on any string beginning with 'linux-gnueabi', as opposed to requiring an explicit match. (closes issue ASTERISK-21006) Reported by: Christian Hesse Tested by: Christian Hesse patches: linux-gnueabihf.patch uploaded by Christian Hesse (license 6459) linux-gnueabihf-autoconf.patch uploaded by Christian Hesse (license 6459) ........ Merged revisions 380520 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@380521 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Walter Doekes	bb450bafd7	Add builtin roundf() for systems lacking it. (closes issue ASTERISK-16854) Review: https://reviewboard.asterisk.org/r/2276 Reported-by: Ovidiu Sas ........ Merged revisions 379547 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@379548 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
David M. Lee	af6b4fed4f	Specify the -rpath linker flag when prefix != /usr. This allows Asterisk to start without having to specify the LD_LIBRARY_PATH. This can be disabled by passing --disable-rpath to configure. (closes issue ASTERISK-20407) Reported by: David M. Lee Review: https://reviewboard.asterisk.org/r/2132/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@379475 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
David M. Lee	12c51024c3	Fix XML encoding of 'identity display' in NOTIFY messages. XML encoding in chan_sip is accomplished by naively building the XML directly from strings. While this usually works, it fails to take into account escaping the reserved characters in XML. This patch adds an 'ast_xml_escape' function, which works similarly to 'ast_uri_encode'. This is used to properly escape the local_display attribute in XML formatted NOTIFY messages. Several things to note: * The Right Thing(TM) to do would probably be to replace the ast_build_string stuff with building an ast_xml_doc. That's a much bigger change, and out of scope for the original ticket, so I refrained myself. * It is with great sadness that I wrote my own ast_xml_escape function. There's one in libxml2, but it's knee-deep in libxml2-ness, and not easily used to one-off escape a string. * I only escaped the string we know is causing problems (local_display). At least some of the other strings are URI-encoded, which should be XML safe. Rather than figuring out what's safe and escaping what's not, it would be much cleaner to simply build an ast_xml_doc for the messages and let the XML library do the XML escaping. Like I said, that's out of scope. (closes issue ABE-2902) Reported by: Guenther Kelleter Tested by: Guenther Kelleter Review: http://reviewboard.digium.internal/r/365/ ........ Merged revision 378919 from https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier ........ Merged revisions 378933 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@378934 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
David M. Lee	1f6610acec	Move declaration of ast_regex_string_to_regex_pattern futher down strings.h. The prior location is before the declaration of struct ast_str, which causes compiler warnings. (closes issue ASTERISK-20852) Reported by: Pavel Troller Patches: strings.diff uploaded by Pavel Troller (license 6302) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@378747 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
David M. Lee	8d7224d5a8	Replace errant tabs with spaces in causes.h. (closes issue ASTERISK-20826) Reported by: snuffy Patches: notabs.dif uploaded by snuffy (license 5024) ........ Merged revisions 378733 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@378734 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	b1366d7acf	Fix AMI redirect action with two channels failing to redirect both channels. The AMI redirect action can fail to redirect two channels that are bridged together. There is a race between the AMI thread redirecting the two channels and the bridge thread noticing that a channel is hungup from the redirects. * Made the bridge wait for both channels to be redirected before exiting. * Made the AMI redirect check that all required headers are present before proceeding with the redirection. * Made the AMI redirect require that any supplied ExtraChannel exist before proceeding. Previously the code fell back to a single channel redirect operation. (closes issue ASTERISK-18975) Reported by: Ben Klang (closes issue ASTERISK-19948) Reported by: Brent Dalgleish Patches: jira_asterisk_19948_v11.patch (license #5621) patch uploaded by rmudgett Tested by: rmudgett, Thomas Sevestre, Deepak Lohani, Kayode Review: https://reviewboard.asterisk.org/r/2243/ ........ Merged revisions 378356 from http://svn.asterisk.org/svn/asterisk/branches/1.8 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@378358 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	eda6664de0	Prevent exhaustion of system resources through exploitation of event cache Asterisk maintains an internal cache for devices in the event subsystem. The device state cache holds the state of each device known to Asterisk, such that consumers of device state information can query for the last known state for a particular device, even if it is not part of an active call. The concept of a device in Asterisk can include entities that do not have a physical representation. One way that this occurred was when anonymous calls are allowed in Asterisk. A device was automatically created and stored in the cache for each anonymous call that occurred; this was possible in the SIP and IAX2 channel drivers and through channel drivers that utilized the res_jabber/res_xmpp resource modules (Gtalk, Jingle, and Motif). These devices are never removed from the system, allowing anonymous calls to potentially exhaust a system's resources. This patch changes the event cache subsystem and device state management to no longer cache devices that are not associated with a physical entity. (issue ASTERISK-20175) Reported by: Russell Bryant, Leif Madsen, Joshua Colp Tested by: kmoore patches: event-cachability-3.diff uploaded by jcolp (license 5000) ........ Merged revisions 378303 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 378320 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@378321 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	797e403982	confbridge: Fix MOH on simultaneous user entry to a new conference. When two users entered a new conference simultaneously, one of the callers hears MOH. This happened if two unmarked users entered simultaneously and also if a waitmarked and a marked user entered simultaneously. * Created a confbridge internal MOH API to eliminate the inlined MOH handling code. Note that the conference mixing bridge needs to be locked when actually starting/stopping MOH because there is a small window between the conference join unsuspend MOH and actually joining the mixing bridge. * Created the concept of suspended MOH so it can be interrupted while conference join announcements to the user and DTMF features can operate. * Suspend any MOH until the user is about to actually join the mixing bridge of the conference. This way any pre-join file playback does not need to worry about MOH. * Made post-join actions only play deferred entry announcement files. Changing the user/conference state during that time is not protected or controlled by the state machine. (closes issue ASTERISK-20606) Reported by: Eugenia Belova Tested by: rmudgett Review: https://reviewboard.asterisk.org/r/2232/ ........ Merged revisions 377992 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@377993 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	869c2b30c9	Cleanup core main on exit. * Cleanup time zones on exit. * Make exit clean/unclean report consistent for AMI and CLI in really_quit(). (issue ASTERISK-20649) Reported by: Corey Farrell Patches: core-cleanup-1_8-10.patch (license #5909) patch uploaded by Corey Farrell core-cleanup-11-trunk.patch (license #5909) patch uploaded by Corey Farrell Modified ........ Merged revisions 377135 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 377136 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@377137 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	224dd2f60a	Made AST_LIST_REMOVE() simpler and use better names. * Update doxygen of AST_LIST_REMOVE(). ........ Merged revisions 376627 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 376628 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@376629 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	c4f013c5c8	Re-initialize logmsgs mutex upon logger initialization to prevent lock errors Similar to the patch that moved the fork earlier in the startup sequence to prevent mutex errors in the recursive mutex surrounding the read/write thread registration lock, this patch re-initializes the logmsgs mutex. Part of the start up sequence before forking the process into the background includes reading asterisk.conf; this has to occur prior to the call to daemon in order to read startup parameters. When reading in a conf file, log statements can be generated. Since this can't be avoided, the mutex instead is re-initialized to ensure a reset of any thread tracking information. This patch also includes some additional debugging to catch errors when locking or unlocking the recursive mutex that surrounds locks when the DEBUG_THREADS build option is enabled. DO_CRASH or THREAD_CRASH will cause an abort() if a mutex error is detected. (issue ASTERISK-19463) Reported by: mjordan Tesetd by: mjordan ........ Merged revisions 376586 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 376587 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@376588 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
David M. Lee	bbf3526314	Migrate hashtest/hashtest2 to be unit tests. Both hashtest and hashtest2 are manual testing apps that thrash hash tables (hashtab and ao2 containers, respectively), by spinning up several threads that randomly insert, delete, lookup and iterate over the hash table. If the app doesn't crash, the hash table probably passes the test. Those utils are not a part of the typical Asterisk build, so they do not usually get compiled. This all makes them less that useful. This patch removes those manual test programs and replaces them with Asterisk unit test modules (test_{hashtab,astobj2}_thrash.so). It also attempts to make the tests more deterministic. * Rather than spinning up some number of threads that operate on the hash table randomly, spin up four threads that concurrenly add, remove, lookup and iterate over the hash table. * Each thread checks the state of the hash table both during and after execution, and indicates a test failure if things are not as expected. * Each thread times out after 60 seconds to prevent deadlocking the unit test run. (closes issue ASTERISK-20505) Reported by: Matt Jordan Review: https://reviewboard.asterisk.org/r/2189/ ........ Merged revisions 376306 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 376315 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@376339 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	1bba4b00b2	Add MALLOC_DEBUG enhancements. * Makes malloc() behave like calloc(). It will return a memory block filled with 0x55. A nonzero value. * Makes free() fill the released memory block and boundary fence's with 0xdeaddead. Any pointer use after free is going to have a pointer pointing to 0xdeaddead. The 0xdeaddead pointer is usually an invalid memory address so a crash is expected. * Puts the freed memory block into a circular array so it is not reused immediately. * When the circular array rotates out a memory block to the heap it checks that the memory has not been altered from 0xdeaddead. * Made the astmm_log message wording better. * Made crash if the DO_CRASH menuselect option is enabled and something is found. * Fixed a potential alignment issue on 64 bit systems. struct ast_region.data[] should now be aligned correctly for all platforms. * Extracted region_check_fences() from __ast_free_region() and handle_memory_show(). * Updated handle_memory_show() CLI usage help. Review: https://reviewboard.asterisk.org/r/2182/ ........ Merged revisions 376029 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 376030 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@376048 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Mark Michelson	a65fbf8012	Multiple revisions 375993-375994 ........ r375993 \| mmichelson \| 2012-11-07 11:01:13 -0600 (Wed, 07 Nov 2012) \| 30 lines Fix misuses of timeouts throughout the code. Prior to this change, a common method for determining if a timeout was reached was to call a function such as ast_waitfor_n() and inspect the out parameter that told how many milliseconds were left, then use that as the input to ast_waitfor_n() on the next go-around. The problem with this is that in some cases, submillisecond timeouts can occur, resulting in the out parameter not decreasing any. When this happens thousands of times, the result is that the timeout takes much longer than intended to be reached. As an example, I had a situation where a 3 second timeout took multiple days to finally end since most wakeups from ast_waitfor_n() were under a millisecond. This patch seeks to fix this pattern throughout the code. Now we log the time when an operation began and find the difference in wall clock time between now and when the event started. This means that sub-millisecond timeouts now cannot play havoc when trying to determine if something has timed out. Part of this fix also includes changing the function ast_waitfor() so that it is possible for it to return less than zero when a negative timeout is given to it. This makes it actually possible to detect errors in ast_waitfor() when there is no timeout. (closes issue ASTERISK-20414) reported by David M. Lee Review: https://reviewboard.asterisk.org/r/2135/ ........ r375994 \| mmichelson \| 2012-11-07 11:08:44 -0600 (Wed, 07 Nov 2012) \| 3 lines Remove some debugging that accidentally made it in the last commit. ........ Merged revisions 375993-375994 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 375995 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@376014 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	647810f1d0	Fix stuck DTMF when bridge is broken. When a bridge is broken by an AMI Redirect action or the ChannelRedirect application, an in progress DTMF digit could be stuck sending forever. * Made simulate a DTMF end event when a bridge is broken and a DTMF digit was in progress. (closes issue ASTERISK-20492) Reported by: Jeremiah Gowdy Patches: bridge_end_dtmf-v3.patch.txt (license #6358) patch uploaded by Jeremiah Gowdy Modified to jira_asterisk_20492_v1.8.patch jira_asterisk_20492_v1.8.patch (license #5621) patch uploaded by rmudgett Tested by: rmudgett Review: https://reviewboard.asterisk.org/r/2169/ ........ Merged revisions 375964 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 375965 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@375966 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	f0cd27e027	Refactor ast_timer_ack to return an error and handle the error in timer users Currently, if an acknowledgement of a timer fails Asterisk will not realize that a serious error occurred and will continue attempting to use the timer's file descriptor. This can lead to situations where errors stream to the CLI/log file. This consumes significant resources, masks the actual problem that occurred (whatever caused the timer to fail in the first place), and can leave channels in odd states. This patch propagates the errors in the timing resource modules up through the timer core, and makes users of these timers handle acknowledgement failures. It also adds some defensive coding around the use of timers to prevent using bad file descriptors in off nominal code paths. Note that the patch created by the issue reporter was modified slightly for this commit and backported to 1.8, as it was originally written for Asterisk 10. Review: https://reviewboard.asterisk.org/r/2178/ (issue ASTERISK-20032) Reported by: Jeremiah Gowdy patches: jgowdy-timerfd-6-22-2012.diff uploaded by Jeremiah Gowdy (license 6358) ........ Merged revisions 375893 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 375894 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@375895 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Andrew Latham	794eb78090	Doxygen Updates Replace links to missing text files removed in the 1.6.x series with links to the wiki. Doxygen can handle URLs fine, don't atempt to quote them. Also update the wiki link in the Readme to get everyone on the same page. (issue ASTERISK-20259) ........ Merged revisions 375698 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 375699 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@375702 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	7c69310497	build_tools: Allow Asterisk to report git SHAs in version string. Make git more attractive for managing work-in-progress. Especially convenient when a potential patch set needs to be tested on multiple platforms since one can use git to keep all the test environments in sync independent of a subversion server. Now the Asterisk version will show the exact git SHA5 that was used when building (still appended by "M" if there are local modifications) from a git clone of the Asterisk repository so the developer can more easily know what is actually under test. You will now get this: $ asterisk -V Asterisk GIT-1698298 Instead of this: $ asterisk -V Asterisk UNKNOWN__and_probably_unsupported This has zero impact for those not using git with the exception of an extra test in the configure script to gather git's path. This is necessary to prevent "sudo make install" from failing since git may not be in the path in make's shell environment. (closes issue ASTERISK-20483) Reported by: Shaun Ruffell Patches: 0001-build_tools-Allow-Asterisk-to-report-git-SHAs-in-ver.patch (license #5417) patch uploaded by Shaun Ruffell Modified ........ Merged revisions 375189 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 375190 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@375191 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Mark Michelson	94c0fa9098	Fix some potential misuses of ast_str in the code. Passing an ast_str pointer by value that then calls ast_str_set(), ast_str_set_va(), ast_str_append(), or ast_str_append_va() can result in the pointer originally passed by value being invalidated if the ast_str had to be reallocated. This fixes places in the code that do this. Only the example in ccss.c could result in pointer invalidation though since the other cases use a stack-allocated ast_str and cannot be reallocated. I've also updated the doxygen in strings.h to include notes about potential misuse of the functions mentioned previously. Review: https://reviewboard.asterisk.org/r/2161 ........ Merged revisions 375025 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 375026 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@375027 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Mark Michelson	ccf01fbfdc	Do not use a FILE handle when doing SIP TCP reads. This is used to solve an issue where a poll on a file descriptor does not necessarily correspond to the readiness of a FILE handle to be read. This change makes it so that for TCP connections, we do a recv() on the file descriptor instead. Because TCP does not guarantee that an entire message or even just one single message will arrive during a read, a loop has been introduced to ensure that we only attempt to handle a single message at a time. The tcptls_session_instance structure has also had an overflow buffer added to it so that if more than one TCP message arrives in one go, there is a place to throw the excess. Huge thanks goes out to Walter Doekes for doing extensive review on this change and finding edge cases where code could fail. (closes issue ASTERISK-20212) reported by Phil Ciccone Review: https://reviewboard.asterisk.org/r/2123 ........ Merged revisions 374905 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 374906 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@374914 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Mark Michelson	b5f231501b	Don't make chan_sip export global symbols. During testing, it was discovered that having chan_sip export global symbols was problematic. The biggest problem was that load order was affected. Trying to use realtime could be problematic since in all likelihood the necessary realtime driver(s) would not be loaded before chan_sip. In addition, it was found that it was impossible to use the Digium Phone Module for Asterisk since it must be loaded before chan_sip since it must hook into chan_sip's configuration parsing. The solution is to use a virtual table in the same manner that other modules in Asterisk do, like app_voicemail. (closes issue ASTERISK-20545) Reported by: kmoore git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@374842 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	8943656ccc	Fix a variety of ref counting issues This patch resolves a number of ref leaks that occur primarily on Asterisk shutdown. It adds a variety of shutdown routines to core portions of Asterisk such that they can reclaim resources allocate duringd initialization. Review: https://reviewboard.asterisk.org/r/2137 ........ Merged revisions 374177 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 374178 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@374196 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Sean Bright	1449b2cad0	app_queue: Support persisting and loading of long member lists. Greenlight in #asterisk brought up that he was receiving an error message "Could not create persistent member string, out of space" when running app_queue in Asterisk 10. dump_queue_members() made an assumption that 8K would be enough to store the generated string, but with queues that have large member lists this is not always the case. This patch removes the limitation and uses ast_str instead of a fixed sized buffer. The complicating factor comes from the fact that ast_db_get requires a buffer and buffer size argument, which doesn't let us pull back more than what we pass in, so I introduced a new ast_db_get_allocated() which returns an ast_strdup()'d copy of the value from astdb. As an aside, I did some testing on the maximum size of data that we can store in the BDB library we distribute and was able to store a 10MB string and retrieve it with no problems, so I feel this is a safe patch. Review: https://reviewboard.asterisk.org/r/2136/ ........ Merged revisions 374108 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 374135 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@374150 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Joshua Colp	f8e894e031	Make res_http_websocket an optional dependency on supported platforms for chan_sip. (closes issue ASTERISK-20439) Reported by: sruffell Patches: 0001-chan_sip-websocket-support-is-an-optional-API.patch uploaded by sruffell (license 5417) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@373914 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Joshua Colp	42ebea2f2f	Add support for DTLS-SRTP to res_rtp_asterisk and chan_sip. As mentioned on the review for this, WebRTC has moved towards choosing DTLS-SRTP as the mechanism for key exchange for SRTP. This commit adds support for this but makes it available for normal SIP clients as well. Testing has been done to ensure that this introduces no regressions with existing behavior and also that it functions as expected. Review: https://reviewboard.asterisk.org/r/2113/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@373229 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	7e9bdcc3e0	Named call pickup groups. Fixes, missing functionality, and improvements. * ASTERISK-20383 Missing named call pickup group features: CHANNEL(callgroup) - Need CHANNEL(namedcallgroup) CHANNEL(pickupgroup) - Need CHANNEL(namedpickupgroup) Pickup() - Needs to also select from named pickup groups. * ASTERISK-20384 Using the pickupexten, the pickup channel selection could fail even though there was a call it could have picked up. In a call pickup race when there are multiple calls to pickup and two extensions try to pickup a call, it is conceivable that the loser will not pick up any call even though it could have picked up the next oldest matching call. Regression because of the named call pickup group feature. * See ASTERISK-20386 for the implementation improvements. These are the changes in channel.c and channel.h. * Fixed some locking issues in CHANNEL(). (closes issue ASTERISK-20383) Reported by: rmudgett (closes issue ASTERISK-20384) Reported by: rmudgett (closes issue ASTERISK-20386) Reported by: rmudgett Tested by: rmudgett Review: https://reviewboard.asterisk.org/r/2112/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@373220 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
David M. Lee	0227e81595	Add -fnested-functions compile flag, if needed. In order to use nested functions on some versions of GCC (e.g. GCC on OS X), the -fnested-functions flag must be passed to the compiler. This patch adds detection logic to ./configure to add the flag if necessary. It also adds a comment to utils.h as to why the nested function needs a prototype. (closes issue ASTERISK-20399) Reported by: David M. Lee Review: https://reviewboard.asterisk.org/r/2102/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@373119 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
David M. Lee	061874d811	Fix timeouts for ast_waitfordigit[_full]. ast_waitfordigit_full would simply pass its timeout to ast_waitfor_nandfds, expecting it to decrement the timeout by however many milliseconds were waited. This is a problem if it consistently waits less than 1ms. The timeout will never be decremented, and we wait... FOREVER! This patch makes ast_waitfordigit_full manage the timeout itself. It maintains the previously undocumented behavior that negative timeouts wait forever. (closes issue ASTERISK-20375) Reported by: Mark Michelson Tested by: Mark Michelson Review: https://reviewboard.asterisk.org/r/2109/ ........ Merged revisions 373024 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 373025 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@373029 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Mark Michelson	46b730b070	Fix inability to shutdown gracefully due to an unending channel reference. message.c makes use of a special message queue channel that exists in thread storage. This channel never goes away due to the fact that the taskprocessor used by message.c does not get shut down, meaning that it never ends the thread that stores the channel. This patch fixes the problem by shutting down the taskprocessor when Asterisk is shut down. In addition, the thread storage has a destructor that will release the channel reference when the taskprocessor is destroyed. (closes issue AST-937) Reported by Jason Parker Patches: AST-937.patch uploaded by Mark Michelson (License #5049) Tested by Jason Parker ........ Merged revisions 372885 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@372888 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Kinsey Moore	05cccdea8c	Deprecate chan_gtalk, chan_jingle, and res_jabber chan_gtalk, chan_jingle, and res_jabber are now deprecated in favor of using chan_motif and res_xmpp. They are a feature-equivalent replacement and are written to be more easily maintainable. (closes issue ASTERISK-20298) Review: https://reviewboard.asterisk.org/r/2082/ Reported-by: Leif Madsen git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@372795 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Mark Michelson	3f6a00ae36	Re-fix sending unnegotiated payloads during a P2P RTP bridge. The previous fix still would look in the static_RTP_PT table, which is inappropriate since we specifically want to find a codec that has been negotiated. (closes issue ASTERISK-20296) reported by NITESH BANSAL Patches: codec_negotiation.patch Uploaded by NITESH BANSAL (License #6418) git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/11@372311 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	fb6238899b	Add private representation of caller, connected and redirecting party ids. This patch adds the feature "Private representation of caller, connected and redirecting party ids", as previously discussed with us (DATUS) and Digium. 1. Feature motivation Until now it is quite difficult to modify a party number or name which can only be seen by exactly one particular instantiated technology channel subscriber. One example where a modified party number or name on one channel is spread over several channels are supplementary services like call transfer or pickup. To implement these features Asterisk internally copies caller and connected ids from one channel to another. Another example are extension subscriptions. The monitoring entities (watchers) are notified of state changes and - if desired - of party numbers or names which represent the involving call parties. One major feature where a private representation of party names is essentially needed, i.e. where a party name shall be exclusively signaled to only one particular user, is a private user-specific name resolution for party numbers. A lookup in a private destination-dependent telephone book shall provide party names which cannot be seen by any other user at any time. 2. Feature Description This feature comes along with the implementation of additional private party id elements for caller id, connected id and redirecting ids inside Asterisk channels. The private party id elements can be read or set by the user using Asterisk dialplan functions. When a technology channel is initiating a call, receives an internal connected-line update event, or receives an internal redirecting update event, it merges the corresponding public id with the private id to create an effective party id. The effective party id is then used for protocol signaling. The channel technologies which initially support the private id representation with this patch are SIP (chan_sip), mISDN (chan_misdn) and PRI (chan_dahdi). Once a private name or number on a channel is set and (implicitly) made valid, it is generally used for any further protocol signaling until it is rewritten or invalidated. To simplify the invalidation of private ids all internally generated connected/redirecting update events and also all connected/redirecting update events which are generated by technology channels -- receiving regarding protocol information - automatically trigger the invalidation of private ids. If not using the private party id representation feature at all, i.e. if using only the 'regular' caller-id, connected and redirecting related functions, the current characteristic of Asterisk is not affected by the new extended functionality. 3. User interface Description To grant access to the private name and number representation from the Asterisk dialplan, the CALLERID, CONNECTEDLINE and REDIRECTING dialplan functions are extended by the following data types. The formats of these data types are equal to the corresponding regular 'non-private' already existing data types: CALLERID: priv-all priv-name priv-name-valid priv-name-charset priv-name-pres priv-num priv-num-valid priv-num-plan priv-num-pres priv-subaddr priv-subaddr-valid priv-subaddr-type priv-subaddr-odd priv-tag CONNECTEDLINE: priv-name priv-name-valid priv-name-pres priv-name-charset priv-num priv-num-valid priv-num-pres priv-num-plan priv-subaddr priv-subaddr-valid priv-subaddr-type priv-subaddr-odd priv-tag REDIRECTING: priv-orig-name priv-orig-name-valid priv-orig-name-pres priv-orig-name-charset priv-orig-num priv-orig-num-valid priv-orig-num-pres priv-orig-num-plan priv-orig-subaddr priv-orig-subaddr-valid priv-orig-subaddr-type priv-orig-subaddr-odd priv-orig-tag priv-from-name priv-from-name-valid priv-from-name-pres priv-from-name-charset priv-from-num priv-from-num-valid priv-from-num-pres priv-from-num-plan priv-from-subaddr priv-from-subaddr-valid priv-from-subaddr-type priv-from-subaddr-odd priv-from-tag priv-to-name priv-to-name-valid priv-to-name-pres priv-to-name-charset priv-to-num priv-to-num-valid priv-to-num-pres priv-to-num-plan priv-to-subaddr priv-to-subaddr-valid priv-to-subaddr-type priv-to-subaddr-odd priv-to-tag Reported by: Thomas Arimont Review: https://reviewboard.asterisk.org/r/2030/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@371120 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	18d5041981	Use better libss7 detection test and move libpri compile test. ........ Merged revisions 371012 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 371013 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@371030 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Mark Michelson	9ee8b3c0f6	Extend extension state callbacks to have more information. Quote from review board: This patch extends the extension state callbacks so that monitoring channels (as chan_sip) get more information of the devices which are responsible for an extension state change. The additional information is needed by chan_sip to present names/numbers of the caller and callee in an early-state SIP notification. Users of extenstion state callback not interested in the additional information are not affected by the changes. Motivation: to present the involved party's name/number in an early-state nofification (used by the notified device as a pickup offer) one after another so that a user can see which call he will pick up in an undirected pickup. Such a pickup offer to a user shall indicate the same call (number/name-A calls number/name-B) as the call which would be picked up when an undirected pickup is executed. Users interested in additional state info must use the new functions ast_extension_state_add_extended() resp. ast_extension_state_add_destroy_extended() to register an extended state callback. When the callback is registered this way, an extra member device_state_info of struct ast_state_cb_info is passed to the callback in addition to the aggregated extension state. This container holds an object for every device of the monitored extension hint consisting of the device name, the device state and a channel reference to the channel which (presumably) caused the device state. The information is used by chan_sip for early-state notifications. When the state of a device changes and the new state contains AST_EVENT_RINGING, an early-state notification is sent to the subscribed devices with the caller/callee names/numbers of the oldest ringing channel of the monitored extension. The notified user may then invoke a direct pickup, which will pickup exactly this channel. Users of the old non-extended callbacks will only be called when the aggregated state did change (same behavior as before). Users of the extended callback will also be called when the state is unchanged but does contain AST_EVENT_RINGING. That could be the case if two channels are ringing at one device and one of them hangs up, so the aggregated state does not change. This way the monitoring channel can create a new early-state notification with the now ringing party-ids. Review: https://reviewboard.asterisk.org/r/2048 This contribution comes from Guenther Kelleter git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370979 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Mark Michelson	eb9e645a27	Allow support for early media on AMI originates and call files. This is based on the work done by Olle Johansson on review board. The idea is that the channel specified in an AMI originate or call file is typically not connected to the outgoing extension until the channel has been answered. With this change, an EarlyMedia header can be specified for AMI originates and an early_media option can be specified in call files. With this option set, once early media is received on a channel, it will be connected with the outgoing extension. (closes issue ASTERISK-18644) Reported by Olle Johansson Review: https://reviewboard.asterisk.org/r/1472 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370951 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Joshua Colp	8c5333f34e	Payload and RTP code are must remain separate since in non-Asterisk format cases they differ. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370860 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Joshua Colp	15e41c7542	Reduce memory consumption significantly for users of the RTP engine API by storing only the payloads present and in use instead of every possible one. Review: https://reviewboard.asterisk.org/r/2052/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370832 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	5c4578f4ad	Add named callgroups/pickupgroups This patch adds named calledgroups/pickupgroups to Asterisk. Named groups are implemented in parallel to the existing numbered callgroup/pickupgroup implementation. However, unlike the existing implementation, which is limited to a maximum of 64 defined groups, the number of defined groups allowed for named callgroups/pickupgroups is effectively unlimited. Named groups are configured with the keywords "namedcallgroup" and "namedpickupgroup". This corresponds to the numbered group definitions of "callgroup" and "pickupgroup". Note that as the implementation of named groups coexists with the existing numbered implementation, a defined named group of "4" does not equate to numbered group 4. Support for the named groups has been added to the SIP, DAHDI, and mISDN channel drivers. Review: https://reviewboard.asterisk.org/r/2043 Uploaded by: Guenther Kelleter(license #6372) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370831 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	ea0732def2	Make astobj2.h not include linkedlists.h. Using astobj2 does not require linkedlists.h be included even though astob2 uses linked lists internally. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370720 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Kinsey Moore	9b16c8b0f6	Clean up and ensure proper usage of alloca() This replaces all calls to alloca() with ast_alloca() which calls gcc's __builtin_alloca() to avoid BSD semantics and removes all NULL checks on memory allocated via ast_alloca() and ast_strdupa(). (closes issue ASTERISK-20125) Review: https://reviewboard.asterisk.org/r/2032/ Patch-by: Walter Doekes (wdoekes) ........ Merged revisions 370642 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 370643 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370655 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Russell Bryant	fd11146592	Add a "corosync ping" CLI command. This patch adds a new CLI command to the res_corosync module. It is primarily used as a debugging tool. It lets you fire off an event which will cause res_corosync on other nodes in the cluster to place messages into the logger if everything is working ok. It verifies that the corosync communication is working as expected. I didn't put anything in the CHANGES file for this, because this module is new in Asterisk 11. There is already a generic "res_corosync new module" entry in there so I figure that covers it just fine. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370535 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Kevin P. Fleming	7d4ccea736	Enable usage of system-provided NetBSD editline library if available. This patch changes the Asterisk configure script and build system to detect the presence of the NetBSD editline library (libedit) on the system. If it is found, it will be used in preference to the version included in the Asterisk source tree. (closes issue ASTERISK-18725) Reported by: Jeffrey C. Ollie Review: https://reviewboard.asterisk.org/r/1528/ Patches: 0001-Allow-linking-building-against-an-external-editline.patch uploaded by jcollie (license #5373) (heavily modified by kpfleming) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370481 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago

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