2529 Commits (34cbefe62fc9851a03aaa3ace803b6c21b68fc25)

Author	SHA1	Message	Date
Richard Mudgett	7c0b94a3e5	Remove init_framer(). It no longer does anything. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@377246 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	77561124b0	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 ........ Merged revisions 377137 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@377138 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	dc653e09bb	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 ........ Merged revisions 376629 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@376630 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	588e90a4d9	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 ........ Merged revisions 376588 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@376589 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	4ccf2c7aa5	Add red-black tree container type to astobj2. ... * Add red-black tree container type. * Add CLI command "astobj2 container dump <name>" * Added ao2_container_dump() so the container could be dumped by other modules for debugging purposes. * Changed ao2_container_stats() so it can be used by other modules like ao2_container_check() for debugging purposes. * Updated the unit tests to check red-black tree containers. (closes issue ASTERISK-19970) Reported by: rmudgett Tested by: rmudgett Review: https://reviewboard.asterisk.org/r/2110/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@376575 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
David M. Lee	752ffc69d4	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 ........ Merged revisions 376339 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@376341 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	3a9640b605	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 ........ Merged revisions 376048 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@376049 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Mark Michelson	f2bb9afe17	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 ........ Merged revisions 376014 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@376015 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	6ad0126425	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 ........ Merged revisions 375966 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@375967 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	a0c363e227	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 ........ Merged revisions 375895 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@375896 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	b0c3d288f2	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 ........ Merged revisions 375191 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@375192 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Andrew Latham	6c20cf2d8a	Doxygen Updates - Title update ... Update and extend the configuration_file group and enable linking. Commit other cleanups from multi-version Doxygen testing. Update title that was left behind many years ago. (issue ASTERISK-20259) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@375182 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Mark Michelson	e9ab568f88	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 ........ Merged revisions 375027 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@375044 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Mark Michelson	c7b23cbb0a	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 ........ Merged revisions 374914 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@374924 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Andrew Latham	7226606f77	Continue to group config files ... (issue ASTERISK-20259) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@374888 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Mark Michelson	825607e09b	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 ........ Merged revisions 374842 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@374849 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Joshua Colp	d78f7f92b2	Add support for applying direct media ACLs between differing channel technologies. ... Review: https://reviewboard.asterisk.org/r/2122/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@374414 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	a094707d51	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 ........ Merged revisions 374196 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@374197 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Andrew Latham	4e228fce03	Doxygen Cleanup ... Start adding configuration file linking and pages. Add module loading doxygen block. Breaking up commits to keep it easy to track (issue ASTERISK-20259) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@374167 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Sean Bright	b9eeff1521	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 ........ Merged revisions 374150 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@374151 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Joshua Colp	0fc114dc65	Add support for retrieving engine specific settings using the speech API and from dialplan. ... (closes issue ASTERISK-17136) Reported by: kenner git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@374096 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Joshua Colp	9f55e5e928	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) ........ Merged revisions 373914 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@373915 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Mark Michelson	fdfb3ae5fa	Allow for redirecting reasons to be set to arbitrary strings. ... This allows for the REDIRECTING dialplan function to be used to set the reason to any string. The SIP channel driver has been modified to set the redirecting reason string to the value received in a Diversion header. In addition, SIP 480 response reason text will set the redirecting reason as well. (closes issue AST-942) reported by Malcolm Davenport (closes issue AST-943) reported by Malcolm Davenport Review: https://reviewboard.asterisk.org/r/2101 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@373701 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Andrew Latham	fd98835f1f	Doxygen Updates Janitor Work ... * Whitespace, doc-blocks, spelling, case, missing and incorrect tags. * Add cleanup to Makefile for the Doxygen configuration update * Start updating Doxygen configuration for cleaner output * Enable inclusion of configuration files into documentation * remove mantisworkflow... * update documentation README * Add markup to Tilghman's email and talk with him about updating his email, he knows... * no code changes on this commit other than the mentioned Makefile change (issue ASTERISK-20259) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@373384 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Andrew Latham	6f61cb50c5	Doxygen Updates - janitor work ... Doxygen updates including mistakes, misspellings, missing parameters, updates for Doxygen style. Some missing txt file links are removed but their content or essense will be included in some later updates. A majority of the txt files were removed in the 1.6 era but never noted. The HR and EXTREF are simple changes that make the documentation more compatable with more versions of Doxygen. Further updates coming. (issue ASTERISK-20259) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@373330 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Joshua Colp	e8380afc8a	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/ ........ Merged revisions 373229 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@373234 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	da5944fc56	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/ ........ Merged revisions 373220 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@373221 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
David M. Lee	f8d815e19f	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/ ........ Merged revisions 373119 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@373120 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
David M. Lee	192e6a0f7a	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 ........ Merged revisions 373029 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@373046 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	fb1d9a90a4	Enhance astobj2 to support other types of containers. ... The new API allows for sorted containers, insertion options, duplicate handling options, and traversal order options. * Adds the ability for containers to be sorted when they are created. * Adds container creation options to handle duplicates when they are inserted. * Adds container creation option to insert objects at the beginning or end of the container traversal order. * Adds OBJ_PARTIAL_KEY to allow searching with a partial key. The partial key works similarly to the OBJ_KEY flag. (The real search speed improvement with this flag will come when red-black trees are added.) * Adds container traversal and iteration order options: Ascending and Descending. * Adds an AST_DEVMODE compile feature to check the stats and integrity of registered containers using the CLI "astobj2 container stats <name>" and "astobj2 container check <name>". The channels container is normally registered since it is one of the most important containers in the system. * Adds ao2_iterator_restart() to allow iteration to be restarted from the beginning. * Changes the generic container object to have a v_method table pointer to support other types of containers. * Changes the container nodes holding objects to be ref counted. The ref counted nodes and v_method table pointer changes pave the way to allow other types of containers. * Includes a large astobj2 unit test enhancement that tests the new features. (closes issue ASTERISK-19969) Reported by: rmudgett Review: https://reviewboard.asterisk.org/r/2078/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@372997 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Mark Michelson	8963829390	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 ........ Merged revisions 372888 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@372891 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Kinsey Moore	d96b832787	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 ........ Merged revisions 372795 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@372796 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Mark Michelson	be500bbafb	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) ........ Merged revisions 372311 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@372319 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	8018b879a2	Clean up doxygen warnings ... This patch fixes numerous doxygen warnings across Asterisk. It also updates the makefile to regenerate the doxygen configuration on the local system before running doxygen to help prevent warnings/errors on the local system. Much thanks to Andrew for tackling one of the Asterisk janitor projects! (issue ASTERISK-20259) Reported by: Andrew Latham Patches: doxygen_partial.diff uploaded by Andrew Latham (license 5985) make_progdocs.diff uploaded by Andrew Latham (license 5985) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@371989 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	f075e7631f	Ensure alignment of in[] field in MD5Context struct. ... The struct MD5Context character buffer is cast to an int32_t* without making sure that said buffer is aligned. Since the buffer follows two uint32_t's, the chance of 'in' being (32 bits) unaligned is nil in practice. But adding code to ensure that 'in' stays aligned costs nothing and removes all doubts about the casts being safe. (closes issue ASTERISK-20241) Reported by: Walter Doekes Patches: tmp.diff (license #5674) patch uploaded by Walter Doekes git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@371952 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Mark Michelson	89a5ff859d	Add scoped locks to Asterisk. ... With the SCOPED_LOCK macro, you can create a variable that locks a specific lock and unlocks the lock when the variable goes out of scope. This is useful for situations where many breaks, continues, returns, or other interruptions would require separate unlock statements. With a scoped lock, these aren't necessary. There are specializations for mutexes, read locks, write locks, ao2 locks, ao2 read locks, ao2 write locks, and channel locks. Each of these is a SCOPED_LOCK at heart though. Review: https://reviewboard.asterisk.org/r/2060 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@371582 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
Kevin P. Fleming	b5193428a7	Enable usage of system-provided iLBC library. ... The WebRTC version of the iLBC codec is now package as a library and is available on some platforms. This patch allows codec_ilbc to be built against that library if it is present. Review: https://reviewboard.asterisk.org/r/1964/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370407 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	b6a0ae0b35	Unit tests for the Jitter Buffer API; remove unnecessary resync ... This patch includes the following: * Unit tests for the abstract Jitter Buffer API. This includes both fixed and adaptive flavors, testing nominal creation, frame input, frame retrieval, resyncing; off nominal frame input overflow, out of order, and others. * Tweaks to the abstract_jb API to remove the unnecessary resync_threshold parameter from the create function (resync_threshold is already in the struct passed into the create function) * Ensure the fixed jitter buffer is empty before destroying it, to avoid an ASSERT * Don't "resync" the adaptive jitter buffer. The mechanism that was being used actually causes the jitter buffer to think its being overflowed by going around the jitterbuf API and attempting to 'resynch' it improperly. If a resync is needed, the jitter buffer will do it properly by itself. Note that this is only an optimization needed for trunk, as the worst that happens is the loss of three voice packets before the adaptive jitter buffer will resync anyway. Review: https://reviewboard.asterisk.org/r/2035 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370387 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Kinsey Moore	cb9756daa2	Add hangupcause translation support ... The HANGUPCAUSE hash (trunk only) meant to replace SIP_CAUSE has now been replaced with the HANGUPCAUSE and HANGUPCAUSE_KEYS dialplan functions to better facilitate access to the AST_CAUSE translations for technology-specific cause codes. The HangupCauseClear application has also been added to remove this data from the channel. (closes issue SWP-4738) Review: https://reviewboard.asterisk.org/r/2025/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@370316 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago