summaryrefslogtreecommitdiff
path: root/main
AgeCommit message (Collapse)Author
2011-08-10Merged revisions 331462 via svnmerge from Richard Mudgett
https://origsvn.digium.com/svn/asterisk/branches/10 ................ r331462 | rmudgett | 2011-08-10 15:41:35 -0500 (Wed, 10 Aug 2011) | 37 lines Merged revisions 331461 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r331461 | rmudgett | 2011-08-10 15:29:59 -0500 (Wed, 10 Aug 2011) | 30 lines Output of queue log not started until logger reloaded. ASTERISK-15863 caused a regression with queue logging. The output of the queue log is not started until the logger configuration is reloaded. * Queue log initialization is completely delayed until the first message is posted to the queue log system. Including the initial opening of the queue log file. * Fixed rotate_file() ROTATE strategy to give the file just rotated out to the configured exec function after rotate. Just like the other strategies. * Fixed logger reload to always post the queue reload entry instead of just if there is a queue log file. * Refactored some code to eliminate some redundancy and to reduce stack utilization. (closes issue ASTERISK-17036) JIRA SWP-2952 Reported by: Juan Carlos Valero Patches: jira_asterisk_17036_v1.8.patch (license #5621) patch uploaded by rmudgett Tested by: rmudgett (closes issue ASTERISK-18208) Reported by: Christian Pinedo Review: https://reviewboard.asterisk.org/r/1333/ ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@331463 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-08-10Merged revisions 331420 via svnmerge from Richard Mudgett
https://origsvn.digium.com/svn/asterisk/branches/10 ........ r331420 | rmudgett | 2011-08-10 14:07:53 -0500 (Wed, 10 Aug 2011) | 2 lines Make sure feature_request_and_dial() initializes outstate if passed in. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@331421 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-08-10Merged revisions 331418 via svnmerge from Richard Mudgett
https://origsvn.digium.com/svn/asterisk/branches/10 ........ r331418 | rmudgett | 2011-08-10 13:25:08 -0500 (Wed, 10 Aug 2011) | 6 lines Revert -r318141. It was a band-aid that only partially fixed parking. A better fix is on reviewboard review 1358. (issue ASTERISK-17374) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@331419 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-08-10Merged revisions 331316 via svnmerge from Kinsey Moore
https://origsvn.digium.com/svn/asterisk/branches/10 ................ r331316 | kmoore | 2011-08-10 08:48:41 -0500 (Wed, 10 Aug 2011) | 15 lines Merged revisions 331315 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r331315 | kmoore | 2011-08-10 08:47:46 -0500 (Wed, 10 Aug 2011) | 8 lines AMI action ModuleReload returns Error if Module: missing or empty An empty string was not being checked for properly causing identification of the module to be reloaded to fail and return an Error with message "No such module." (closes issue AST-616) ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@331317 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-08-09Merged revisions 331265 via svnmerge from Richard Mudgett
https://origsvn.digium.com/svn/asterisk/branches/10 ................ r331265 | rmudgett | 2011-08-09 18:12:49 -0500 (Tue, 09 Aug 2011) | 22 lines Merged revisions 331248 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r331248 | rmudgett | 2011-08-09 17:12:59 -0500 (Tue, 09 Aug 2011) | 15 lines Misc minor items found in code. * Add some reentrancy protection in pbx.c when creating the contexts_table hash table. * Fix inverted test in chan_sip.c conditional code. * Fix uninitialized variable and use of the wrong variable in chan_iax2.c. * Fix test of return value in app_parkandannounce.c. Explicitly testing for -1 is bad if the function does not actually return that value when it fails. * Fixup some comments and add some curly braces in features.c. ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@331266 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-08-09Allow ENUM query functions to report lookup errorsKinsey Moore
The ENUM dialplan functions do not report DNS query errors properly. It is useful to differentiate between failed query (e.g. non-existent domain) vs. no data records of the appropriate type. This is required to make overlapped dialing work. (closes issue ASTERISK-13769) Review: https://reviewboard.asterisk.org/r/1355/ Patch-by: Timo Teras git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@331201 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-08-08Merged revisions 331041 via svnmerge from Terry Wilson
https://origsvn.digium.com/svn/asterisk/branches/10 ........ r331041 | twilson | 2011-08-08 16:12:51 -0500 (Mon, 08 Aug 2011) | 6 lines Replace AMI Unlink events with Bridge events A previous update converted some of the Link and Unlink events to Bridge events, but a couple of Unlink events were missed. This patch rectifies the situation. (closes issues ASTERISK-17455) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@331042 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-08-03Merged revisions 330763 via svnmerge from Kinsey Moore
https://origsvn.digium.com/svn/asterisk/branches/10 ................ r330763 | kmoore | 2011-08-03 10:15:26 -0500 (Wed, 03 Aug 2011) | 16 lines Merged revisions 330762 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r330762 | kmoore | 2011-08-03 10:14:36 -0500 (Wed, 03 Aug 2011) | 9 lines editing files in main/editline does not ensure rebuild of libedit.a When editing a source file in main/editline, the build system does not rebuild libedit.a and uses the already existing one instead. Adding a PHONY to CHECK_SUBDIR fixes this problem. (closes issue ASTERISK-16221) Patch-by: Walter Doekes ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@330764 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-08-01Merged revisions 330434 via svnmerge from Kinsey Moore
https://origsvn.digium.com/svn/asterisk/branches/10 ................ r330434 | kmoore | 2011-08-01 10:23:29 -0500 (Mon, 01 Aug 2011) | 16 lines Merged revisions 330433 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r330433 | kmoore | 2011-08-01 10:22:10 -0500 (Mon, 01 Aug 2011) | 9 lines Incorrect playback for Spanish in some circumstances When you say the time in spanish and it is 01:00 - 01:59 or 13:00 - 13:59 you must use female pronunciation "1F". The function "say_date_with_format_es" does not take this in account. (closes ASTERISK-15016) Patch-by: Luis Jimenez ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@330435 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-07-31Fixed compiler warning and a couple prototype mismatches.Richard Mudgett
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@330379 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-07-31Merged revisions 330369 via svnmerge from Richard Mudgett
https://origsvn.digium.com/svn/asterisk/branches/10 ................ r330369 | rmudgett | 2011-07-30 18:57:56 -0500 (Sat, 30 Jul 2011) | 11 lines Merged revisions 330368 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r330368 | rmudgett | 2011-07-30 18:56:29 -0500 (Sat, 30 Jul 2011) | 4 lines Remove some redundant locking code in ast_do_masquerade(). Also updated some comments. ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@330370 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-07-30Merged revisions 330312 via svnmerge from Gregory Nietsky
https://origsvn.digium.com/svn/asterisk/branches/10 ................ r330312 | irroot | 2011-07-30 17:34:41 +0200 (Sat, 30 Jul 2011) | 15 lines Merged revisions 330311 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r330311 | irroot | 2011-07-30 17:25:16 +0200 (Sat, 30 Jul 2011) | 9 lines prevent double masqurading channels when one is been hung up and deadlock avoidance is used. There is a race condition in ast_do_masquerade / ast_hangup (at least) Reported by me signed off by schmidts with input from David Vossel Review: https://reviewboard.asterisk.org/r/1323/ ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@330313 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-07-29astobj2: Avoid using temporary objects + ao2_find() with OBJ_POINTER.Russell Bryant
There is a fairly common pattern making its way through the code base where we put a temporary object on the stack so we can call ao2_find() with OBJ_POINTER. The purpose is so that it can be passed into the object hash function. However, this really seems like a hack and potentially error prone. This patch is a first stab at approach to avoid having to do that. It adds a new flag, OBJ_KEY, which can be used instead of OBJ_POINTER in these situations. Then, the hash function can know whether it was given an object or some custom data to hash. The patch also changes some uses of ao2_find() for iax2_user and iax2_peer objects to reflect how OBJ_KEY would be used. So long, and thanks for all the fish. Review: https://reviewboard.asterisk.org/r/1184/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@330273 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-07-28Merged revisions 330108 via svnmerge from Terry Wilson
https://origsvn.digium.com/svn/asterisk/branches/10 ................ r330108 | twilson | 2011-07-28 16:44:31 -0500 (Thu, 28 Jul 2011) | 9 lines Merged revisions 330107 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r330107 | twilson | 2011-07-28 16:42:41 -0500 (Thu, 28 Jul 2011) | 2 lines Make console colors work for TERM=xterm-256color ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@330109 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-07-27reverting 329840 due to failing tests. Going to change this feature to be ↵Jonathan Rose
purely optional. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@329856 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-07-27Adds cdr logging of calls resulting in CONGESTIONJonathan Rose
Applies a patch made a long time ago by alecdavis which adds a CDR feature for logging calls that failed due to congestion. (closes issue #15907) Reported by: alecdavis Patches: cdr_congestion.diff.txt uploaded by alecdavis (license #5546) Review: https://reviewboard.asterisk.org/r/454/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@329835 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-07-27Merged revisions 329670 via svnmerge from Sean Bright
https://origsvn.digium.com/svn/asterisk/branches/10 ........ r329670 | seanbright | 2011-07-27 11:25:53 -0400 (Wed, 27 Jul 2011) | 2 lines Sort the module list so that 'module show' is alphabetical. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@329671 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-07-26Merged revisions 329528 via svnmerge from Jonathan Rose
https://origsvn.digium.com/svn/asterisk/branches/10 ................ r329528 | jrose | 2011-07-26 08:52:34 -0500 (Tue, 26 Jul 2011) | 24 lines Merged revisions 329527 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r329527 | jrose | 2011-07-26 08:25:35 -0500 (Tue, 26 Jul 2011) | 17 lines Fixes some voicemail forwarding behavior based around prepend mode. Formerly, prepend forwarding would have the user record a message with no useful prompt and an expectation for the user to push a button on the phone when finished recording. If a length of silence was detected instead, the recording would be canceled and the user would re-enter the voicemail forwarding menu. Subsequent time-outs in prepend recording would also bug out in the sense that they would write over the original message and get sent to the recipient regardless of whether they timed out or were accepted. This patch fixes this issue and adds a prompt which will be played after a timeout informing the user that they needed to press a button. Currently, the sound files that we have are somewhat inadquate for this, so after the call we simply have Allison say "Please try again. Then press pound." which actually relies on two separate sound files. Just one would be more appropriate. reporter: Vlad Povorozniuc Review: https://reviewboard.asterisk.org/r/1327/ ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@329530 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-07-25Merged revisions 329472 via svnmerge from Paul Belanger
https://origsvn.digium.com/svn/asterisk/branches/10 ................ r329472 | pabelanger | 2011-07-25 15:55:33 -0400 (Mon, 25 Jul 2011) | 9 lines Merged revisions 329471 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r329471 | pabelanger | 2011-07-25 15:49:40 -0400 (Mon, 25 Jul 2011) | 2 lines Decrease verbose messages to debug, to help clean up CLI. ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@329473 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-07-25dsp_process was enhanced to work with alaw and ulaw in addition to slin.Gregory Nietsky
noticed that some functions could be refactored here it is. Reported by: irroot Tested by: irroot, mnicholson Review: https://reviewboard.asterisk.org/r/1304/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@329432 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-07-22Merged revisions 329334 via svnmerge from Richard Mudgett
https://origsvn.digium.com/svn/asterisk/branches/10 ........ r329334 | rmudgett | 2011-07-22 16:14:22 -0500 (Fri, 22 Jul 2011) | 1 line Make use less redundant loop construct for iterating over hints. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@329335 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-07-22Merged revisions 329331 via svnmerge from Richard Mudgett
https://origsvn.digium.com/svn/asterisk/branches/10 ................ r329331 | rmudgett | 2011-07-22 15:43:07 -0500 (Fri, 22 Jul 2011) | 55 lines Merged revisions 329299 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r329299 | rmudgett | 2011-07-22 10:44:58 -0500 (Fri, 22 Jul 2011) | 48 lines Deadlocks dealing with dialplan hints during reload. There are two remaining different deadlocks reported dealing with dialplan hints. The deadlock in ASTERISK-17666 is caused by invalid locking order in ast_remove_hint(). The hints container must be locked before the hint object. The deadlock in ASTERISK-17760 is caused by a catch-22 situation in handle_statechange(). The deadlock is caused by not having the conlock before calling the watcher callbacks. Unfortunately, having that lock causes a different deadlock as reported in ASTERISK-16961. * Fixed ast_remove_hint() locking order. * Made handle_statechange() no longer call the watcher callbacks holding any locks that matter. * Made hint ao2 destructor do the watcher callbacks for extension deactivation to guarantee that they get called. * Fixed hint reference leak in ast_add_hint() if the callback container constructor failed. * Fixed hint reference leak in complete_core_show_hint() for every hint it found for CLI tab completion. * Adjusted locking in ast_merge_contexts_and_delete() for safety. * Added context_merge_lock to prevent ast_merge_contexts_and_delete() and handle_statechange() from interfering with each other. * Fixed ast_change_hint() not taking into account that the extension is used for the hash key. (closes issue ASTERISK-17666) Reported by: irroot Tested by: irroot JIRA SWP-3318 (closes issue ASTERISK-17760) Reported by: Byron Clark Tested by: irroot JIRA SWP-3393 Review: https://reviewboard.asterisk.org/r/1313/ ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@329332 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-07-21Merged revisions 329257 via svnmerge from Russell Bryant
https://origsvn.digium.com/svn/asterisk/branches/10 ........ r329257 | russell | 2011-07-21 15:22:36 -0500 (Thu, 21 Jul 2011) | 2 lines s/1.10/10.0/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@329258 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-07-21Merged revisions 329145 via svnmerge from Richard Mudgett
https://origsvn.digium.com/svn/asterisk/branches/10 ................ r329145 | rmudgett | 2011-07-21 11:52:17 -0500 (Thu, 21 Jul 2011) | 16 lines Merged revisions 329144 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r329144 | rmudgett | 2011-07-21 11:46:21 -0500 (Thu, 21 Jul 2011) | 9 lines Dialplan bridge() app mutex 'current_dest_chan' freed more times than we've locked! This appears to be a leftover from when ast_channel was converted to ao2 objects. Simply removed the extraneous unlock. (closes issue ASTERISK-17772) ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@329146 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-07-19Merged revisions 328824 via svnmerge from Kinsey Moore
https://origsvn.digium.com/svn/asterisk/branches/1.10 ................ r328824 | kmoore | 2011-07-19 13:05:21 -0500 (Tue, 19 Jul 2011) | 18 lines Merged revisions 328823 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r328823 | kmoore | 2011-07-19 12:57:18 -0500 (Tue, 19 Jul 2011) | 11 lines RTP bridge away with inband DTMF and feature detection When deciding whether Asterisk was allowed to bridge the call away from the core, chan_sip did not take into account the usage of features on dialed channels that require monitoring of DTMF on channels utilizing inband DTMF. This would cause Asterisk to allow the call to be locally or remotely bridged, preventing access to the data required to detect activations of such features. (closes 17237) Review: https://reviewboard.asterisk.org/r/1302/ ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@328825 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-07-18Merged revisions 328609 via svnmerge from Mark Murawki
https://origsvn.digium.com/svn/asterisk/branches/1.10 ................ r328609 | markm | 2011-07-18 08:37:53 -0400 (Mon, 18 Jul 2011) | 15 lines Merged revisions 328593 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r328593 | markm | 2011-07-18 08:06:50 -0400 (Mon, 18 Jul 2011) | 8 lines Fixed invalid read and null pointer deref on asterisk shutdown. In some cases when starting asterisk with -c and hitting control-c to shutdown, there will be an invalid read and null pointer deref causing a crash. (closes issue ASTERISK-17927) Reported by: Mark Murawski Tested by: Mark Murawski, Kinsey Moore, Tilghman Lesher ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@328610 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-07-15Merged revisions 328329 via svnmerge from Richard Mudgett
https://origsvn.digium.com/svn/asterisk/branches/1.10 ........ r328329 | rmudgett | 2011-07-14 19:19:32 -0500 (Thu, 14 Jul 2011) | 2 lines Make hint watcher callback take const strings for context and exten parameters. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@328344 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-07-14Merged revisions 328247 via svnmerge from Leif Madsen
https://origsvn.digium.com/svn/asterisk/branches/1.10 ................ r328247 | lmadsen | 2011-07-14 16:25:31 -0400 (Thu, 14 Jul 2011) | 14 lines Merged revisions 328209 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r328209 | lmadsen | 2011-07-14 16:13:06 -0400 (Thu, 14 Jul 2011) | 6 lines Introduce <support_level> tags in MODULEINFO. This change introduces MODULEINFO into many modules in Asterisk in order to show the community support level for those modules. This is used by changes committed to menuselect by Russell Bryant recently (r917 in menuselect). More information about the support level types and what they mean is available on the wiki at https://wiki.asterisk.org/wiki/display/AST/Asterisk+Module+Support+States ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@328259 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-07-14Merged revisions 328162 via svnmerge from Matthew Nicholson
https://origsvn.digium.com/svn/asterisk/branches/1.10 ........ r328162 | mnicholson | 2011-07-14 12:46:32 -0500 (Thu, 14 Jul 2011) | 3 lines tune the v21 preamble detector to properly detect the desired sequence of hits and misses ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@328163 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-07-12Merged revisions 327950 via svnmerge from Kevin P. Fleming
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r327950 | kpfleming | 2011-07-12 17:53:53 -0500 (Tue, 12 Jul 2011) | 14 lines Correct double-free situation in manager output processing. The process_output() function calls ast_str_append() and xml_translate() on its 'out' parameter, which is a pointer to an ast_str buffer. If either of these functions need to reallocate the ast_str so it will have more space, they will free the existing buffer and allocate a new one, returning the address of the new one. However, because process_output only receives a pointer to the ast_str, not a pointer to its caller's variable holding the pointer, if the original ast_str is freed, the caller will not know, and will continue to use it (and later attempt to free it). (reported by jkroon on #asterisk-dev) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@327953 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-07-12do v21 detection instead of CED detection for the fax gatewayMatthew Nicholson
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@327769 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-07-12Send video update frame to new video source in follow_talker correctly.David Vossel
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@327749 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-07-11Updates follow_talker video_mode in confbridge application.David Vossel
follow_talker mode originally echoed the same video stream to all participants. As the primary talker switched around, the video stream would result in the talker seeing themselves. Now the primary talker sees the last person who was talking rather than themselves. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@327640 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-07-11Merged revisions 327512 via svnmerge from Matthew Nicholson
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r327512 | mnicholson | 2011-07-11 08:53:59 -0500 (Mon, 11 Jul 2011) | 2 lines reset our buffer each iteration when doing variable substitution ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@327513 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-07-11Merged revisions 327411 via svnmerge from Tzafrir Cohen
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r327411 | tzafrir | 2011-07-11 13:46:34 +0300 (ב', 11 יול 2011) | 5 lines fix building the Debian armhf (HardFloat) port Fixes http://buildd.debian-ports.org/status/fetch.php?pkg=asterisk&arch=armhf&ver=1%3A1.8.4.4~dfsg-2&stamp=1309935385 (Missing pthreads) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@327413 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-07-08Merged revisions 327106 via svnmerge from Matthew Nicholson
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r327106 | mnicholson | 2011-07-08 14:52:51 -0500 (Fri, 08 Jul 2011) | 11 lines Reset our ast_str before passing it on to dialplan function backends. It is possible for a dialplan backend to not modify the given buffer or ast_str and still return success. This causes any previous value stored in the buffer to be used as if the new function call provided it. Some functions also append to the given buffer assuming it is empty. The test_substitution unit test has also been modified to detect this problem. (closes issue ASTERISK-17878) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@327107 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-07-08Merged revisions 326985 via svnmerge from Richard Mudgett
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r326985 | rmudgett | 2011-07-07 20:08:05 -0500 (Thu, 07 Jul 2011) | 12 lines Some code cleanup in pbx.c * Mostly comment and format changes. * ast_context_remove_extension_callerid() and ast_add_extension_nolock() will write lock the found specific context. * ast_context_find() will now tolerate a NULL name. * Eliminated some inlined versions of find_context() and find_context_locked(). ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@327000 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-07-07Adds pass-through support for codec CELT.David Vossel
This patch adds pass-through support for CELT. CELT formats are defined in codecs.conf and can be configured to any sample rate a CELT endpoint supports. This patch also addresses a crash in channel.c resulting from a frame list being freed incorrectly. This crash was discovered while testing a CELT translator which had to split encoded audio into multiple frames. The codec translator is not a part of this patch, but may be contributed in the future. Review: https://reviewboard.asterisk.org/r/1294/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@326855 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-07-07Use older functions out of deference to older distrosTerry Wilson
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@326750 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-07-06Replace Berkeley DB with SQLite 3Terry Wilson
There were some bugs in the very ancient version of Berkeley DB that Asterisk used. Instead of spending the time tracking down the bugs in the Berkeley code we move to the much better documented SQLite 3. Conversion of the old astdb happens at runtime by running the included astdb2sqlite3 utility. The ast_db API with SQLite 3 backend should behave identically to the old Berkeley backend, but in the future we could offer a much more robust interface. We do not include the SQLite 3 library in the source tree, but instead rely upon the distribution-provided libraries. SQLite is so ubiquitous that this should not place undue burden on administrators. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@326589 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-07-05New feature: AMI Action FilterAddMark Murawki
This adds a new action, FilterAdd to the manager interface that allows control over event filters for the current session (closes issue ASTERISK-16795) Reported by: kobaz Tested by: kobaz,loloski git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@326267 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-07-05Merged revisions 326209 via svnmerge from Matthew Jordan
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r326209 | mjordan | 2011-07-05 08:23:57 -0500 (Tue, 05 Jul 2011) | 7 lines Updated filestream destructor to block until move is complete when cache is used When a cache directory is used, the process is forked and a mv command is executed to move the temporary file to the permanent location. This caused issues with voicemail, where a race condition occurred when the parent expected the file to be in the permanent location prior to the mv command completing. The parent process is now blocked until the mv command completes. (closes issue ASTERISK-17724) Reported by: Adiren P. Tested by: mjordan ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@326210 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-06-30Video support for ConfBridge.David Vossel
Review: https://reviewboard.asterisk.org/r/1288/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@325931 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-06-30copy all flags on asterisk frames instead of just the timing flagMatthew Nicholson
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@325815 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-06-29Merged revisions 325545 via svnmerge from Matthew Nicholson
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r325545 | mnicholson | 2011-06-29 11:18:39 -0500 (Wed, 29 Jun 2011) | 2 lines make framehooks prevent native bridging (for real this time) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@325547 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-06-29Merged revisions 325537 via svnmerge from Matthew Nicholson
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r325537 | mnicholson | 2011-06-29 10:34:47 -0500 (Wed, 29 Jun 2011) | 2 lines don't do native/remote bridging if a framehook is active on the channel ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@325538 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-06-27Merged revisions 324955 via svnmerge from Tilghman Lesher
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r324955 | tilghman | 2011-06-27 11:30:50 -0500 (Mon, 27 Jun 2011) | 5 lines Save and restore errno from within signal handlers. This is recommended by the POSIX standard, as well as by the sigaction(2) manpage for various platforms that we support (e.g. Mac OS X). ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@324961 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-06-23Merged revisions 324652 via svnmerge from David Vossel
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r324652 | dvossel | 2011-06-23 13:23:21 -0500 (Thu, 23 Jun 2011) | 20 lines Merged revisions 324634 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r324634 | dvossel | 2011-06-23 13:18:46 -0500 (Thu, 23 Jun 2011) | 13 lines Merged revisions 324627 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r324627 | dvossel | 2011-06-23 13:16:52 -0500 (Thu, 23 Jun 2011) | 7 lines Addresses AST-2011-010, remote crash in IAX2 driver Thanks to twilson for identifying the issue and providing the patches. AST-2011-010 ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@324664 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-06-22Merged revisions 324484 via svnmerge from Terry Wilson
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r324484 | twilson | 2011-06-22 13:52:04 -0500 (Wed, 22 Jun 2011) | 20 lines Stop sending IPv6 link-local scope-ids in SIP messages The idea behind the patch listed below was used, but in a more targeted manner. There are now address stringification functions for addresses that are meant to be sent to a remote party. Link-local scope-ids only make sense on the machine from which they originate and so are stripped in the new functions. There is also a host sanitization function added to chan_sip which is used for when peer and dialog tohost fields or sip_registry hostnames are used to craft a SIP message. Also added are some basic unit tests for netsock2 address parsing. (closes issue ASTERISK-17711) Reported by: ch_djalel Patches: asterisk-1.8.3.2-ipv6_ll_scope.patch uploaded by ch_djalel (license 1251) Review: https://reviewboard.asterisk.org/r/1278/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@324487 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-06-21Merged revisions 324364 via svnmerge from David Vossel
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r324364 | dvossel | 2011-06-21 15:11:52 -0500 (Tue, 21 Jun 2011) | 10 lines Fixes locking inversion issue in ast_async_goto() During this function we can not hold the "chan" lock while doing the masquerade, the explicit goto on the tmp chan, or the channel alloc. Instead we need to get the channel lock, store off information about the channel that we need, and then let the channel lock go for the remainder of the function. Review: https://reviewboard.asterisk.org/r/1275/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@324365 65c4cc65-6c06-0410-ace0-fbb531ad65f3
generated by cgit v1.2.3 (git 2.39.1) at 2024-07-20 01:01:09 +0000