3212 Commits (d4e239fa603c7521c76dc83e2e3a4379caaafb8f)

Author	SHA1	Message	Date
Jonathan Rose	d170e5e829	reverting 329840 due to failing tests. Going to change this feature to be purely optional. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@329856 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Jonathan Rose	3ee80d6a90	Adds cdr logging of calls resulting in CONGESTION ... 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	15 years ago
Sean Bright	5858e755e4	Merged revisions 329670 via svnmerge from ... 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	15 years ago
Jonathan Rose	462e0fe530	Merged revisions 329528 via svnmerge from ... 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	15 years ago
Paul Belanger	06343443e1	Merged revisions 329472 via svnmerge from ... 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	15 years ago
Gregory Nietsky	3b1cc6de8d	dsp_process was enhanced to work with alaw and ulaw in addition to slin. ...	15 years ago
Richard Mudgett	c0f592df46	Merged revisions 329334 via svnmerge from ... 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	15 years ago
Richard Mudgett	a5c65bb939	Merged revisions 329331 via svnmerge from ... 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	15 years ago
Russell Bryant	f243d129c9	Merged revisions 329257 via svnmerge from ... 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	15 years ago
Richard Mudgett	3b80737787	Merged revisions 329145 via svnmerge from ... 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	15 years ago
Kinsey Moore	1dc97eb69b	Merged revisions 328824 via svnmerge from ... 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	15 years ago
Mark Murawki	23140a044e	Merged revisions 328609 via svnmerge from ... 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	15 years ago
Richard Mudgett	145c174565	Merged revisions 328329 via svnmerge from ... 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	15 years ago
Leif Madsen	a525edea59	Merged revisions 328247 via svnmerge from ... 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	15 years ago
Matthew Nicholson	e46aea196c	Merged revisions 328162 via svnmerge from ... 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	15 years ago
Kevin P. Fleming	d37ac6a8a0	Merged revisions 327950 via svnmerge from ... 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	15 years ago
Matthew Nicholson	3f44b08b7b	do v21 detection instead of CED detection for the fax gateway ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@327769 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
David Vossel	3e272bb0b6	Send video update frame to new video source in follow_talker correctly. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@327749 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
David Vossel	881173268c	Updates follow_talker video_mode in confbridge application. ... 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	15 years ago
Matthew Nicholson	7eda60dca1	Merged revisions 327512 via svnmerge from ... 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	15 years ago
Tzafrir Cohen	55eaa8568c	Merged revisions 327411 via svnmerge from ... 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	15 years ago
Matthew Nicholson	2ac180275d	Merged revisions 327106 via svnmerge from ... 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	15 years ago
Richard Mudgett	a0cbad527c	Merged revisions 326985 via svnmerge from ... 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	15 years ago
David Vossel	513c680b8c	Adds pass-through support for codec CELT. ... 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	15 years ago
Terry Wilson	f0c8b18c18	Use older functions out of deference to older distros ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@326750 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Terry Wilson	efd040cd11	Replace Berkeley DB with SQLite 3 ... 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	15 years ago
Mark Murawki	8b20d4ffe8	New feature: AMI Action FilterAdd ... 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	15 years ago
Matthew Jordan	67945ce627	Merged revisions 326209 via svnmerge from ... 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	15 years ago
David Vossel	1339a0a535	Video support for ConfBridge. ... Review: https://reviewboard.asterisk.org/r/1288/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@325931 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Matthew Nicholson	82d28452ca	copy all flags on asterisk frames instead of just the timing flag ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@325815 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Matthew Nicholson	1da3304813	Merged revisions 325545 via svnmerge from ... 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	15 years ago
Matthew Nicholson	6c7d437287	Merged revisions 325537 via svnmerge from ... 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	15 years ago
Tilghman Lesher	db15b0010c	Merged revisions 324955 via svnmerge from ... 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	15 years ago
David Vossel	d5ea9e5ae2	Merged revisions 324652 via svnmerge from ... 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	15 years ago
Terry Wilson	385b8c6f8b	Merged revisions 324484 via svnmerge from ... 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	15 years ago
David Vossel	09a359449e	Merged revisions 324364 via svnmerge from ... 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	15 years ago
Leif Madsen	3d6c1ccd91	Merged revisions 324178 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r324178 | lmadsen | 2011-06-17 14:51:16 -0400 (Fri, 17 Jun 2011) | 2 lines Add Username and Secret fields to manager Login action. Pointed out by Vlad Povorozniuc ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@324179 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Leif Madsen	71e4b2a5d1	Merged revisions 324115 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r324115 | lmadsen | 2011-06-17 11:14:54 -0400 (Fri, 17 Jun 2011) | 3 lines Fix grammar in documentation for Goto() and GotoIf() (closes issue ASTERISK-18023) Reported by: Tim Osman ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@324131 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Terry Wilson	34e2305ae7	Merged revisions 324048 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r324048 | twilson | 2011-06-16 17:35:41 -0500 (Thu, 16 Jun 2011) | 8 lines Lock the channel before calling the setoption callback The channel needs to be locked before calling these callback functions. Also, sip_setoption needs to lock the pvt and a check p->rtp is non-null before using it. Review: https://reviewboard.asterisk.org/r/1220/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@324050 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Terry Wilson	c33e1b0e27	Merged revisions 323754 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r323754 | twilson | 2011-06-15 13:21:52 -0500 (Wed, 15 Jun 2011) | 23 lines Merged revisions 323733 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r323733 | twilson | 2011-06-15 13:13:00 -0500 (Wed, 15 Jun 2011) | 16 lines Merged revisions 323732 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r323732 | twilson | 2011-06-15 13:06:24 -0500 (Wed, 15 Jun 2011) | 9 lines Fix DYNAMIC_FEATURES DYNAMIC_FEATURES were broken by a recent DTMF change. This patch makes sure that dynamic features are also checked when deciding whether or not to pass DTMF through or store it for interpreting. (closes issue ASTERISK-17914) Reported by: vrban ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@323760 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Richard Mudgett	b2d0ea5fea	Merged revisions 323669-323670 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r323669 | rmudgett | 2011-06-15 11:43:18 -0500 (Wed, 15 Jun 2011) | 21 lines [regression] Voicemail MWI is no longer sent. When leaving a voicemail, the MWI message is never sent. The same thing happens when checking a voicemail and marking it as read. If you restart Asterisk, everything comes up at that state correctly, but changes to the messages in voicemail causes the light to not be set appropriately. Very easy to reproduce. * Made ast_event_check_subscriber() return TRUE if there are ANY subscribers to an event type when there are no restricting ie values passed. This allows an event being queued to be queued. (closes issue ASTERISK-18002) Reported by: lmadsen Tested by: lmadsen, irroot Patches: jira_asterisk_18002_v1.8.patch uploaded by rmudgett (License #5621) (closes issue ASTERISK-18019) ........ r323670 | rmudgett | 2011-06-15 11:43:31 -0500 (Wed, 15 Jun 2011) | 7 lines Add a test to the event unit tests to catch ASTERISK-18002. The new tests check to see if there are ANY subscribers to the event type when ast_event_check_subscriber() is not passed any specific ie values. (issue ASTERISK-18002) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@323671 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Sean Bright	affae67cd2	Merged revisions 323608 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r323608 | seanbright | 2011-06-15 11:31:53 -0400 (Wed, 15 Jun 2011) | 39 lines Merged revisions 323579 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r323579 | seanbright | 2011-06-15 11:22:50 -0400 (Wed, 15 Jun 2011) | 32 lines Merged revisions 323559 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r323559 | seanbright | 2011-06-15 11:15:30 -0400 (Wed, 15 Jun 2011) | 25 lines Resolve a segfault/bus error when we try to map memory that falls on a page boundary. The fix for ASTERISK-15359 was incorrect in that it added 1 to the length of the mmap'd region. The problem with this is that reading/writing to that extra byte outside of the bounds of the underlying fd causes a bus error. The real issue is that we are working with both a FILE * and the raw fd underneath it and not synchronizing between them. The code that was removed in ASTERISK-15359 was correct, but we weren't flushing the FILE * before mapping the fd. Looking at the manager code in 1.4 reveals that the FILE * in 'struct mansession' is never used except to create a temporary file that we immediately fdopen. This means we just need to write a 0 byte to the fd and everything will just work. The other branches require a call to fflush() which, while not a guaranteed fix, should reduce the likelihood of a crash. This all makes sense in my head. (closes issue ASTERISK-16460) Reported by: Ravelomanantsoa Hoby (hoby) Patches: issue17747_1.4_svn_markII.patch uploaded by Sean Bright (license #5060) ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@323609 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Richard Mudgett	70d9527951	Merged revisions 323456 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r323456 | rmudgett | 2011-06-14 19:50:20 -0500 (Tue, 14 Jun 2011) | 1 line Add missing break in ast_event_get_cached(). ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@323457 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Richard Mudgett	9ff8844c7f	Merged revisions 323392,323394 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r323392 | rmudgett | 2011-06-14 12:21:24 -0500 (Tue, 14 Jun 2011) | 6 lines Add more strict hostname checking to ast_dnsmgr_lookup(). Change suggested in review. Review: https://reviewboard.asterisk.org/r/1240/ ........ r323394 | rmudgett | 2011-06-14 12:21:39 -0500 (Tue, 14 Jun 2011) | 2 lines Made ast_sockaddr_split_hostport() port warning msgs more meaningful. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@323397 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Terry Wilson	abd7ef817e	Merged revisions 323370 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r323370 | twilson | 2011-06-14 09:33:55 -0700 (Tue, 14 Jun 2011) | 10 lines Add rtpkeepalives back to 1.8 The RTP-engine conversion left out support for handling rtpkeepalives. This patch adds them back. (closes issue ASTERISK-17304) Reported by: lmadsen Review: https://reviewboard.asterisk.org/r/1226/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@323374 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Leif Madsen	dafa8a659b	Merged revisions 323213 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r323213 | lmadsen | 2011-06-13 15:51:52 -0400 (Mon, 13 Jun 2011) | 6 lines Avoid dividing by zero with L() option to Dial() Reported by: nicolasom Patches: issue-17995.patch - nicolasom (License #5994) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@323214 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Terry Wilson	58ca560291	Merged revisions 322981 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r322981 | twilson | 2011-06-10 08:29:00 -0700 (Fri, 10 Jun 2011) | 11 lines Avoid a DB1 infinite loop bug Explicity check the last entry in the DB and make sure that we don't iterate past it. Since there can be no duplicates, this just makes sure that we stop after matching the last key. This patch also refactors the code to get away from some code duplication. A previous patch added many astdb tests and this patch passed them. Review: https://reviewboard.asterisk.org/r/1259/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@322982 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Richard Mudgett	0a8f9d2cf0	Merged revisions 322749 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r322749 | rmudgett | 2011-06-09 11:31:53 -0500 (Thu, 09 Jun 2011) | 15 lines Remove potential deadlock in call pickup race. Deadlock is possible in ast_do_pickup() when holding the target channel lock and trying to get the chan channel lock. Also, holding the target lock when calling ast_channel_masquerade() is not a good idea because that routine does deadlock avoidance. * Removed the need to hold the target lock after marking the target with a datastore and getting the connected line data off of the target channel. * Moved can_pickup() to ast_can_pickup() in features.c. Now all the call pickup methods use the same basic call pickup availability check. Review: https://reviewboard.asterisk.org/r/1234/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@322750 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Richard Mudgett	4b773e2ed9	Merged revisions 322425 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r322425 | rmudgett | 2011-06-08 13:46:30 -0500 (Wed, 08 Jun 2011) | 16 lines SRV lookup attempted for SIP peers listed as an IP address. Asterisk attempts to SRV lookup a host name even if the host name is an IP address. Regression introduced when IPv6 support was added. * Restored the check in ast_dnsmgr_lookup() to see if the given host name is an IP address. The IP address could be in either IPv4 or IPv6 formats. (closes issue ASTERISK-17815) Reported by: Byron Clark Tested by: Byron Clark, Richard Mudgett Patches: issue19248_v1.8.patch - uploaded by Richard Mudgett (License #5621) Review: https://reviewboard.asterisk.org/r/1240/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@322426 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Jonathan Rose	4ab3825fe4	Merged revisions 322069 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r322069 | jrose | 2011-06-06 14:07:56 -0500 (Mon, 06 Jun 2011) | 8 lines Fixes level toggling for logger set levels since it was reversed (closes issue ASTERISK-17850) Reported by: Luke H Tested by: jrose, Luke H Review: https://reviewboard.asterisk.org/r/1244/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@322070 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago