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2009-03-17Tweak the handling of the frame list inside of ast_answer().Russell Bryant
This does not change any behavior, but moves the frames from the local frame list back to the channel read queue using an O(n) algorithm instead of O(n^2). git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@182553 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-03-17correct logic flaw in ast_answer() changes in r182525Kevin P. Fleming
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@182530 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-03-17Improve behavior of ast_answer() to not lose incoming framesKevin P. Fleming
ast_answer(), when supplied a delay before returning to the caller, use ast_safe_sleep() to implement the delay. Unfortunately during this time any incoming frames are discarded, which is problematic for T.38 re-INVITES and other sorts of channel operations. When a delay is not passed to ast_answer(), it still delays for up to 500 milliseconds, waiting for media to arrive. Again, though, it discards any control frames, or non-voice media frames. This patch rectifies this situation, by storing all incoming frames during the delay period on a list, and then requeuing them onto the channel before returning to the caller. http://reviewboard.digium.com/r/196/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@182525 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-03-17Merged revisions 182449 via svnmerge from Tilghman Lesher
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r182449 | tilghman | 2009-03-17 00:50:52 -0500 (Tue, 17 Mar 2009) | 7 lines Fix race in astdb The underlying db1 implementation does not fully isolate the pages retrieved from astdb, so the lock protecting accesses needs to be extended until the copy from the shared memory structure is done. (closes issue #14682) Reported by: makoto ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@182450 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-03-16Fix a memory leak in the ast_answer / __ast_answer API call.Joshua Colp
For a channel that is not yet answered this API call will wait until a voice frame is received on the channel before returning. It does this by waiting for frames on the channel and reading them in. The frames read in were not freed when they should have been. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@182171 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-03-13Remove ast_ prefix from functions which are not public.Mark Michelson
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@182071 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-03-13Merged revisions 181990 via svnmerge from Mark Michelson
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r181990 | mmichelson | 2009-03-13 12:12:32 -0500 (Fri, 13 Mar 2009) | 35 lines Check the DYNAMIC_FEATURES of both the chan and peer when interpreting DTMF. Dynamic features defined in the applicationmap section of features.conf allow one to specify whether the caller, callee, or both have the ability to use the feature. The documentation in the features.conf.sample file could be interpreted to mean that one only needs to set the DYNAMIC_FEATURES channel variable on the calling channel in order to allow for the callee to be able to use the features which he should have permission to use. However, the DYNAMIC_FEATURES variable would only be read from the channel of the participant that pressed the DTMF sequence to activate the feature. The result of this was that the callee was unable to use dynamic features unless the dialplan writer had taken measures to be sure that the DYNAMIC_FEATURES variable was set on the callee's channel. This commit changes the behavior of ast_feature_interpret to concatenate the values of DYNAMIC_FEATURES from both parties involved in the bridge. The features themselves determine who has permission to use them, so there is no reason to believe that one side of the bridge could gain the ability to perform an action that they should not have the ability to perform. Kevin Fleming pointed out on the asterisk-users list that the typical way that this was worked around in the past was by setting _DYNAMIC_FEATURES on the calling channel so that the value would be inherited by the called channel. While this works, the documentation alone is not enough to figure out why this is necessary for the callee to be able to use dynamic features. In this particular case, changing the code to match the documentation is safe, easy, and will generally make things easier for people for future installations. This bug was originally reported on the asterisk-users list by David Ruggles. (closes issue #14657) Reported by: mmichelson Patches: 14657.patch uploaded by mmichelson (license 60) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@182029 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-03-12Adjust translation table column widths based upon the translation times.Tilghman Lesher
Previously, only 5 columns were displayed, and if a translation time exceeded 99,999 useconds, it would be displayed as 0, instead of its actual time. (closes issue #14532) Reported by: pj Patches: 20090311__bug14532.diff.txt uploaded by tilghman (license 14) Tested by: pj git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@181731 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-03-11Make handling of the BRIDGE_PLAY_SOUND variable thread-safe.Russell Bryant
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@181465 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-03-11Make handling of the BRIDGEPVTCALLID variable thread-safe.Russell Bryant
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@181428 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-03-11Merged revisions 181423 via svnmerge from Russell Bryant
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r181423 | russell | 2009-03-11 16:42:58 -0500 (Wed, 11 Mar 2009) | 9 lines Make code that updates BRIDGEPEER variable thread-safe. It is not safe to read the name field of an ast_channel without the channel locked. This patch fixes some places in channel.c where this was being done, and lead to crashes related to masquerades. (closes issue #14623) Reported by: guillecabeza ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@181424 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-03-11Fix malloc debug macros to work properly with h323.Jeff Peeler
The main problem here was that cstdlib was undefining free thereby causing the proper debug macros to not be used. ast_h323.cxx has been changed to call ast_free instead to avoid the issue. A few other issues were addressed: - There were a few instances of functions improperly passing ast_free instead of ast_free_ptr. - Some clean up was done to avoid the debug macros intentionally being redefined. (copied below from Kevin's commit, appreciate the help) - disable astmm.h from doing anything when STANDALONE is defined, which is used by the tools in the utils/ directory that use parts of Asterisk header files in hackish ways; also ensure that utils/extconf.c and utils/conf2ael.c are compiled with STANDALONE defined. (closes issue #13593) Reported by: pj git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@181135 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-03-11Add MALLOC_DEBUG to various utility APIs, so that memory leaks can be ↵Tilghman Lesher
tracked back to their source. (related to issue #14636) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@181028 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-03-11Spacing changes onlyTilghman Lesher
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@181027 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-03-10Reset the thread local string buffer when handling the UserEvent action.Joshua Colp
(closes issue #14593) Reported by: JimDickenson git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@180800 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-03-09Add Doxygen documentation for API changes from 1.6.0 to 1.6.1Jeff Peeler
Copied from my review board description: This is a continuation of the API changes documentation started for describing changes between releases. Most of the API changes were pretty simple needing only to be brought to attention via the new "Asterisk API Changes" list. However, if you see anything that needs further explanation feel free to supplement what is there. The current method of documenting is to add (in the header file): \version <ver number> <description of changes> and then to add the function to the change list in doxyref.h on the AstAPIChanges page. I also made sure all the functions that were newly added were tagged with \since 1.6.1. I think this is a good habit to start both for the historical aspect as well as for the future ability to easily add a "New Asterisk API" page. Review: http://reviewboard.digium.com/r/190/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@180719 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-03-06Merged revisions 180532 via svnmerge from David Vossel
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r180532 | dvossel | 2009-03-06 11:19:55 -0600 (Fri, 06 Mar 2009) | 9 lines Fix handling of backreferences for ENUM lookups enum.c did not handle regex backtraces correctly. The '\1' in the regex is a backreference that requires a pattern match to be inserted. The way the code used to work is that it would find the backreference and insert the entire input string minus the '+'. This is incorrect. The regexec() function takes in a variable called pmatch which is an array of structs containing the start and end indexes for each backreference substring. The original code actually passed the pmatch array pointer into regexec but never did anything with it. Now when a backtrace is found, the backtrace number is looked up in the pmatch array and the correct substring is inserted. (closes issue #14576) Reported by: chris-mac Review: http://reviewboard.digium.com/r/187/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@180534 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-03-05Merged revisions 180372 via svnmerge from Kevin P. Fleming
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r180372 | kpfleming | 2009-03-05 12:22:16 -0600 (Thu, 05 Mar 2009) | 9 lines Fix problems when RTP packet frame size is changed During some code analysis, I found that calling ast_rtp_codec_setpref() on an ast_rtp session does not work as expected; it does not adjust the smoother that may on the RTP session, in fact it summarily drops it, even if it has data in it, even if the current format's framing size has not changed. This is not good. This patch changes this behavior, so that if the packetization size for the current format changes, any existing smoother is safely updated to use the new size, and if no smoother was present, one is created. A new API call for smoothers, ast_smoother_reconfigure(), was required to implement these changes. Review: http://reviewboard.digium.com/r/184/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@180373 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-03-05Merge phase 1 support for the new bridging architecture.Joshua Colp
This commit brings in the bridging core, bridging technologies, and the ConfBridge application. For usage information on the ConfBridge application please see the output of "core show application ConfBridge" from the CLI. For API documentation please see the doxygen page describing the architecture and the documentation for each API call. Review: http://reviewboard.digium.com/r/93/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@180369 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-03-04Spacing changes onlyTilghman Lesher
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@180259 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-03-04Merged revisions 180194 via svnmerge from Joshua Colp
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r180194 | file | 2009-03-04 15:22:50 -0400 (Wed, 04 Mar 2009) | 4 lines Look for the number in a callerid string starting from the end. This way a value using <> can exist in the name portion. (issue #AST-194) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@180195 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-03-03app_read does not break from prompt loop with user terminated empty stringDavid Vossel
In app.c, ast_app_getdata is called to stream the prompts and receive DTMF input. If ast_app_getdata() receives an empty string caused by the user inputing the end of string character, in this case '#', it should break from the prompt loop and return to app_read, but instead it cycles through all the prompts. I've added a return value for this special case in ast_readstring() which uses an enum I've delcared in apps.h. This enum is now used as a return value for ast_app_getdata(). (closes issue #14279) Reported by: Marquis Patches: fix_app_read.patch uploaded by Marquis (license 32) read-ampersanmd.patch2 uploaded by dvossel (license 671) Tested by: Marquis, dvossel Review: http://reviewboard.digium.com/r/177/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@180032 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-03-03Merged revisions 179807 via svnmerge from Steve Murphy
https://origsvn.digium.com/svn/asterisk/branches/1.4 I had some work to do to port these changes to trunk; the check_expr stuff hasn't been updated here for quite some time, it appears. I added some more tests to the check_expr2 suite. I had to play around with the makefile a bit, etc. I added STANDALONE2 #ifdefs to ast_expr2.y so as not to conflict structure with aelparse. ........ r179807 | murf | 2009-03-03 11:11:34 -0700 (Tue, 03 Mar 2009) | 19 lines These changes allow AEL to better check ${} constructs within $[...], that are concatenated with text. I modified and added rules in ast_expr2.fl to better handle the concatenations. I added some default routines to ast_expr2.y so the standalone would compile. It also looks like I haven't run this thru bison since 2.1, so it's good to get this updated. The Makefile has comments added now for check_expr2 and check_expr to explain what they are for, and how to run them. The testexpr2s stuff has been removed, in favor of check_expr2. expr2.testinput has been updated to include the two expressions that inspired these changes (from mcnobody on #asterisk this morning) The regression has been run and all looks well. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@179973 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-03-03Merged revisions 179840 via svnmerge from Joshua Colp
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r179840 | file | 2009-03-03 14:27:09 -0400 (Tue, 03 Mar 2009) | 9 lines Do not assume that the bridge_cdr is still attached to the channel when the 'h' exten is finished executing. It is possible for a masquerade operation to occur when the 'h' exten is operating. This operation moves the CDR records around causing the bridge_cdr to no longer exist on the channel where it is expected to. We can not safely modify it afterwards because of this, so don't even try. (closes issue #14564) Reported by: meric ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@179841 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-03-03Merged revisions 179741 via svnmerge from Russell Bryant
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r179741 | russell | 2009-03-03 10:45:46 -0600 (Tue, 03 Mar 2009) | 6 lines Ensure chan->fdno always gets reset to -1 after handling a channel fd event. Since setting fdno to -1 had to be moved, a couple of other code paths that do process an fd event return early and do not pass through the code path where it was moved to. So, set it to -1 in a few other places, too. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@179742 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-03-03Merged revisions 179671 via svnmerge from Joshua Colp
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r179671 | file | 2009-03-03 10:38:09 -0400 (Tue, 03 Mar 2009) | 3 lines Move where fdno is set to the default value to *after* the read callback of the channel driver is called. We have to do this as the underlying channel driver may need the fdno value to determine what to read. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@179672 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-03-03Merged revisions 179608 via svnmerge from Russell Bryant
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r179608 | russell | 2009-03-03 07:53:52 -0600 (Tue, 03 Mar 2009) | 9 lines Make it easier to detect an improper call to ast_read(). When you call ast_waitfor() on a channel, the index into the channel fds array that holds the file descriptor that poll() determines has input available is stored in fdno. This patch clears out this value after a call to ast_read() and also reports errors if ast_read() is called without an fdno set. From a discussion on the asterisk-dev list. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@179609 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-03-03Merged revisions 179536 via svnmerge from Jeff Peeler
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r179536 | jpeeler | 2009-03-02 17:54:39 -0600 (Mon, 02 Mar 2009) | 15 lines Fix bridging regression from commit 176701 This fixes a bad regression where the bridge would exit after an attended transfer was made. The problem was due to nexteventts getting set after the masquerade which caused the bridge to return AST_BRIDGE_COMPLETE. (closes issue #14315) Reported by: tim_ringenbach ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@179537 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-03-02Merged revisions 179468 via svnmerge from Tilghman Lesher
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r179468 | tilghman | 2009-03-02 17:09:01 -0600 (Mon, 02 Mar 2009) | 10 lines When ending a recording with silence detection, remember to reduce the duration. The end of the recording is correspondingly trimmed, but the duration was not trimmed by the number of seconds trimmed, so the saved duration was necessarily longer than the actual soundfile duration. (closes issue #14406) Reported by: sasargen Patches: 20090226__bug14406.diff.txt uploaded by tilghman (license 14) Tested by: sasargen ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@179469 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-03-02Merged revisions 179461 via svnmerge from Russell Bryant
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r179461 | russell | 2009-03-02 16:58:18 -0600 (Mon, 02 Mar 2009) | 8 lines Ensure that only one thread is calling ast_settimeout() on a channel at a time. For example, with an IAX2 channel, you can have both the channel thread and the chan_iax2 processing threads calling this function, and doing so twice at the same time is a bad thing. (Found in a debugging session with dvossel and mmichelson) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@179462 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-03-02Merged revisions 179395 via svnmerge from Jason Parker
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r179395 | qwell | 2009-03-02 14:14:57 -0600 (Mon, 02 Mar 2009) | 1 line Remove several silly warnings in editline. One about a broken preprocessor directive, and another about strlcpy/strlcat. (closes issue #14264) Reported by: dimas ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@179396 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-03-02Fix issue where changing the volume of both directions of audio did not work.Joshua Colp
(closes issue #14574) Reported by: KNK Patches: audiohook_volume_fix.diff uploaded by KNK (license 545) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@179291 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-27Merged revisions 178956 via svnmerge from Steve Murphy
https://origsvn.digium.com/svn/asterisk/branches/1.4 In this case, it's just a matter of reducing the default timeouts from 2000 to 1000 msec, as the max def feature digit timeout is no longer halved. ........ r178956 | murf | 2009-02-26 14:27:32 -0700 (Thu, 26 Feb 2009) | 18 lines This change moves the default feature digit timeout to 1000 ms from the previous default of 500. As per bug 14515, a dev discussion arrived at a "mediated concensus" of a default feature digit timeout of 1.0 sec. Some voted for 1300; ctooley thought 1500 for distracted phone users in phone booths; kpfleming put his foot down at 1.0 sec. Users who found the previous default max delay of 250 msec perfect, are welcome to override the new default. Notice that I said that 250 msec was the default; wait a minute, you might say, the config file said it was 500 msec!; well, because of the bug fix for 14515, we found that 500 msec was actually enforcing a max of 250. The bug fix would restore 500 msec, but we felt even that was a bit tight for most users... 2000 msec was pushed earlier by mmichelson, so that reduces to 1000 msec after the bug fix. Enjoy! ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@178986 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-26Sound confirmation of call pickup success.Tilghman Lesher
(closes issue #13826) Reported by: azielke Patches: pickupsound2-trunk.patch uploaded by azielke (license 548) __20081124_bug_13826_updated.patch uploaded by lmadsen (license 10) Tested by: lmadsen git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@178919 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-26Merged revisions 178804 via svnmerge from Steve Murphy
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r178804 | murf | 2009-02-26 10:09:03 -0700 (Thu, 26 Feb 2009) | 28 lines This patch prevents the feature detection timeout from being cut in half. Because the ast_channel_bridge() call will return 0 and pass a frame pointer for both DTMF_BEGIN and DTMF_END, the feature_timer field in hte config struct is getting decremented twice, which effectively cuts the digittimeout in half. I added conditions to the if statement to only let DTMF_END frames to flow thru, which solved the problem. Also, when the frame pointer is null, let control flow thru-- this usually happens on timeouts. I added a comment to the code to explain what's going on and why. Many thanks to sodom for reporting this problem. Personnally, it always seemed like something was wrong with the featuredigittimeout, but I never could quite decide what... and was too busy to investigate. This bug forced the issue, and now we know. Sodom had other issues in 14515, but I couldn't reproduce them. If he still has problems, and wants to get them solved, he is welcome to reopen 14515. (closes issue #14515) Reported by: sodom Patches: 14515.patch uploaded by murf (license 17) Tested by: murf, sodom ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@178828 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-26Fix an issue where the timer for file playback would not be stopped if DAHDI ↵Joshua Colp
was not installed. (closes issue #14541) Reported by: grant git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@178801 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-26Ensure there is a valid tone part before trying to play tones.Joshua Colp
(closes issue #14558) Reported by: alecdavis git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@178764 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-25Picky, picky buildbotsTilghman Lesher
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@178607 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-25Use notification when timezone files change and re-scan then.Tilghman Lesher
(closes issue #14300) Reported by: jamessan Patches: 20090127__bug14300.diff.txt uploaded by tilghman (license 14) 20090224__bug14300.diff uploaded by jamessan (license 246) Tested by: jamessan Review: http://reviewboard.digium.com/r/136/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@178605 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-25Merged revisions 178508 via svnmerge from Russell Bryant
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r178508 | russell | 2009-02-25 06:43:36 -0600 (Wed, 25 Feb 2009) | 2 lines Update the copyright year for the main page of the doxygen documentation. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@178509 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-24Apparently, a void cast doesn't override warn_unused_result.Tilghman Lesher
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@178381 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-24The 3 possible errors with pipe(2) are all impossible in this situation.Tilghman Lesher
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@178375 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-24Merged revisions 178373 via svnmerge from Russell Bryant
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r178373 | russell | 2009-02-24 14:36:19 -0600 (Tue, 24 Feb 2009) | 6 lines Only set dtmfcount on BEGIN, and ensure it gets reset to 0 properly. (issue #14460) Reported by: moliveras Tested by: russell ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@178374 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-24Use a SIGPIPE to kill the process, instead of depending upon the astcanary ↵Tilghman Lesher
process being inherited by init. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@178342 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-23Merged revisions 178141 via svnmerge from Russell Bryant
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r178141 | russell | 2009-02-23 17:09:01 -0600 (Mon, 23 Feb 2009) | 14 lines Fix infinite DTMF when a BEGIN is received without an END. This commit is related to rev 175124 of 1.4 where a previous attempt was made to fix this problem. The problem with the previous patch was that the inserted code needed to go _before_ setting the lastrxts to the current timestamp. Because those were the same, the dtmfcount variable was never decremented, and so the END was never sent. In passing, I removed the dtmfsamples variable which was completed unused. I also removed a redundant setting of the lastrxts variable. (closes issue #14460) Reported by: moliveras ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@178142 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-23Fix a regression in scheduler entry ordering, and add a regression test for it.Russell Bryant
(closes issue #14522) Reported by: pj Tested by: russell git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@178022 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-21add extra check for sysinfo/sysctlMichiel van Baak
(closes issue #14513) Reported by: snuffy Patches: bug14513_fixsysinfo.diff uploaded by snuffy (license 35) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@177913 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-21Trailing whitespace, minor coding guideline fixes, and start beefing up theSean Bright
hashtab documentation a bit. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@177884 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-20Merged revisions 177786 via svnmerge from Tilghman Lesher
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r177786 | tilghman | 2009-02-20 16:59:52 -0600 (Fri, 20 Feb 2009) | 9 lines Don't print the CR-NL combination when we aren't outputting to the manager. An embedded CR-NL in a CLI command screws up several AMI parsers that don't expect to see that combination in the middle of output. (Closes issue #14305) Reported by: martins Patch by: tilghman ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@177787 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-02-20Allow semicolons to be escaped, when passing arguments to the System command.Tilghman Lesher
(closes issue #14231) Reported by: jcovert Patches: 20090113__bug14231__2.diff.txt uploaded by Corydon76 (license 14) corrected_20090113__bug14231__2.diff.txt uploaded by jcovert (license 551) Tested by: jcovert git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@177664 65c4cc65-6c06-0410-ace0-fbb531ad65f3
generated by cgit v1.2.3 (git 2.39.1) at 2024-07-03 02:14:33 +0000