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2009-03-11Merged revisions 181428 via svnmerge from russell1-8/+17
https://origsvn.digium.com/svn/asterisk/trunk ........ r181428 | russell | 2009-03-11 17:14:55 -0500 (Wed, 11 Mar 2009) | 2 lines Make handling of the BRIDGEPVTCALLID variable thread-safe. ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@181430 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-11Merged revisions 181424 via svnmerge from russell1-14/+31
https://origsvn.digium.com/svn/asterisk/trunk ................ r181424 | russell | 2009-03-11 16:49:29 -0500 (Wed, 11 Mar 2009) | 17 lines Merged revisions 181423 via svnmerge from 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: http://svn.digium.com/svn/asterisk/branches/1.6.1@181426 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-11Merged revisions 181135 via svnmerge from jpeeler1-1/+1
https://origsvn.digium.com/svn/asterisk/trunk ........ r181135 | jpeeler | 2009-03-10 23:06:44 -0500 (Tue, 10 Mar 2009) | 20 lines Fix malloc debug macros to work properly with h323. 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: http://svn.digium.com/svn/asterisk/branches/1.6.1@181199 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-10Merged revisions 180800 via svnmerge from file1-0/+3
https://origsvn.digium.com/svn/asterisk/trunk ........ r180800 | file | 2009-03-10 11:40:38 -0300 (Tue, 10 Mar 2009) | 5 lines Reset the thread local string buffer when handling the UserEvent action. (closes issue #14593) Reported by: JimDickenson ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@180802 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-09Merged revisions 180719 via svnmerge from jpeeler3-18/+0
https://origsvn.digium.com/svn/asterisk/trunk ........ r180719 | jpeeler | 2009-03-09 15:58:17 -0500 (Mon, 09 Mar 2009) | 16 lines Add Doxygen documentation for API changes from 1.6.0 to 1.6.1 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: http://svn.digium.com/svn/asterisk/branches/1.6.1@180740 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-06Merged revisions 180534 via svnmerge from dvossel1-34/+47
https://origsvn.digium.com/svn/asterisk/trunk ................ r180534 | dvossel | 2009-03-06 11:26:38 -0600 (Fri, 06 Mar 2009) | 15 lines Merged revisions 180532 via svnmerge from 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: http://svn.digium.com/svn/asterisk/branches/1.6.1@180537 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-05Merged revisions 180373 via svnmerge from kpfleming2-36/+103
https://origsvn.digium.com/svn/asterisk/trunk ................ r180373 | kpfleming | 2009-03-05 12:29:38 -0600 (Thu, 05 Mar 2009) | 15 lines Merged revisions 180372 via svnmerge from 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: http://svn.digium.com/svn/asterisk/branches/1.6.1@180378 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-04Merged revisions 180195 via svnmerge from file1-1/+1
https://origsvn.digium.com/svn/asterisk/trunk ................ r180195 | file | 2009-03-04 15:24:59 -0400 (Wed, 04 Mar 2009) | 11 lines Merged revisions 180194 via svnmerge from 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: http://svn.digium.com/svn/asterisk/branches/1.6.1@180197 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-03Merged revisions 180032 via svnmerge from dvossel2-11/+20
https://origsvn.digium.com/svn/asterisk/trunk ........ r180032 | dvossel | 2009-03-03 17:21:18 -0600 (Tue, 03 Mar 2009) | 14 lines app_read does not break from prompt loop with user terminated empty string 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: http://svn.digium.com/svn/asterisk/branches/1.6.1@180080 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-03Merged revisions 179973 via svnmerge from murf5-184/+312
https://origsvn.digium.com/svn/asterisk/trunk ................ r179973 | murf | 2009-03-03 15:12:02 -0700 (Tue, 03 Mar 2009) | 33 lines Merged revisions 179807 via svnmerge from 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: http://svn.digium.com/svn/asterisk/branches/1.6.1@180077 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-03Merged revisions 179841 via svnmerge from file1-2/+7
https://origsvn.digium.com/svn/asterisk/trunk ................ r179841 | file | 2009-03-03 14:28:46 -0400 (Tue, 03 Mar 2009) | 16 lines Merged revisions 179840 via svnmerge from 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: http://svn.digium.com/svn/asterisk/branches/1.6.1@179843 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-03Merged revisions 179742 via svnmerge from russell1-1/+4
https://origsvn.digium.com/svn/asterisk/trunk ................ r179742 | russell | 2009-03-03 10:47:28 -0600 (Tue, 03 Mar 2009) | 14 lines Merged revisions 179741 via svnmerge from 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: http://svn.digium.com/svn/asterisk/branches/1.6.1@179744 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-03Merged revisions 179672 via svnmerge from file1-6/+6
https://origsvn.digium.com/svn/asterisk/trunk ................ r179672 | file | 2009-03-03 10:40:04 -0400 (Tue, 03 Mar 2009) | 10 lines Merged revisions 179671 via svnmerge from 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: http://svn.digium.com/svn/asterisk/branches/1.6.1@179674 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-03Merged revisions 179609 via svnmerge from russell1-0/+12
https://origsvn.digium.com/svn/asterisk/trunk ................ r179609 | russell | 2009-03-03 07:54:41 -0600 (Tue, 03 Mar 2009) | 17 lines Merged revisions 179608 via svnmerge from 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: http://svn.digium.com/svn/asterisk/branches/1.6.1@179611 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-03Merged revisions 179537 via svnmerge from jpeeler1-1/+1
https://origsvn.digium.com/svn/asterisk/trunk ................ r179537 | jpeeler | 2009-03-02 18:01:51 -0600 (Mon, 02 Mar 2009) | 21 lines Merged revisions 179536 via svnmerge from 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: http://svn.digium.com/svn/asterisk/branches/1.6.1@179539 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-02Merged revisions 179469 via svnmerge from tilghman1-1/+9
https://origsvn.digium.com/svn/asterisk/trunk ................ r179469 | tilghman | 2009-03-02 17:10:18 -0600 (Mon, 02 Mar 2009) | 17 lines Merged revisions 179468 via svnmerge from 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: http://svn.digium.com/svn/asterisk/branches/1.6.1@179471 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-02Merged revisions 179462 via svnmerge from russell1-0/+5
https://origsvn.digium.com/svn/asterisk/trunk ................ r179462 | russell | 2009-03-02 17:00:30 -0600 (Mon, 02 Mar 2009) | 16 lines Merged revisions 179461 via svnmerge from 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: http://svn.digium.com/svn/asterisk/branches/1.6.1@179464 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-02Merged revisions 179396 via svnmerge from qwell4-5/+129
https://origsvn.digium.com/svn/asterisk/trunk ................ r179396 | qwell | 2009-03-02 14:16:51 -0600 (Mon, 02 Mar 2009) | 9 lines Merged revisions 179395 via svnmerge from 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: http://svn.digium.com/svn/asterisk/branches/1.6.1@179407 f38db490-d61c-443f-a65b-d21fe96a405b
2009-03-02Merged revisions 179291 via svnmerge from file1-2/+4
https://origsvn.digium.com/svn/asterisk/trunk ........ r179291 | file | 2009-03-02 10:13:45 -0400 (Mon, 02 Mar 2009) | 7 lines Fix issue where changing the volume of both directions of audio did not work. (closes issue #14574) Reported by: KNK Patches: audiohook_volume_fix.diff uploaded by KNK (license 545) ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@179293 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-27Merged revisions 178986 via svnmerge from murf1-1/+1
https://origsvn.digium.com/svn/asterisk/trunk ................ r178986 | murf | 2009-02-26 20:45:58 -0700 (Thu, 26 Feb 2009) | 26 lines Merged revisions 178956 via svnmerge from 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: http://svn.digium.com/svn/asterisk/branches/1.6.1@178988 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-26Merged revisions 178828 via svnmerge from murf1-2/+9
https://origsvn.digium.com/svn/asterisk/trunk ................ r178828 | murf | 2009-02-26 10:22:11 -0700 (Thu, 26 Feb 2009) | 34 lines Merged revisions 178804 via svnmerge from 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: http://svn.digium.com/svn/asterisk/branches/1.6.1@178869 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-26Merged revisions 178801 via svnmerge from file1-2/+0
https://origsvn.digium.com/svn/asterisk/trunk ........ r178801 | file | 2009-02-26 12:42:36 -0400 (Thu, 26 Feb 2009) | 5 lines Fix an issue where the timer for file playback would not be stopped if DAHDI was not installed. (closes issue #14541) Reported by: grant ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@178803 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-25Merged revisions 178509 via svnmerge from russell1-1/+1
https://origsvn.digium.com/svn/asterisk/trunk ................ r178509 | russell | 2009-02-25 06:45:30 -0600 (Wed, 25 Feb 2009) | 10 lines Merged revisions 178508 via svnmerge from 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: http://svn.digium.com/svn/asterisk/branches/1.6.1@178511 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-24Merged revisions 178381 via svnmerge from tilghman1-1/+4
https://origsvn.digium.com/svn/asterisk/trunk ........ r178381 | tilghman | 2009-02-24 14:52:44 -0600 (Tue, 24 Feb 2009) | 2 lines Apparently, a void cast doesn't override warn_unused_result. ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@178383 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-24Merged revisions 178374 via svnmerge from russell1-2/+4
https://origsvn.digium.com/svn/asterisk/trunk ................ r178374 | russell | 2009-02-24 14:39:57 -0600 (Tue, 24 Feb 2009) | 14 lines Merged revisions 178373 via svnmerge from 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: http://svn.digium.com/svn/asterisk/branches/1.6.1@178379 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-24Merged revisions 178375 via svnmerge from tilghman1-1/+1
https://origsvn.digium.com/svn/asterisk/trunk ........ r178375 | tilghman | 2009-02-24 14:40:02 -0600 (Tue, 24 Feb 2009) | 2 lines The 3 possible errors with pipe(2) are all impossible in this situation. ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@178377 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-24Merged revisions 178342 via svnmerge from tilghman1-1/+17
https://origsvn.digium.com/svn/asterisk/trunk ........ r178342 | tilghman | 2009-02-24 14:06:48 -0600 (Tue, 24 Feb 2009) | 2 lines Use a SIGPIPE to kill the process, instead of depending upon the astcanary process being inherited by init. ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@178344 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-23Merged revisions 178142 via svnmerge from russell1-6/+3
https://origsvn.digium.com/svn/asterisk/trunk ................ r178142 | russell | 2009-02-23 17:11:37 -0600 (Mon, 23 Feb 2009) | 22 lines Merged revisions 178141 via svnmerge from 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: http://svn.digium.com/svn/asterisk/branches/1.6.1@178172 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-20Merged revisions 177787 via svnmerge from tilghman1-11/+11
https://origsvn.digium.com/svn/asterisk/trunk ................ r177787 | tilghman | 2009-02-20 17:02:35 -0600 (Fri, 20 Feb 2009) | 16 lines Merged revisions 177786 via svnmerge from 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: http://svn.digium.com/svn/asterisk/branches/1.6.1@177789 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-20Merged revisions 177664 via svnmerge from tilghman1-0/+27
https://origsvn.digium.com/svn/asterisk/trunk ........ r177664 | tilghman | 2009-02-20 11:29:51 -0600 (Fri, 20 Feb 2009) | 8 lines Allow semicolons to be escaped, when passing arguments to the System command. (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: http://svn.digium.com/svn/asterisk/branches/1.6.1@177761 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-20Merged revisions 177595 via svnmerge from murf2-1940/+166
https://origsvn.digium.com/svn/asterisk/trunk ................ r177595 | murf | 2009-02-19 16:56:50 -0700 (Thu, 19 Feb 2009) | 32 lines Merged revisions 177540 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 Trunk was already pretty 8-bit clean; but I'm still removing the --full from the flex command so everything is uniform. ........ r177540 | murf | 2009-02-19 15:51:37 -0700 (Thu, 19 Feb 2009) | 21 lines This patch fixes a problem with 8-bit input to the ast_expr2 scanner. The real culprit was the --full argument to flex in the Makefile! This causes a 7-bit scanner to be generated. I reviewed the rules and found one rule where I needed to specifically include 8-bit chars for a token. I tested against the text supplied by ibercom, and all looks very well. This has been there a surprisingly long time! (closes issue #14498) Reported by: ibercom Patches: 14498.patch uploaded by murf (license 17) Tested by: murf ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@177623 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-19Merged revisions 177356 via svnmerge from jpeeler1-1/+0
https://origsvn.digium.com/svn/asterisk/trunk ........ r177356 | jpeeler | 2009-02-19 09:56:31 -0600 (Thu, 19 Feb 2009) | 4 lines Fix mismerge from revision 176708 pointed out by Kaloyan Kovachev on the asterisk-dev mailing list. Thanks! ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@177358 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-18Merged revisions 177229 via svnmerge from kpfleming1-2/+2
https://origsvn.digium.com/svn/asterisk/trunk ........ r177229 | kpfleming | 2009-02-18 17:09:58 -0600 (Wed, 18 Feb 2009) | 3 lines fix two very minor bugs: if anyone ever uses SLINEAR16 as a format in RTP, ensure that the samples are byte-swapped to network order if needed. also, when a smoother is operating on a format that has a sample rate other than 8000 samples per second, use the proper sample rate for computing delivery timestamps. ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@177230 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-18Merged revisions 177226 via svnmerge from dvossel1-30/+45
https://origsvn.digium.com/svn/asterisk/trunk ........ r177226 | dvossel | 2009-02-18 16:51:38 -0600 (Wed, 18 Feb 2009) | 9 lines Locking issue in action_bridge and bridge_exec action_bridge() and bridge_exec() both search for the channels to bridge to, and then immediately drop the lock. Instead, they should hold the lock until the masquerade is complete. This will guarantee the channel remains and prevent any other weirdness from occurring. In action_bridge() some more weirdness comes into play. Both channels are needlessly locked at the same time and perform the exact same logic. It makes sense from a coding organizational standpoint, but could cause a theoretical deadlock so I split the code up. There is an issue associated with this, but since its a rather complicated thing to reproduce I'm not certain this alone will close it. issue# 14296 Review: http://reviewboard.digium.com/r/167/ ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@177228 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-18Merged revisions 177035 via svnmerge from dbailey1-1/+1
https://origsvn.digium.com/svn/asterisk/trunk ........ r177035 | dbailey | 2009-02-18 11:24:07 -0600 (Wed, 18 Feb 2009) | 2 lines Fixed error where a check for an zero length, terminated string was needed. ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@177037 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-18Merged revisions 176948 via svnmerge from dbailey1-1/+1
https://origsvn.digium.com/svn/asterisk/trunk ........ r176948 | dbailey | 2009-02-18 10:09:12 -0600 (Wed, 18 Feb 2009) | 2 lines Need to take into account the \0 terminator of the old string to determine the amount available. ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@176962 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-18Merged revisions 176943 via svnmerge from murf1-17/+110
https://origsvn.digium.com/svn/asterisk/trunk ........ r176943 | murf | 2009-02-18 08:35:26 -0700 (Wed, 18 Feb 2009) | 45 lines This patch fixes merge_contexts_and_delete so it does not deadlock when hints are present. Reason: when I re-engineered the merge_and_delete func to reduce its lock time, I failed to notice that the functions it calls still also do locking as before. This leads to deadlocks on dialplan reloads, when there are actually living, subscribed hints registered in the system. While the reporter come across this problem while using AEL, I might note that these deadlocks should also happen if extensions.conf were used. Here I added these routines to pbx.c: ast_add_extension_nolock add_pri_lockopt ast_add_extension2_lockopt find_context add_hint_nolock All of the above routines are static and restricted to be used only within pbx.c, and more specifically within the merge_contexts_and_delete routine. They are pretty much the same as their counterparts except they don't lock contexts or hints. Most of them now do the real work of their name-alike, with optional locking via extra arguments, and are called by their name-alike. The goal was to have the original functions so they would behave exactly as before. Both PJ and I tested these fixes, and the deadlocking problem is no longer encountered. (closes issue #14357) Reported by: pj Patches: 14357.diff uploaded by murf (license 17) Tested by: pj, murf ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@176946 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-18Merged revisions 176901 via svnmerge from russell1-8/+15
https://origsvn.digium.com/svn/asterisk/trunk ........ r176901 | russell | 2009-02-18 00:00:40 -0600 (Wed, 18 Feb 2009) | 9 lines Fix a number of incorrect uses of strncpy(). The big problem here is that the 3rd argument provided in these uses of strncpy() did not reserve a byte for the null terminator, leaving the potential for writing one byte past the end of the buffer. Aside from this, there were coding guidelines violations with regards to spacing, as well as hard coded lengths being used instead of sizeof(). ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@176903 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-17Merged revisions 176708 via svnmerge from jpeeler2-14/+31
https://origsvn.digium.com/svn/asterisk/trunk ................ r176708 | jpeeler | 2009-02-17 16:08:00 -0600 (Tue, 17 Feb 2009) | 23 lines Merged revisions 176701 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r176701 | jpeeler | 2009-02-17 15:54:34 -0600 (Tue, 17 Feb 2009) | 17 lines Modify bridging to properly evaluate DTMF after first warning is played The main problem is currently if the Dial flag L is used with a warning sound, DTMF is not evaluated after the first warning sound. To fix this, a flag has been added in ast_generic_bridge for playing the warning which ensures that if a scheduled warning is missed, multiple warrnings are not played back (due to a feature evaluation or waiting for digits). ast_channel_bridge was modified to store the nexteventts in the ast_bridge_config structure as that information was lost every time ast_channel_bridge was reentered, causing a hangup due to incorrect time calculations. (closes issue #14315) Reported by: tim_ringenbach Reviewed on reviewboard: http://reviewboard.digium.com/r/163/ ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@176711 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-17Merged revisions 176666 via svnmerge from russell2-86/+120
https://origsvn.digium.com/svn/asterisk/trunk ........ r176666 | russell | 2009-02-17 15:22:40 -0600 (Tue, 17 Feb 2009) | 16 lines Update the timing API to have better support for multiple timing interfaces. 1) Add module use count handling so that timing modules can be unloaded. 2) Implement unload_module() functions for the timing interface modules. 3) Allow multiple timing modules to be loaded, and use the one with the highest priority value. 4) Report which timing module is being use in the "timing test" CLI command. (closes issue #14489) Reported by: russell Review: http://reviewboard.digium.com/r/162/ ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@176675 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-17Merged revisions 176632 via svnmerge from russell2-1/+285
https://origsvn.digium.com/svn/asterisk/trunk ........ r176632 | russell | 2009-02-17 14:51:10 -0600 (Tue, 17 Feb 2009) | 8 lines Add an implementation of the heap data structure. A heap is a convenient data structure for implementing a priority queue. Code from svn/asterisk/team/russell/heap/. Review: http://reviewboard.digium.com/r/160/ ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@176634 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-17Merged revisions 176557 via svnmerge from russell1-1/+1
https://origsvn.digium.com/svn/asterisk/trunk ........ r176557 | russell | 2009-02-17 11:33:38 -0600 (Tue, 17 Feb 2009) | 12 lines Fix a race condition that caused device states to become incorrect for hints. The problem here is that the hint processing code was subscribed to the wrong event type. So, it started processing state for a hint too soon, before the device state cache had been updated. Also, fix a similar bug in app_queue, as it was also subscribed to the wrong event type. (closes issue #14461) Reported by: alecdavis ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@176559 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-16Merged revisions 176255 via svnmerge from kpfleming1-18/+38
https://origsvn.digium.com/svn/asterisk/trunk ................ r176255 | kpfleming | 2009-02-16 15:45:54 -0600 (Mon, 16 Feb 2009) | 13 lines Merged revisions 176216 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r176216 | kpfleming | 2009-02-16 15:10:38 -0600 (Mon, 16 Feb 2009) | 3 lines fix a flaw in the ast_string_field_build() family of API calls; these functions made no attempt to reuse the space already allocated to a field, so every time the field was written it would allocate new space, leading to what appeared to be a memory leak. ........ r176254 | kpfleming | 2009-02-16 15:41:46 -0600 (Mon, 16 Feb 2009) | 3 lines correct a logic error in the last stringfields commit... don't mark additional space as allocated if the string was built using already-allocated space ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@176259 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-16Merged revisions 176174 via svnmerge from mmichelson1-2/+7
https://origsvn.digium.com/svn/asterisk/trunk ........ r176174 | mmichelson | 2009-02-16 12:25:57 -0600 (Mon, 16 Feb 2009) | 11 lines Assist proper thread synchronization when stopping the logger thread. I was finding that on my dev box, occasionally attempting to "stop now" in trunk would cause Asterisk to hang. I traced this to the fact that the logger thread was waiting on a condition which had already been signalled. The logger thread also need to be sure to check the value of the close_logger_thread variable. The close_logger_thread variable is only checked when the list of logmessages is empty. This allows for the logger thread to print and free any pending messages before exiting. ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@176176 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-15Merged revisions 175882 via svnmerge from russell1-2/+5
https://origsvn.digium.com/svn/asterisk/trunk ........ r175882 | russell | 2009-02-15 15:27:33 -0600 (Sun, 15 Feb 2009) | 2 lines Make ast_sched_report() and ast_sched_dump() thread safe. ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@175890 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-15Merged revisions 175829 via svnmerge from russell1-22/+15
https://origsvn.digium.com/svn/asterisk/trunk ........ r175829 | russell | 2009-02-15 14:56:27 -0600 (Sun, 15 Feb 2009) | 14 lines Fix a number of problems with ast_sched_report(). 1) It had numerous coding guidelines violations with regards to formatting. 2) It allocated memory using ast_calloc() that was never freed. 3) It didn't check for failure from the allocation. 4) It used sprintf() and strcat() to build the result, doing zero checking to prevent writing past the end of the provided buffer. The function also lacks API documentation, but that has not been addressed in this commit. ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@175831 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-12Merged revisions 175334 via svnmerge from tilghman1-23/+51
https://origsvn.digium.com/svn/asterisk/trunk ................ r175334 | tilghman | 2009-02-12 15:25:14 -0600 (Thu, 12 Feb 2009) | 16 lines Merged revisions 175311 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r175311 | tilghman | 2009-02-12 15:19:40 -0600 (Thu, 12 Feb 2009) | 9 lines Fix crashes when receiving certain T.38 packets. Also, increase the maximum size of T.38 packets and warn users when they try to set the limits above those maximums. (closes issue #13050) Reported by: schern Patches: 20090212__bug13050.diff.txt uploaded by Corydon76 (license 14) Tested by: schern ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@175342 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-12Merged revisions 175298 via svnmerge from jpeeler1-1/+8
https://origsvn.digium.com/svn/asterisk/trunk ................ r175298 | jpeeler | 2009-02-12 14:48:56 -0600 (Thu, 12 Feb 2009) | 15 lines Merged revisions 175294 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r175294 | jpeeler | 2009-02-12 14:34:36 -0600 (Thu, 12 Feb 2009) | 9 lines Fix ParkedCall event information for From field in the case of a blind transfer If the parker information can not be obtained from the peer, try and see if the BLINDTRANSFER channel variable has been set. Previously, a blind transfer to the ParkAndAnnounce app would return nothing for the From. Closes AST-189 ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@175300 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-12Merged revisions 175188 via svnmerge from jpeeler1-1/+2
https://origsvn.digium.com/svn/asterisk/trunk ................ r175188 | jpeeler | 2009-02-12 12:00:11 -0600 (Thu, 12 Feb 2009) | 12 lines Merged revisions 175187 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r175187 | jpeeler | 2009-02-12 11:57:10 -0600 (Thu, 12 Feb 2009) | 6 lines Fix crash in event of failed attempt to transfer to parking The peer may not necessarily exist, such as in the case of a transfer to ParkAndAnnounce. In this case don't try to play a sound to it. ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@175190 f38db490-d61c-443f-a65b-d21fe96a405b
2009-02-12Merged revisions 175125 via svnmerge from russell1-0/+15
https://origsvn.digium.com/svn/asterisk/trunk ................ r175125 | russell | 2009-02-12 10:57:25 -0600 (Thu, 12 Feb 2009) | 35 lines Merged revisions 175124 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r175124 | russell | 2009-02-12 10:51:13 -0600 (Thu, 12 Feb 2009) | 27 lines Don't send DTMF for infinite time if we do not receive an END event. I thought that this was going to end up being a pretty gnarly fix, but it turns out that there was actually already a configuration option in rtp.conf, dtmftimeout, that was intended to handle this situation. However, in between Asterisk 1.2 and Asterisk 1.4, the code that processed the option got lost. So, this commit brings it back to life. The default timeout is 3 seconds. However, it is worth noting that having this be configurable at all is not really the recommended behavior in RFC 2833. From Section 3.5 of RFC 2833: Limiting the time period of extending the tone is necessary to avoid that a tone "gets stuck". Regardless of the algorithm used, the tone SHOULD NOT be extended by more than three packet interarrival times. A slight extension of tone durations and shortening of pauses is generally harmless. Three seconds will pretty much _always_ be far more than three packet interarrival times. However, that behavior is not required, so I'm going to leave it with our legacy behavior for now. Code from svn/asterisk/team/russell/issue_14460 (closes issue #14460) Reported by: moliveras ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@175129 f38db490-d61c-443f-a65b-d21fe96a405b
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