2008-10-01 Russell Bryant * Asterisk 1.6.0 released. 2008-09-09 Russell Bryant * Asterisk 1.6.0-rc6 released. 2008-09-09 15:44 +0000 [r142065] Russell Bryant * /, main/features.c: Merged revisions 142064 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r142064 | russell | 2008-09-09 10:44:10 -0500 (Tue, 09 Sep 2008) | 13 lines Merged revisions 142063 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r142063 | russell | 2008-09-09 10:40:24 -0500 (Tue, 09 Sep 2008) | 5 lines Ensure that the stored CDR reference is still valid after the bridge before poking at it. Also, keep the channel locked while messing with this CDR. (fixes crashes reported in issue #13409) ........ ................ 2008-09-09 12:34 +0000 [r141996-141999] Mark Michelson * channels/chan_oss.c, /: Merged revisions 141995 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r141995 | mmichelson | 2008-09-09 05:20:58 -0500 (Tue, 09 Sep 2008) | 8 lines Fix a memory leak in chan_oss (closes issue #13311) Reported by: eliel Patches: chan_oss.c.patch uploaded by eliel (license 64) ........ 2008-09-09 01:49 +0000 [r141950] Russell Bryant * main/channel.c, /: Merged revisions 141949 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r141949 | russell | 2008-09-08 20:47:56 -0500 (Mon, 08 Sep 2008) | 9 lines Modify ast_answer() to not hold the channel lock while calling ast_safe_sleep() or when calling ast_waitfor(). These are inappropriate times to hold the channel lock. This is what has caused "could not get the channel lock" messages from chan_sip and has likely caused a negative impact on performance results of SIP in Asterisk 1.6. Thanks to file for pointing out this section of code. (closes issue #13287) (closes issue #13115) ........ 2008-09-08 21:07 +0000 [r141808] Russell Bryant * main/pbx.c, /: Merged revisions 141807 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r141807 | russell | 2008-09-08 16:05:01 -0500 (Mon, 08 Sep 2008) | 15 lines Merged revisions 141806 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r141806 | russell | 2008-09-08 16:02:36 -0500 (Mon, 08 Sep 2008) | 7 lines When doing an async goto, detect if the channel is already in the middle of a masquerade. This can happen when chan_local is trying to optimize itself out. If this happens, fail the async goto instead of bursting into flames. (closes issue #13435) Reported by: geoff2010 ........ ................ 2008-09-08 Russell Bryant * Asterisk 1.6.0-rc5 released. 2008-09-08 20:19 +0000 [r141746] Jason Parker * Makefile, /, redhat (removed): Merged revisions 141745 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r141745 | qwell | 2008-09-08 15:18:17 -0500 (Mon, 08 Sep 2008) | 16 lines Merged revisions 141741 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r141741 | qwell | 2008-09-08 15:15:42 -0500 (Mon, 08 Sep 2008) | 8 lines Remove RPM package targets from Makefile (and all associated parts). This has never worked in 1.4, and we decided that it makes no sense to be done here. There are many distros out there that already have "proper" spec files that can be (re)used. Closes issue #13113 Closes issue #10950 Closes issue #10952 ........ ................ 2008-09-08 17:14 +0000 [r141683] Sean Bright * /, build_tools/make_buildopts_h: Merged revisions 141682 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r141682 | seanbright | 2008-09-08 13:13:04 -0400 (Mon, 08 Sep 2008) | 9 lines Quote the arguments to grep so that sh on various platforms doesn't choke on the special characters (like ^). (closes issue #13417) Reported by: dougm Patches: 13417.make_buildopts_h.patch uploaded by seanbright (license 71) Tested by: dougm ........ 2008-09-06 20:21 +0000 [r141567] Steve Murphy * /, channels/chan_sip.c: Merged revisions 141566 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r141566 | murf | 2008-09-06 14:19:50 -0600 (Sat, 06 Sep 2008) | 9 lines Merged revisions 141565 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r141565 | murf | 2008-09-06 14:13:16 -0600 (Sat, 06 Sep 2008) | 1 line This fix comes from Joshua Colp The Brilliant, who, given the trace, came up with a solution. This will most likely will close 13235 and 13409. I'll wait till Monday to verify, and then close these bugs. ........ ................ 2008-09-06 15:40 +0000 [r141505-141508] Tilghman Lesher * /, res/res_agi.c: Merged revisions 141504 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r141504 | tilghman | 2008-09-06 10:26:45 -0500 (Sat, 06 Sep 2008) | 12 lines Merged revisions 141503 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r141503 | tilghman | 2008-09-06 10:23:42 -0500 (Sat, 06 Sep 2008) | 4 lines Reverting behavior change (AGI should not exit non-zero on SUCCESS) (closes issue #13434) Reported by: francesco_r ........ ................ 2008-09-05 22:06 +0000 [r141368-141426] Mark Michelson * /, channels/chan_agent.c: Merged revisions 141367 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r141367 | mmichelson | 2008-09-05 16:12:09 -0500 (Fri, 05 Sep 2008) | 15 lines Merged revisions 141366 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r141366 | mmichelson | 2008-09-05 16:10:32 -0500 (Fri, 05 Sep 2008) | 7 lines Agent's should not try to call a channel's indicate callback if the channel has been hung up. It will likely crash otherwise ABE-1159 ........ ................ 2008-09-05 14:24 +0000 [r141116-141158] Steve Murphy * main/channel.c, /: Merged revisions 141157 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r141157 | murf | 2008-09-05 08:18:43 -0600 (Fri, 05 Sep 2008) | 9 lines Merged revisions 141156 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r141156 | murf | 2008-09-05 08:15:43 -0600 (Fri, 05 Sep 2008) | 1 line A small change to prevent double-posting of CDR's; thanks to Daniel Ferrer for bringing it to our attention ........ ................ * pbx/ael/ael-test/ref.ael-vtest25 (added), /, pbx/ael/ael-test/ael-vtest25/extensions.ael, pbx/ael/ael-test/ael-vtest25 (added), res/ael/ael_lex.c, pbx/ael/ael-test/ref.ael-test6, res/ael/ael.flex: Merged revisions 141115 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r141115 | murf | 2008-09-04 17:31:41 -0600 (Thu, 04 Sep 2008) | 78 lines Merged revisions 141094 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r141094 | murf | 2008-09-04 17:15:07 -0600 (Thu, 04 Sep 2008) | 70 lines (closes issue #13357) Reported by: pj Tested by: murf (closes issue #13416) Reported by: yarns Tested by: murf If you find this message overly verbose, relax, it's probably not meant for you. This message is meant for probably only two people in the whole world: me, or the poor schnook that has to maintain this code because I'm either dead or unavailable at the moment. This fix solves two reports, both having to do with embedding a function call in a ${} construct. It was tricky because the funccall syntax has parenthesis () in it. And up till now, the 'word' token in the flex stuff didn't allow that, because it would tend to steal the LP and RP tokens. To be truthful, the "word" token was the trickiest, most unstable thing in the whole lexer. I was lucky it made this long without complaints. I had to choose every character in the pattern with extreme care, and I knew that someday I'd have to revisit it. Well, the day has come. So, my brilliant idea (and I'm being modest), was to use the surrounding ${} construct to make a state machine and capture everything in it, no matter what it contains. But, I have to now treat the word token like I did with comments, in that I turn the whole thing into a state-machine sort of spec, with new contexts "curlystate", "wordstate", and "brackstate". Wait a minute, "brackstate"? Yes, well, it didn't take very many regression tests to point out if I do this for ${} constructs, I also have to do it with the $[] constructs, too. I had to create a separate pcbstack2 and pcbstack3 because these constructs can occur inside macro argument lists, and when we have two state machines operating on the same structures we'd get problems otherwise. I guess I could have stopped at pcbstack2 and had the brackstate stuff share it, but it doesn't hurt to be safe. So, the pcbpush and pcbpop routines also now have versions for "2" and "3". I had to add the {KEYWORD} construct to the initial pattern for "word", because previously word would match stuff like "default7", because it was a longer match than the keyword "default". But, not any more, because the word pattern only matches only one or two characters now, and it will always lose. So, I made it the winner again by making an optional match on any of the keywords before it's normal pattern. I added another regression test to make sure we don't lose this in future edits, and had to fix just one regression, where it no longer reports a 'cascaded' error, which I guess is a plus. I've given some thought as to whether to apply these fixes to 1.4 and the 1.6.x releases, vs trunk; I decided to put it in 1.4 because one of the bug reports was against 1.4; and it is unexpected that AEL cannot handle this situation. It actually reduced the amount of useless "cascade" error messages that appeared in the regressions (by one line, ehhem). There is a possible side-effect in that it does now do more careful checking of what's in those ${} constructs, as far as matching parens, and brackets are concerned. Some users may find a an insidious problem and correct it this way. This should be exceedingly rare, I hope. ........ ................ 2008-09-04 18:35 +0000 [r141086] Jeff Peeler * /, main/features.c, res/res_agi.c: Merged revisions 141039 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r141039 | jpeeler | 2008-09-04 12:27:56 -0500 (Thu, 04 Sep 2008) | 15 lines Merged revisions 141028 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r141028 | jpeeler | 2008-09-04 12:00:29 -0500 (Thu, 04 Sep 2008) | 7 lines (closes issue #11979) Fixes multiple parking problems: Crash when executing a park on an extension dialed by AGI due to not returning the proper return code. Crash when using a builtin feature that was a subset of a enabled dynamic feature. Crash due to always hanging up the peer despite the fact that the peer was supposed to be parked. ........ ................ 2008-09-03 20:18 +0000 [r140976] Mark Michelson * /, apps/app_queue.c: Merged revisions 140975 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r140975 | mmichelson | 2008-09-03 15:16:12 -0500 (Wed, 03 Sep 2008) | 4 lines Fix some locking order issues in app_queue. This was brought up by atis on IRC a while ago. ........ 2008-09-03 Russell Bryant * Asterisk 1.6.0-rc4 released. 2008-09-03 14:17 +0000 [r140825-140827] Steve Murphy * main/cdr.c, /: Merged revisions 140749 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r140749 | murf | 2008-09-02 17:44:04 -0600 (Tue, 02 Sep 2008) | 11 lines Merged revisions 140747 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r140747 | murf | 2008-09-02 17:36:56 -0600 (Tue, 02 Sep 2008) | 1 line I am turning the warnings generated in ast_cdr_free and post_cdr into verbose level 2 messages. Really, they matter little to end users. You either get the CDR's you wanted, or you don't, and it is a bug. For trunk, I am going one step further. These messages were pretty worthless even for debug, so I'm completely removing them. ........ ................ * main/channel.c, /: Merged revisions 140692 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r140692 | murf | 2008-09-02 16:55:12 -0600 (Tue, 02 Sep 2008) | 13 lines Merged revisions 140690 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r140690 | murf | 2008-09-02 16:40:13 -0600 (Tue, 02 Sep 2008) | 1 line After reconsidering, with respect to 13409, ast_cdr_detach should be OK, better in fact, than ast_cdr_free, which generates lots of useless warnings that will undoubtably generate complaints. Hmmm. It doesn't hush the useless warnings, but it does allow control of posting via the detach and post routines, for those possible situations, where you'd want to post single-channel cdrs. ........ ................ * main/channel.c, main/pbx.c, /: Merged revisions 140691 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r140691 | murf | 2008-09-02 16:50:59 -0600 (Tue, 02 Sep 2008) | 22 lines Merged revisions 140670 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r140670 | murf | 2008-09-02 16:15:57 -0600 (Tue, 02 Sep 2008) | 14 lines (closes issue #13409) Reported by: tomaso Patches: asterisk-1.6.0-rc2-cdrmemleak.patch uploaded by tomaso (license 564) I basically spent the day, verifying that this patch solves the problem, and doesn't hurt in non-problem cases. Why valgrind did not plainly reveal this leak absolutely mystifies and stuns me. Many, many thanks to tomaso for finding and providing the fix. ........ ................ 2008-09-03 13:27 +0000 [r140818] Russell Bryant * main/poll.c, /: Merged revisions 140817 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r140817 | russell | 2008-09-03 08:26:43 -0500 (Wed, 03 Sep 2008) | 12 lines Merged revisions 140816 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r140816 | russell | 2008-09-03 08:24:35 -0500 (Wed, 03 Sep 2008) | 4 lines Don't freak out if the poll emulation receives NULL for the pollfds array (closes issue #13307) Reported by: jcovert ........ ................ 2008-09-02 18:17 +0000 [r140607] Sean Bright * /, channels/chan_iax2.c: Merged revisions 140606 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r140606 | seanbright | 2008-09-02 14:15:54 -0400 (Tue, 02 Sep 2008) | 16 lines Merged revisions 140605 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r140605 | seanbright | 2008-09-02 14:14:57 -0400 (Tue, 02 Sep 2008) | 8 lines Make sure to use the correct length of the mohinterpret and mohsuggest buffers when copying configuration values. (closes issue #13336) Reported by: decryptus_proformatique Patches: chan_iax2_mohinterpret_mohsuggest_general_settings.patch uploaded by decryptus (license 555) ........ ................ 2008-09-02 15:12 +0000 [r140564-140567] Russell Bryant * apps/app_jack.c, /, codecs/codec_resample.c: Merged revisions 140566 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r140566 | russell | 2008-09-02 10:11:53 -0500 (Tue, 02 Sep 2008) | 2 lines Update instructions for getting libresample ........ 2008-08-27 20:15 +0000 [r140302-140304] Mark Michelson * channels/chan_sip.c: Revert commit 140302. Should not be merging changes like that into a release-candidate branch * channels/chan_sip.c: Merged revisions 140301 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r140301 | mmichelson | 2008-08-27 15:11:22 -0500 (Wed, 27 Aug 2008) | 19 lines Merged revisions 140299 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r140299 | mmichelson | 2008-08-27 14:49:20 -0500 (Wed, 27 Aug 2008) | 11 lines Fix tag checking in get_sip_pvt_byid_locked when in pedantic mode. The problem was that the wrong tags would be compared depending on the direction of the call. (closes issue #13353) Reported by: flefoll Patches: chan_sip.c.br14.139015.patch-refer-pedantic uploaded by flefoll (license 244) ........ ................ 2008-08-26 18:12 +0000 [r140170] Russell Bryant * Makefile, /: Merged revisions 140169 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r140169 | russell | 2008-08-26 13:11:49 -0500 (Tue, 26 Aug 2008) | 4 lines Fix building menuselect-tree with PRINT_DIR set. We _must_ use the --quiet flag here, or else some arbitrary text will end up in the resulting menuselect-tree file and things will explode. ........ 2008-08-25 21:33 +0000 [r139918] Sean Bright * build_tools/get_moduleinfo, /, build_tools/get_makeopts: Merged revisions 139915 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r139915 | seanbright | 2008-08-25 17:32:10 -0400 (Mon, 25 Aug 2008) | 17 lines Merged revisions 139909 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r139909 | seanbright | 2008-08-25 17:31:03 -0400 (Mon, 25 Aug 2008) | 9 lines Some versions of awk (nawk, for example) don't like empty regular expressions so be slightly more verbose. (closes issue #13374) Reported by: dougm Patches: 13374.diff uploaded by seanbright (license 71) Tested by: dougm ........ ................ 2008-08-25 21:05 +0000 [r139872] Terry Wilson * /, channels/chan_sip.c: Merged revisions 139870 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r139870 | twilson | 2008-08-25 15:59:58 -0500 (Mon, 25 Aug 2008) | 10 lines Merged revisions 139869 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r139869 | twilson | 2008-08-25 15:46:10 -0500 (Mon, 25 Aug 2008) | 2 lines Make SIPADDHEADER() propagate indefinitely ........ ................ 2008-08-25 16:00 +0000 [r139774] Steve Murphy * main/pbx.c, /, main/features.c: Merged revisions 139770 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r139770 | murf | 2008-08-25 09:54:18 -0600 (Mon, 25 Aug 2008) | 17 lines Merged revisions 139764 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r139764 | murf | 2008-08-25 09:33:14 -0600 (Mon, 25 Aug 2008) | 9 lines This patch reverts the changes made via 139347, and 139635, as users are seeing adverse difference. I will un-close 13251. Back to the drawing board/ concept/ beginning/ whatever! ........ ................ 2008-08-24 16:30 +0000 [r139705-139708] Tilghman Lesher * /, cdr/cdr_pgsql.c: Merged revisions 139707 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r139707 | tilghman | 2008-08-24 11:26:48 -0500 (Sun, 24 Aug 2008) | 2 lines Memory leak ........ 2008-08-22 22:35 +0000 [r139628-139671] Steve Murphy * /, main/features.c: Merged revisions 139662 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r139662 | murf | 2008-08-22 16:32:35 -0600 (Fri, 22 Aug 2008) | 14 lines Merged revisions 139635 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r139635 | murf | 2008-08-22 16:24:02 -0600 (Fri, 22 Aug 2008) | 6 lines I found some problems with the code I committed earlier, when I merged them into trunk, so I'm coming back to clean up. And, in the process, I found an error in the code I added to trunk and 1.6.x, that I'll fix using this patch also. ........ ................ * apps/app_dial.c, main/pbx.c, /, main/features.c: Merged revisions 139627 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r139627 | murf | 2008-08-22 16:03:13 -0600 (Fri, 22 Aug 2008) | 59 lines Merged revisions 139347 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r139347 | murf | 2008-08-21 17:03:50 -0600 (Thu, 21 Aug 2008) | 47 lines (closes issue #13251) Reported by: sergee Tested by: murf THis is a bold move for a static release fix, but I wouldn't have made it if I didn't feel confident (at least a *bit* confident) that it wouldn't mess everyone up. The reasoning goes something like this: 1. We simply cannot do anything with CDR's at the current point (in pbx.c, after the __ast_pbx_run loop). It's way too late to have any affect on the CDRs. The CDR is already posted and gone, and the remnants have been cleared. 2. I was very much afraid that moving the running of the 'h' extension down into the bridge code (where it would be now practical to do it), would result in a lot more calls to the 'h' exten, so I implemented it as another exten under another name, but found, to my pleasant surprise, that there was a 1:1 correspondence to the running of the 'h' exten in the pbx_run loop, and the new spot at the end of the bridge. So, I ifdef'd out the current 'h' loop, and moved it into the bridge code. The only difference I can see is the stuff about the AST_PBX_KEEPALIVE, and hopefully, if this is still an important decision point, I can replicate it if there are complaints. To be perfectly honest, the KEEPALIVE situation is not totally clear to me, and how it relates to a post-bridge situation is less clear. I suspect the users will point out everything in total clarity if this steps on anyone's toes! 3. I temporarily swap the bridge_cdr into the channel before running the 'h' exten, which makes it possible for users to edit the cdr before it goes out the door. And, of course, with the endbeforehexten config var set, the users can also get at the billsec/duration vals. After the h exten finishes, the cdr is swapped back and processing continues as normal. Please, all who deal with CDR's, please test this version of Asterisk, and file bug reports as appropriate! ........ I also made a little fix to the app_dial's 'e' option, that is related to my updates. ................ 2008-08-22 20:21 +0000 [r139458-139564] Mark Michelson * include/asterisk/threadstorage.h, /: Merged revisions 139554 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r139554 | mmichelson | 2008-08-22 14:45:41 -0500 (Fri, 22 Aug 2008) | 16 lines Merged revisions 139553 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r139553 | mmichelson | 2008-08-22 14:45:19 -0500 (Fri, 22 Aug 2008) | 8 lines Fix compilation when DEBUG_THREAD_LOCALS is selected (closes issue #13298) Reported by: snuffy Patches: bug13298_20080822.diff uploaded by snuffy (license 35) ........ ................ * /, channels/chan_iax2.c: Merged revisions 139469 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r139469 | mmichelson | 2008-08-22 12:25:12 -0500 (Fri, 22 Aug 2008) | 11 lines Merged revisions 139466 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r139466 | mmichelson | 2008-08-22 12:24:47 -0500 (Fri, 22 Aug 2008) | 3 lines Fix the build. Thanks, mvanbaak! ........ ................ * /, channels/chan_iax2.c: Merged revisions 139457 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r139457 | mmichelson | 2008-08-22 11:58:21 -0500 (Fri, 22 Aug 2008) | 15 lines Merged revisions 139456 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r139456 | mmichelson | 2008-08-22 11:57:38 -0500 (Fri, 22 Aug 2008) | 7 lines Prevent a deadlock in chan_iax2 resulting from incorrect locking order between iax2_pvt and ast_channel structures. AST-13 ........ ................ 2008-08-21 23:46 +0000 [r139400] Jeff Peeler * channels/chan_dahdi.c, /: Merged revisions 139391 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r139391 | jpeeler | 2008-08-21 18:41:50 -0500 (Thu, 21 Aug 2008) | 11 lines Merged revisions 139387 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r139387 | jpeeler | 2008-08-21 18:39:31 -0500 (Thu, 21 Aug 2008) | 3 lines Fixes loop that could possibly never exit in the event of a channel never being able to be opened or specify after a restart. (closes issue #11017) ........ ................ 2008-08-21 10:02 +0000 [r139282] Philippe Sultan * /, channels/chan_gtalk.c: Merged revisions 139281 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r139281 | phsultan | 2008-08-21 11:55:31 +0200 (Thu, 21 Aug 2008) | 5 lines Fix two memory leaks in chan_gtalk, thanks Eliel! (closes issue #13310) Reported by: eliel Patches: chan_gtalk.c.patch uploaded by eliel (license 64) ........ 2008-08-20 Kevin P. Fleming * Asterisk 1.6.0-rc3 released. 2008-08-20 22:17 +0000 [r139216] Russell Bryant * apps/app_chanspy.c, /: Merged revisions 139215 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r139215 | russell | 2008-08-20 17:16:36 -0500 (Wed, 20 Aug 2008) | 19 lines Merged revisions 139213 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r139213 | russell | 2008-08-20 17:14:35 -0500 (Wed, 20 Aug 2008) | 11 lines Fix a crash in the ChanSpy application. The issue here is that if you call ChanSpy and specify a spy group, and sit in the application long enough looping through the channel list, you will eventually run out of stack space and the application with exit with a seg fault. The backtrace was always inside of a harmless snprintf() call, so it was tricky to track down. However, it turned out that the call to snprintf() was just the biggest stack consumer in this code path, so it would always be the first one to hit the boundary. (closes issue #13338) Reported by: ruddy ........ ................ 2008-08-20 20:12 +0000 [r139155] Shaun Ruffell * codecs/codec_dahdi.c: Fix bug where the samples were not accurate when in G723 mode, which would cause the timestamp field of the RTP header to be invalid. 2008-08-20 17:30 +0000 [r139104] Steve Murphy * main/cdr.c, /: Merged revisions 139083 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r139083 | murf | 2008-08-20 11:25:07 -0600 (Wed, 20 Aug 2008) | 20 lines Merged revisions 139074 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r139074 | murf | 2008-08-20 11:14:55 -0600 (Wed, 20 Aug 2008) | 12 lines (closes issue #13263) Reported by: brainy Tested by: murf The specialized reset routine is tromping on the flags field of the CDR. I made a change to not reset the DISABLED bit. This should get rid of this problem. ........ ................ 2008-08-20 15:39 +0000 [r138889-139017] Mark Michelson * /, channels/chan_sip.c: Merged revisions 139016 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r139016 | mmichelson | 2008-08-20 10:38:47 -0500 (Wed, 20 Aug 2008) | 14 lines Merged revisions 139015 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r139015 | mmichelson | 2008-08-20 10:37:56 -0500 (Wed, 20 Aug 2008) | 6 lines sip_read should properly handle a NULL return from sip_rtp_read. (closes issue #13257) Reported by: travishein ........ ................ * apps/app_chanspy.c: Manually add revision 138887 from trunk to the 1.6.0 branch. I had misunderstood the policy for when to merge to 1.6.0 since it moved to rc status. 2008-08-19 16:38 +0000 [r138846-138847] Steve Murphy * utils/conf2ael.c, /, res/ael/ael.tab.c, res/ael/ael.y, res/ael/ael.tab.h, utils/ael_main.c: Merged revisions 138845 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r138845 | murf | 2008-08-19 10:31:24 -0600 (Tue, 19 Aug 2008) | 1 line Oops. put a decl in a generated file. My bad, but fixed now. ........ * main/pbx.c, /, res/ael/ael.tab.c, res/ael/ael.y, res/ael/ael.tab.h: Merged revisions 138815 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r138815 | murf | 2008-08-19 09:59:12 -0600 (Tue, 19 Aug 2008) | 19 lines These changes are in regards to bug 13249, where users are being surprised by the changes made to the Set app in trunk/1.6.x, as they come from the 1.4 world. They are only bitten if they write their AEL dialplan in the 1.4 world, and then carry it over to a trunk/1.6.x installation where a "make samples" was executed, or where they hand-edited the asterisk.conf file and added the [compat] category with app_set = 1.6 (or higher). (this commit does not totally solve 13249, at least not yet) The change involves issueing a single warning while the AEL file is loading, if: 1. app_set is present in the config file, and set to 1.6 or higher. 2. there are double quotes in an assignment statement (eg x = "hi there";) 3. the warning was not already issued. The standalone app, aelparse, does not (yet) issue this warning. I'd have to have it read in the asterisk.conf file, and that's a bit of hassle. I'll add it if users request it, tho. ........ 2008-08-19 00:15 +0000 [r138776-138781] Sean Bright * /, channels/chan_sip.c: Merged revisions 138778-138780 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r138778 | seanbright | 2008-08-18 20:08:27 -0400 (Mon, 18 Aug 2008) | 1 line While we're at it, make this machine parseable too. ........ r138779 | seanbright | 2008-08-18 20:09:38 -0400 (Mon, 18 Aug 2008) | 1 line And remove code we don't need anymore. ........ r138780 | seanbright | 2008-08-18 20:10:56 -0400 (Mon, 18 Aug 2008) | 1 line Let it compile now, too (woops) ........ * /, channels/chan_sip.c: Merged revisions 138775 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r138775 | seanbright | 2008-08-18 19:42:36 -0400 (Mon, 18 Aug 2008) | 3 lines Change event header to RegistrationTime to be more consistent (and avoid breaking existing frameworks). Pointed out by Laureano on #asterisk-dev. ........ 2008-08-18 20:23 +0000 [r138688-138695] Mark Michelson * /, apps/app_queue.c: Merged revisions 138687 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r138687 | mmichelson | 2008-08-18 15:04:10 -0500 (Mon, 18 Aug 2008) | 18 lines Merged revisions 138685 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r138685 | mmichelson | 2008-08-18 15:01:14 -0500 (Mon, 18 Aug 2008) | 10 lines Change the inequalities used in app_queue with regards to timeouts from being strict to non-strict for more accuracy. (closes issue #13239) Reported by: atis Patches: app_queue_timeouts_v2.patch uploaded by atis (license 242) ........ ................ 2008-08-18 15:54 +0000 [r138632] Jason Parker * Makefile, /: Merged revisions 138631 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r138631 | qwell | 2008-08-18 10:54:07 -0500 (Mon, 18 Aug 2008) | 1 line Remove option that isn't valid here. ........ 2008-08-18 02:14 +0000 [r138519] Jeff Peeler * channels/chan_dahdi.c, /: Merged revisions 138518 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r138518 | jpeeler | 2008-08-17 21:13:04 -0500 (Sun, 17 Aug 2008) | 1 line add missing define for SS7 in dahdi_restart ........ 2008-08-17 14:14 +0000 [r138443-138483] Sean Bright * /, main/features.c: Merged revisions 138482 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r138482 | seanbright | 2008-08-17 10:12:11 -0400 (Sun, 17 Aug 2008) | 6 lines Move Uniqueid to the end of the event for those that rely on the position of the name/value pairs, pointed out by snuffy-home on #asterisk-commits. For those of you who rely on the position of name/value pairs in manager events... stop... that is why associative arrays were invented. ........ * /, main/features.c: Merged revisions 138479 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r138479 | seanbright | 2008-08-17 09:51:08 -0400 (Sun, 17 Aug 2008) | 7 lines Add Uniqueid header to ParkedCall manager event. (closes issue #13323) Reported by: srt Patches: 13323_unique_id_for_parkedcalls_event.diff uploaded by srt (license 378) ........ * main/rtp.c, /: Merged revisions 138476 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r138476 | seanbright | 2008-08-17 09:40:36 -0400 (Sun, 17 Aug 2008) | 7 lines Add missing colons to RTCPReceived and RTCPSent manager events. (closes issue #13319) Reported by: srt Patches: 13319_rtcp_manager_event_headers.diff uploaded by srt (license 378) ........ * /, channels/chan_iax2.c: Merged revisions 138473 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r138473 | seanbright | 2008-08-17 09:31:54 -0400 (Sun, 17 Aug 2008) | 7 lines Fix the output of the JitterBufStats manager event. (closes issue #13324) Reported by: srt Patches: 13324_missing_nl_in_jitterbufstats_event_2.diff uploaded by srt (license 378) ........ * configs/cdr_tds.conf.sample, /: Merged revisions 138442 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r138442 | seanbright | 2008-08-16 12:40:43 -0400 (Sat, 16 Aug 2008) | 4 lines Since it's introduction in revision 3497, cdr_tds has *never* read the port configuration option from cdr_tds.conf. So go ahead and remove it from the sample config. ........ 2008-08-16 13:07 +0000 [r138410-138413] Tilghman Lesher * channels/chan_dahdi.c, /: Merged revisions 138412 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r138412 | tilghman | 2008-08-16 08:07:08 -0500 (Sat, 16 Aug 2008) | 2 lines Fix compilation warnings (found with dev-mode) ........ 2008-08-16 01:14 +0000 [r138333-138362] Jeff Peeler * channels/chan_dahdi.c, /: Merged revisions 138361 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r138361 | jpeeler | 2008-08-15 20:13:26 -0500 (Fri, 15 Aug 2008) | 9 lines Merged revisions 138360 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r138360 | jpeeler | 2008-08-15 20:12:18 -0500 (Fri, 15 Aug 2008) | 1 line fixes use count to properly decrement if an active dahdi channel is destroyed allowing module to be unloaded ........ ................ * channels/chan_dahdi.c, /: Merged revisions 138311 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r138311 | jpeeler | 2008-08-15 18:46:09 -0500 (Fri, 15 Aug 2008) | 20 lines Merged revisions 138119,138151,138238 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r138119 | jpeeler | 2008-08-15 14:21:51 -0500 (Fri, 15 Aug 2008) | 4 lines Fixes the dahdi restart functionality. Dahdi restart allows one to restart all DAHDI channels, even if they are currently in use. This is different from unloading and then loading the module since unloading requires the use count to be zero. Reloading the module is different in that the signalling is not changed from what it was originally configured. Also, this fixes not closing all the file descriptors for D-channels upon module unload (which would prevent loading the module afterwards). (closes issue #11017) ........ r138151 | jpeeler | 2008-08-15 14:41:29 -0500 (Fri, 15 Aug 2008) | 1 line declared static mutexes using AST_MUTEX_DEFINE_STATIC macro ........ r138238 | jpeeler | 2008-08-15 16:28:26 -0500 (Fri, 15 Aug 2008) | 1 line initialize condition variable ss_thread_complete using ast_cond_init ........ ................ 2008-08-15 23:03 +0000 [r138207-138262] Tilghman Lesher * /, channels/chan_sip.c, configs/sip.conf.sample: Merged revisions 138260 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r138260 | tilghman | 2008-08-15 17:54:57 -0500 (Fri, 15 Aug 2008) | 16 lines Merged revisions 138258 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r138258 | tilghman | 2008-08-15 17:33:42 -0500 (Fri, 15 Aug 2008) | 8 lines More fixes for realtime peers. (closes issue #12921) Reported by: Nuitari Patches: 20080804__bug12921.diff.txt uploaded by Corydon76 (license 14) 20080815__bug12921.diff.txt uploaded by Corydon76 (license 14) Tested by: Corydon76 ........ ................ * configs/extensions.conf.sample, main/pbx.c, /: Merged revisions 138206 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r138206 | tilghman | 2008-08-15 15:35:24 -0500 (Fri, 15 Aug 2008) | 4 lines Remove deprecated syntax from sample config file (closes issue #13314) Reported by: kue ........ 2008-08-15 20:20 +0000 [r138156-138157] Jeff Peeler * channels/chan_dahdi.c: rename all zfd instances in chan_dahdi to dfd to match 1.4 (left over from DAHDI transition) 2008-08-15 15:12 +0000 [r138029] Russell Bryant * main/autoservice.c, /: Merged revisions 138028 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r138028 | russell | 2008-08-15 10:09:46 -0500 (Fri, 15 Aug 2008) | 17 lines Merged revisions 138027 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r138027 | russell | 2008-08-15 10:07:16 -0500 (Fri, 15 Aug 2008) | 9 lines Ensure that when a hangup occurs in autoservice, that a hangup frame gets properly deferred to be read from the channel owner when it gets taken out of autoservice. (closes issue #12874) Reported by: dimas Patches: v1-12874.patch uploaded by dimas (license 88) ........ ................ 2008-08-15 15:04 +0000 [r138025] Tilghman Lesher * /, funcs/func_strings.c: Merged revisions 138024 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r138024 | tilghman | 2008-08-15 10:03:32 -0500 (Fri, 15 Aug 2008) | 16 lines Merged revisions 138023 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r138023 | tilghman | 2008-08-15 09:51:12 -0500 (Fri, 15 Aug 2008) | 8 lines Additional check for more string specifiers than arguments. (closes issue #13299) Reported by: adomjan Patches: 20080813__bug13299.diff.txt uploaded by Corydon76 (license 14) func_strings.c-sprintf.patch uploaded by adomjan (license 487) Tested by: adomjan ........ ................ 2008-08-14 22:43 +0000 [r137988] Russell Bryant * /, doc/tex/Makefile: Merged revisions 137987 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r137987 | russell | 2008-08-14 17:43:15 -0500 (Thu, 14 Aug 2008) | 2 lines Fix a bashism that causes an error when trying to build the pdf on ubuntu ........ 2008-08-14 18:48 +0000 [r137934] Sean Bright * cdr/cdr_sqlite3_custom.c, /: Merged revisions 137933 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r137933 | seanbright | 2008-08-14 14:47:28 -0400 (Thu, 14 Aug 2008) | 8 lines Fix memory leak in cdr_sqlite3_custom. (closes issue #13304) Reported by: eliel Patches: sqlite.patch uploaded by eliel (license 64) (Slightly modified by me) ........ 2008-08-14 17:01 +0000 [r137849-137852] Tilghman Lesher * channels/chan_dahdi.c, /: Merged revisions 137848 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r137848 | tilghman | 2008-08-14 11:52:43 -0500 (Thu, 14 Aug 2008) | 17 lines Merged revisions 137847 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r137847 | tilghman | 2008-08-14 11:47:30 -0500 (Thu, 14 Aug 2008) | 9 lines When creating the secondary subchannel name, it is necessary to compare to the existing channel name without the "Zap/" or "DAHDI/" prefix, since our test string is also without that prefix. (closes issue #13027) Reported by: dferrer Patches: chan_zap-1.4.21.1_fix2.patch uploaded by dferrer (license 525) (Slightly modified by me, to compensate for both names) ........ ................ 2008-08-14 Jason Parker * Asterisk 1.6.0-rc2 released. 2008-08-14 15:37 +0000 [r137814] Jason Parker * /, channels/chan_sip.c: Merged revisions 137812 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r137812 | qwell | 2008-08-14 10:32:16 -0500 (Thu, 14 Aug 2008) | 8 lines Make sure we set the socket port, so we don't try to use :0. (closes issue #13255) Reported by: falves11 Patches: 13255-socketport.diff uploaded by qwell (license 4) Tested by: falves11 ........ 2008-08-14 15:20 +0000 [r137783] Russell Bryant * /, configs/sip.conf.sample: Merged revisions 137732 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r137732 | russell | 2008-08-14 09:15:50 -0500 (Thu, 14 Aug 2008) | 12 lines Merged revisions 137731 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r137731 | russell | 2008-08-14 09:05:23 -0500 (Thu, 14 Aug 2008) | 4 lines Comments in this config file were aligned only if your tab size was set to 8. So, convert tabs to spaces so that things should be aligned regardless of what tab size you use in your editor. ........ ................ 2008-08-14 15:05 +0000 [r137781] Sean Bright * cdr/cdr_tds.c, /: Merged revisions 137780 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r137780 | seanbright | 2008-08-14 11:03:03 -0400 (Thu, 14 Aug 2008) | 8 lines If we detect that we are no longer connected, try to reconnect a few times before giving up. This relies on the timeout settings in the freetds.conf file and, unfortunately, on a recent version of FreeTDS (0.82 or newer). I either need to change the current execs to be non-blocking (which I do not want to do) or we have to force people to run with the latest and greatest of FreeTDS. I'm on the fence... ........ 2008-08-14 02:04 +0000 [r137681] Kevin P. Fleming * /, Zaptel-to-DAHDI.txt: Merged revisions 137680 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r137680 | kpfleming | 2008-08-13 21:03:47 -0500 (Wed, 13 Aug 2008) | 9 lines Merged revisions 137679 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r137679 | kpfleming | 2008-08-13 21:03:04 -0500 (Wed, 13 Aug 2008) | 1 line forgot one module name that changed ........ ................ 2008-08-13 Kevin P. Fleming * Asterisk 1.6.0-rc1 released. 2008-08-13 23:00 +0000 [r137631-137641] Kevin P. Fleming * /, build_tools/prep_tarball: Merged revisions 137640 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r137640 | kpfleming | 2008-08-13 18:00:37 -0500 (Wed, 13 Aug 2008) | 1 line make this script actually work ........ * /, Zaptel-to-DAHDI.txt (added), UPGRADE.txt: Merged revisions 137627 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r137627 | kpfleming | 2008-08-13 17:33:32 -0500 (Wed, 13 Aug 2008) | 9 lines Merged revisions 137530 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r137530 | kpfleming | 2008-08-13 15:46:19 -0500 (Wed, 13 Aug 2008) | 1 line add document describing what users will need to be aware of when upgrading to this version and using DAHDI ........ ................ 2008-08-13 21:09 +0000 [r137497-137533] Jason Parker * /, channels/chan_sip.c: Merged revisions 137532 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r137532 | qwell | 2008-08-13 16:08:58 -0500 (Wed, 13 Aug 2008) | 8 lines Correctly end locally ended calls. (closes issue #12170) Reported by: pj Patches: 20080702__issue12170_clear_pendinginvite.diff uploaded by bbryant (license 36) Tested by: bbryant, pabelanger ........ * /, apps/app_fax.c: Merged revisions 137496 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r137496 | qwell | 2008-08-13 15:05:50 -0500 (Wed, 13 Aug 2008) | 6 lines Add FAXMODE variable with what fax transport was used. (closes issue #13252) Patches: v1-13252.patch uploaded by dimas (license 88) ........ 2008-08-13 14:47 +0000 [r137350-137407] Sean Bright * /, doc/tex/cdrdriver.tex: Merged revisions 137406 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r137406 | seanbright | 2008-08-13 10:41:49 -0400 (Wed, 13 Aug 2008) | 9 lines Merged revisions 137405 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r137405 | seanbright | 2008-08-13 10:33:49 -0400 (Wed, 13 Aug 2008) | 1 line Update docs to reflect the change to cdr_tds ........ ................ * cdr/cdr_tds.c, /: Merged revisions 137403 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r137403 | seanbright | 2008-08-13 10:22:47 -0400 (Wed, 13 Aug 2008) | 1 line Use the ast_vasprintf macro instead of vasprintf directly. ........ 2008-08-12 19:48 +0000 [r137300-137302] Russell Bryant * doc/tex/asterisk.tex, /: Merged revisions 137301 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r137301 | russell | 2008-08-12 14:48:38 -0500 (Tue, 12 Aug 2008) | 2 lines Grammar hax from Qwell ........ * doc/tex/asterisk.tex, /: Merged revisions 137299 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r137299 | russell | 2008-08-12 14:40:35 -0500 (Tue, 12 Aug 2008) | 3 lines Note that developer documentation belongs in doxygen, and not integrated with the user manual stuff in doc/tex/. ........ 2008-08-11 16:15 +0000 [r137240] Russell Bryant * Makefile, /: Merged revisions 137239 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r137239 | russell | 2008-08-11 11:14:29 -0500 (Mon, 11 Aug 2008) | 2 lines Make PRINT_DIR work as advertised. ........ 2008-08-11 14:31 +0000 [r137217] Sean Bright * cdr/cdr_tds.c, /, UPGRADE.txt: Merged revisions 137203 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r137203 | seanbright | 2008-08-11 10:25:15 -0400 (Mon, 11 Aug 2008) | 7 lines Log the userfield CDR variable like the other CDR backends, assuming the column is actually there. If it's not, we still log everything else as before. (closes issue #13281) Reported by: falves11 ........ 2008-08-11 00:27 +0000 [r137160] Tilghman Lesher * res/res_odbc.c, /: Merged revisions 137150 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r137150 | tilghman | 2008-08-10 19:25:28 -0500 (Sun, 10 Aug 2008) | 13 lines Merged revisions 137138 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r137138 | tilghman | 2008-08-10 19:20:38 -0500 (Sun, 10 Aug 2008) | 5 lines Deallocate database connection handle on disconnect, as we allocate another one on connect. (closes issue #13271) Reported by: dveiga ........ ................ 2008-08-09 15:27 +0000 [r136948] Tilghman Lesher * /, include/asterisk/compat.h, include/asterisk/astobj2.h: Merged revisions 136947 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r136947 | tilghman | 2008-08-09 10:26:27 -0500 (Sat, 09 Aug 2008) | 18 lines Merged revisions 136946 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r136946 | tilghman | 2008-08-09 10:25:36 -0500 (Sat, 09 Aug 2008) | 10 lines Merged revisions 136945 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r136945 | tilghman | 2008-08-09 10:24:36 -0500 (Sat, 09 Aug 2008) | 2 lines Regression fixes for Solaris ........ ................ ................ 2008-08-09 01:16 +0000 [r136860] Tilghman Lesher * /, res/res_agi.c: Merged revisions 136859 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r136859 | tilghman | 2008-08-08 20:15:38 -0500 (Fri, 08 Aug 2008) | 4 lines Update documentation as to the behavior of AGI in 1.6.0 and higher. Also, add an OOB message that answers the question of, if AGI no longer shuts down the connection on hangup, how will FastAGI know when to stop processing the call? ........ 2008-08-08 15:33 +0000 [r136785] Mark Michelson * apps/app_voicemail.c, /: Merged revisions 136784 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r136784 | mmichelson | 2008-08-08 10:31:31 -0500 (Fri, 08 Aug 2008) | 3 lines Fix compilation for ODBC voicemail ........ 2008-08-08 06:45 +0000 [r136778] Steve Murphy * pbx/ael/ael-test/ref.ael-test8, pbx/ael/ael-test/ref.ael-test18, pbx/ael/ael-test/ref.ael-test19, pbx/ael/ael-test/ref.ael-vtest13, res/ael/pval.c, /, pbx/ael/ael-test/ref.ael-ntest10, include/asterisk/ael_structs.h, utils/ael_main.c: Merged revisions 136746 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r136746 | murf | 2008-08-07 18:48:35 -0600 (Thu, 07 Aug 2008) | 40 lines Merged revisions 136726 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r136726 | murf | 2008-08-07 18:15:34 -0600 (Thu, 07 Aug 2008) | 32 lines (closes issue #13236) Reported by: korihor Wow, this one was a challenge! I regrouped and ran a new strategy for setting the ~~MACRO~~ value; I set it once per extension, up near the top. It is only set if there is a switch in the extension. So, I had to put in a chunk of code to detect a switch in the pval tree. I moved the code to insert the set of ~~exten~~ up to the beginning of the gen_prios routine, instead of down in the switch code. I learned that I have to push the detection of the switches down into the code, so everywhere I create a new exten in gen_prios, I make sure to pass onto it the values of the mother_exten first, and the exten next. I had to add a couple fields to the exten struct to accomplish this, in the ael_structs.h file. The checked field makes it so we don't repeat the switch search if it's been done. I also updated the regressions. ........ ................ 2008-08-08 02:36 +0000 [r136753] Tilghman Lesher * /: Merged revisions 136751 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r136751 | tilghman | 2008-08-07 21:34:17 -0500 (Thu, 07 Aug 2008) | 2 lines Removing bad properties ........ 2008-08-07 23:42 +0000 [r136719-136724] Mark Michelson * apps/app_voicemail.c: This is weird. Either SVN or vim tabbed a bunch of functions over one level during a merge. * apps/app_voicemail.c, /: Merged revisions 136722 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r136722 | mmichelson | 2008-08-07 18:39:50 -0500 (Thu, 07 Aug 2008) | 3 lines Remove one last batch of debug messages ........ * apps/app_voicemail.c, /: Merged revisions 136715 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r136715 | mmichelson | 2008-08-07 17:25:50 -0500 (Thu, 07 Aug 2008) | 18 lines Merging the imap_consistency_trunk branch to trunk. For an explanation of what "imap_consistency" is, please see svn revision 134223 to the 1.4 branch. Coincidentally, this also fixes a recent bug report regarding the inability to save messages to the new folder when using IMAP storage since they will would be flagged as "seen" and not be recognized as new messages. (closes issue #13234) Reported by: jaroth ........ 2008-08-07 20:41 +0000 [r136672-136674] Shaun Ruffell * codecs/codec_dahdi.c: Removing code that was commented out. * codecs/codec_dahdi.c: Updated codec_dahdi to use the transcoder interface in the DAHDI. (Issue: DAHDI-42) 2008-08-07 20:26 +0000 [r136632-136663] Mark Michelson * /, main/features.c: Merged revisions 136660 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r136660 | mmichelson | 2008-08-07 15:25:43 -0500 (Thu, 07 Aug 2008) | 4 lines Bump a LOG_NOTICE message to LOG_DEBUG since it appears once for every bridged call ........ * main/pbx.c, /: Merged revisions 136635 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r136635 | mmichelson | 2008-08-07 14:58:32 -0500 (Thu, 07 Aug 2008) | 5 lines Don't allow Answer() to accept a negative argument. Negative argument means an infinite delay and we don't want that. ........ * main/channel.c, /: Merged revisions 136633 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r136633 | mmichelson | 2008-08-07 14:54:27 -0500 (Thu, 07 Aug 2008) | 7 lines Fix a calculation error I had made in the poll. The poll would reset to 500 ms every time a non-voice frame was received. The total time we poll should be 500 ms, so now we save the amount of time left after the poll returned and use that as our argument for the next call to poll ........ * main/channel.c, /: Merged revisions 136631 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r136631 | mmichelson | 2008-08-07 14:36:46 -0500 (Thu, 07 Aug 2008) | 13 lines Scrap the 500 ms delay when Asterisk auto-answers a channel. Instead, poll the channel until receiving a voice frame. The cap on this poll is 500 ms. The optional delay is still allowable in the Answer() application, but the delay has been moved back to its original position, after the call to the channel's answer callback. The poll for the voice frame will not happen if a delay is specified when calling Answer(). (closes issue #12708) Reported by: kactus ........ 2008-08-07 19:19 +0000 [r136598] Richard Mudgett * channels/misdn_config.c, channels/chan_misdn.c, /, configs/misdn.conf.sample: Merged revisions 136594 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r136594 | rmudgett | 2008-08-07 14:01:03 -0500 (Thu, 07 Aug 2008) | 13 lines Merged revisions 136241 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r136241 | rmudgett | 2008-08-06 16:18:53 -0500 (Wed, 06 Aug 2008) | 5 lines * The allowed_bearers setting in misdn.conf misspelled one of its options: digital_restricted. * Fixed some other spelling errors and typos. ........ ................ 2008-08-07 17:44 +0000 [r136506-136543] Kevin P. Fleming * include/asterisk/doxyref.h, /: Merged revisions 136542 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r136542 | kpfleming | 2008-08-07 12:44:20 -0500 (Thu, 07 Aug 2008) | 6 lines Merged revisions 136541 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ ........ ................ 2008-08-07 16:57 +0000 [r136490] Tilghman Lesher * /, apps/app_queue.c: Merged revisions 136489 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r136489 | tilghman | 2008-08-07 11:55:57 -0500 (Thu, 07 Aug 2008) | 15 lines Merged revisions 136488 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r136488 | tilghman | 2008-08-07 11:50:47 -0500 (Thu, 07 Aug 2008) | 7 lines Update persistent state on all exit conditions. (closes issue #12916) Reported by: sgenyuk Patches: app_queue.patch.txt uploaded by neutrino88 (license 297) Tested by: sgenyuk, aragon ........ ................ 2008-08-06 20:16 +0000 [r136113-136192] Tilghman Lesher * /, contrib/init.d/rc.redhat.asterisk: Merged revisions 136191 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r136191 | tilghman | 2008-08-06 15:15:34 -0500 (Wed, 06 Aug 2008) | 12 lines Merged revisions 136190 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r136190 | tilghman | 2008-08-06 15:14:54 -0500 (Wed, 06 Aug 2008) | 4 lines -C option takes a filename, not a directory path. (closes issue #13007) Reported by: klaus3000 ........ ................ * /, funcs/func_dialgroup.c: Merged revisions 136112 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r136112 | tilghman | 2008-08-06 11:58:42 -0500 (Wed, 06 Aug 2008) | 7 lines Persist DIALGROUP() values in astdb (closes issue #13138) Reported by: Corydon76 Patches: 20080725__bug13138.diff.txt uploaded by Corydon76 (license 14) Tested by: pj ........ 2008-08-06 16:00 +0000 [r136064] Mark Michelson * main/rtp.c, /, channels/chan_skinny.c: Merged revisions 136063 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r136063 | mmichelson | 2008-08-06 10:59:29 -0500 (Wed, 06 Aug 2008) | 24 lines Merged revisions 136062 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r136062 | mmichelson | 2008-08-06 10:58:40 -0500 (Wed, 06 Aug 2008) | 16 lines Since adding the AST_CONTROL_SRCUPDATE frame type, there are places where ast_rtp_new_source may be called where the tech_pvt of a channel may not yet have an rtp structure allocated. This caused a crash in chan_skinny, which was fixed earlier, but now the same crash has been reported against chan_h323 as well. It seems that the best solution is to modify ast_rtp_new_source to not attempt to set the marker bit if the rtp structure passed in is NULL. This change to ast_rtp_new_source also allows the removal of what is now a redundant pointer check from chan_skinny. (closes issue #13247) Reported by: pj ........ ................ 2008-08-06 13:59 +0000 [r136006] Olle Johansson * /, res/res_jabber.c: Merged revisions 136005 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r136005 | oej | 2008-08-06 15:34:08 +0200 (Ons, 06 Aug 2008) | 6 lines - Formatting - Changing debug messages from VERBOSE to DEBUG channel - Adding a few todo's - Adding a few more "XMPP"'s to compliment Jabber... ........ 2008-08-06 03:56 +0000 [r135901-135951] Tilghman Lesher * main/channel.c, /: Merged revisions 135950 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r135950 | tilghman | 2008-08-05 22:55:49 -0500 (Tue, 05 Aug 2008) | 12 lines Merged revisions 135949 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r135949 | tilghman | 2008-08-05 22:53:36 -0500 (Tue, 05 Aug 2008) | 4 lines Fix a longstanding bug in channel walking logic, and fix the explanation to make sense. (Closes issue #13124) ........ ................ * /, main/translate.c: Merged revisions 135938 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r135938 | tilghman | 2008-08-05 22:29:42 -0500 (Tue, 05 Aug 2008) | 12 lines Merged revisions 135915 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r135915 | tilghman | 2008-08-05 22:24:56 -0500 (Tue, 05 Aug 2008) | 4 lines Since powerof() can return an error condition, it's foolhardy not to detect and deal with that condition. (Related to issue #13240) ........ ................ * include/asterisk/threadstorage.h, include/asterisk/utils.h, /: Merged revisions 135900 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r135900 | tilghman | 2008-08-05 22:04:01 -0500 (Tue, 05 Aug 2008) | 12 lines Merged revisions 135899 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r135899 | tilghman | 2008-08-05 22:02:59 -0500 (Tue, 05 Aug 2008) | 4 lines 1) Bugfix for debugging code 2) Reduce compiler warnings for another section of debugging code (Closes issue #13237) ........ ................ 2008-08-06 00:31 +0000 [r135852] Mark Michelson * include/asterisk/abstract_jb.h, main/channel.c, /, main/abstract_jb.c, main/fixedjitterbuf.h: Merged revisions 135851 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r135851 | mmichelson | 2008-08-05 19:30:53 -0500 (Tue, 05 Aug 2008) | 48 lines Merged revisions 135841,135847,135850 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r135841 | mmichelson | 2008-08-05 19:25:10 -0500 (Tue, 05 Aug 2008) | 27 lines Merging the issue11259 branch. The purpose of this branch was to take into account "burps" which could cause jitterbuffers to misbehave. One such example is if the L option to Dial() were used to inject audio into a bridged conversation at regular intervals. Since the audio here was not passed through the jitterbuffer, it would cause a gap in the jitterbuffer's timestamps which would cause a frames to be dropped for a brief period. Now ast_generic_bridge will empty and reset the jitterbuffer each time it is called. This causes injected audio to be handled properly. ast_generic_bridge also will empty and reset the jitterbuffer if it receives an AST_CONTROL_SRCUPDATE frame since the change in audio source could negatively affect the jitterbuffer. All of this was made possible by adding a new public API call to the abstract_jb called ast_jb_empty_and_reset. (closes issue #11259) Reported by: plack Tested by: putnopvut ........ r135847 | mmichelson | 2008-08-05 19:27:54 -0500 (Tue, 05 Aug 2008) | 4 lines Revert inadvertent changes to app_skel that occurred when I was testing for a memory leak ........ r135850 | mmichelson | 2008-08-05 19:29:54 -0500 (Tue, 05 Aug 2008) | 3 lines Remove properties that should not be here ........ ................ 2008-08-05 23:52 +0000 [r135822] Steve Murphy * apps/app_dial.c, main/cdr.c, main/channel.c, /, main/features.c, include/asterisk/cdr.h: Merged revisions 135821 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r135821 | murf | 2008-08-05 17:45:32 -0600 (Tue, 05 Aug 2008) | 42 lines Merged revisions 135799 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r135799 | murf | 2008-08-05 17:13:20 -0600 (Tue, 05 Aug 2008) | 34 lines (closes issue #12982) Reported by: bcnit Tested by: murf I discovered that also, in the previous bug fixes and changes, the cdr.conf 'unanswered' option is not being obeyed, so I fixed this. And, yes, there are two 'answer' times involved in this scenario, and I would agree with you, that the first answer time is the time that should appear in the CDR. (the second 'answer' time is the time that the bridge was begun). I made the necessary adjustments, recording the first answer time into the peer cdr, and then using that to override the bridge cdr's value. To get the 'unanswered' CDRs to appear, I purposely output them, using the dial cmd to mark them as DIALED (with a new flag), and outputting them if they bear that flag, and you are in the right mode. I also corrected one small mention of the Zap device to equally consider the dahdi device. I heavily tested 10-sec-wait macros in dial, and without the macro call; I tested hangups while the macro was running vs. letting the macro complete and the bridge form. Looks OK. Removed all the instrumentation and debug. ........ ................ 2008-08-05 21:38 +0000 [r135749] Tilghman Lesher * /, channels/chan_iax2.c: Merged revisions 135748 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r135748 | tilghman | 2008-08-05 16:37:35 -0500 (Tue, 05 Aug 2008) | 17 lines Merged revisions 135747 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r135747 | tilghman | 2008-08-05 16:34:46 -0500 (Tue, 05 Aug 2008) | 9 lines In a conversion to use ast_strlen_zero, the meaning of the flag IAX_HASCALLERID was perverted. This change reverts IAX2 to the original meaning, which was, that the callerid set on the client should be overridden on the server, even if that means the resulting callerid is blank. In other words, if you set "callerid=" in the IAX config, then the callerid should be overridden to blank, even if set on the client. Note that there's a distinction, even on realtime, between the field not existing (NULL in databases) and the field existing, but set to blank (override callerid to blank). ........ ................ 2008-08-05 13:27 +0000 [r135599] Sean Bright * main/cli.c, /: Merged revisions 135598 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r135598 | seanbright | 2008-08-05 09:26:34 -0400 (Tue, 05 Aug 2008) | 9 lines Merged revisions 135597 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r135597 | seanbright | 2008-08-05 09:25:00 -0400 (Tue, 05 Aug 2008) | 1 line Use PATH_MAX for filenames ........ ................ 2008-08-04 20:15 +0000 [r135538] Russell Bryant * configs/chan_dahdi.conf.sample, /: Merged revisions 135537 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r135537 | russell | 2008-08-04 15:15:27 -0500 (Mon, 04 Aug 2008) | 10 lines Merged revisions 135536 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r135536 | russell | 2008-08-04 15:15:03 -0500 (Mon, 04 Aug 2008) | 2 lines fix a config sample typo ........ ................ 2008-08-04 17:12 +0000 [r135478-135486] Tilghman Lesher * contrib/init.d/rc.mandriva.asterisk (added), Makefile, contrib/init.d/rc.mandrake.asterisk (removed), /, contrib/init.d/rc.mandriva.zaptel (added), contrib/init.d/rc.mandrake.zaptel (removed): Merged revisions 135485 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r135485 | tilghman | 2008-08-04 12:12:15 -0500 (Mon, 04 Aug 2008) | 3 lines Rename Mandrake scripts to Mandriva (Closes issue #13221) ........ * contrib/init.d/rc.mandrake.asterisk, /: Merged revisions 135483 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r135483 | tilghman | 2008-08-04 12:08:42 -0500 (Mon, 04 Aug 2008) | 11 lines Merged revisions 135482 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r135482 | tilghman | 2008-08-04 12:07:52 -0500 (Mon, 04 Aug 2008) | 2 lines Define ASTSBINDIR for script (Closes issue #13221) ........ ................ * apps/app_voicemail.c, /: Merged revisions 135480 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r135480 | tilghman | 2008-08-04 11:58:29 -0500 (Mon, 04 Aug 2008) | 14 lines Merged revisions 135479 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r135479 | tilghman | 2008-08-04 11:56:19 -0500 (Mon, 04 Aug 2008) | 6 lines Memory leak on unload (closes issue #13231) Reported by: eliel Patches: app_voicemail.leak.patch uploaded by eliel (license 64) ........ ................ 2008-08-04 16:28 +0000 [r135440-135475] Russell Bryant * configs/chan_dahdi.conf.sample, /: Merged revisions 135474 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r135474 | russell | 2008-08-04 11:28:07 -0500 (Mon, 04 Aug 2008) | 10 lines Merged revisions 135473 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r135473 | russell | 2008-08-04 11:26:17 -0500 (Mon, 04 Aug 2008) | 2 lines Add a minor clarification to the documentation of mohinterpret and mohsuggest ........ ................ * /, channels/chan_console.c: Merged revisions 135439 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r135439 | russell | 2008-08-04 10:02:12 -0500 (Mon, 04 Aug 2008) | 4 lines Be explicit that we don't want a result from this callback. The callback would never indicate a match, so nothing would have been returned anyway, but it was still a poor example of proper usage. ........ 2008-08-02 05:15 +0000 [r135266] Steve Murphy * main/pbx.c, /: Merged revisions 135265 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r135265 | murf | 2008-08-01 22:51:29 -0600 (Fri, 01 Aug 2008) | 31 lines (closes issue #13202) Reported by: falves11 Tested by: murf falves11 == The changes I introduce here seem to clear up the problem for me. However, if they do not for you, please reopen this bug, and we'll keep digging. The root of this problem seems to be a subtle memory corruption introduced when creating an extension with an empty extension name. While valgrind cannot detect it outside of DEBUG_MALLOC mode, when compiled with DEBUG_MALLOC, this is certain death. The code in main/features.c is a puzzle to me. On the initial module load, the code is attempting to add the parking extension before the features.conf file has even been opened! I just wrapped the offending call with an if() that will not try to add the extension if the extension name is empty. THis seems to solve the corruption, and let the "memory show allocations" work as one would expect. But, really, adding an extension with an empty name is a seriously bad thing to allow, as it will mess up all the pattern matching algorithms, etc. So, I added a statement to the add_extension2 code to return a -1 if this is attempted. in 1.6.0, the changes to only main/pbx.c were applicable, as apparently the code added to main/features by jpeeler were not included in 1.6.0. ........ 2008-08-01 19:30 +0000 [r135198] Sean Bright * channels/chan_mgcp.c, /: Merged revisions 135197 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r135197 | seanbright | 2008-08-01 15:29:26 -0400 (Fri, 01 Aug 2008) | 6 lines Remove some code that used to do something but does not anymore, mainly to get rid of a shadow warning (but this seemed legitimate enough to fix here instead of in my branch). Thanks to putnopvut for taking a look as well. ........ 2008-08-01 17:10 +0000 [r135127-135129] Tilghman Lesher * /, channels/chan_sip.c: Merged revisions 135128 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r135128 | tilghman | 2008-08-01 12:09:50 -0500 (Fri, 01 Aug 2008) | 2 lines Picky, picky, buildbot ........ * /, channels/chan_sip.c, configs/sip.conf.sample: Merged revisions 135126 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r135126 | tilghman | 2008-08-01 11:39:51 -0500 (Fri, 01 Aug 2008) | 9 lines SIP should use the transport type set in the Moved Temporarily for the next invite. (closes issue #11843) Reported by: pestermann Patches: 20080723__issue11843_302_ignores_transport_16branch.diff uploaded by bbryant (license 36) 20080723__issue11843_302_ignores_transport_trunk.diff uploaded by bbryant (license 36) Tested by: pabelanger ........ 2008-08-01 14:43 +0000 [r135070] Mark Michelson * apps/app_voicemail.c, /, configs/voicemail.conf.sample: Merged revisions 135067-135068 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r135067 | mmichelson | 2008-08-01 09:29:48 -0500 (Fri, 01 Aug 2008) | 13 lines IMAP storage functioned under the assumption that folders such as "Work" and "Family" would be subfolders of the INBOX. This is an invalid assumption to make, but it could be desirable to set up folders in this manner, so a new option for voicemail.conf, "imapparentfolder" has been added to allow for this. (closes issue #13142) Reported by: jaroth Patches: parentfolder.patch uploaded by jaroth (license 50) ........ r135068 | mmichelson | 2008-08-01 09:42:24 -0500 (Fri, 01 Aug 2008) | 3 lines IMAP-specific items must go in IMAP_STORAGE defines... ........ 2008-08-01 12:18 +0000 [r135057-135062] Michiel van Baak * /, apps/app_ices.c: Merged revisions 135059 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r135059 | mvanbaak | 2008-08-01 13:47:34 +0200 (Fri, 01 Aug 2008) | 10 lines Merged revisions 135058 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r135058 | mvanbaak | 2008-08-01 13:43:46 +0200 (Fri, 01 Aug 2008) | 2 lines make app_ices compile on OpenBSD. ........ ................ * /, channels/chan_skinny.c: Merged revisions 135056 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r135056 | mvanbaak | 2008-08-01 13:00:13 +0200 (Fri, 01 Aug 2008) | 16 lines Merged revisions 135055 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r135055 | mvanbaak | 2008-08-01 12:55:27 +0200 (Fri, 01 Aug 2008) | 8 lines fix some potential deadlocks in chan_skinny (closes issue #13215) Reported by: qwell Patches: 2008080100_bug13215.diff.txt uploaded by mvanbaak (license 7) Tested by: mvanbaak ........ ................ 2008-07-31 22:34 +0000 [r135034] Kevin P. Fleming * /, main/http.c: Merged revisions 135016 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r135016 | kpfleming | 2008-07-31 17:28:42 -0500 (Thu, 31 Jul 2008) | 11 lines Merged revisions 134983 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r134983 | kpfleming | 2008-07-31 17:18:11 -0500 (Thu, 31 Jul 2008) | 3 lines accomodate users who seem to lack a sense of humor :-) ........ ................ 2008-07-31 21:58 +0000 [r134926-134981] Tilghman Lesher * sample.call, main/manager.c, pbx/pbx_spool.c, /: Merged revisions 134980 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r134980 | tilghman | 2008-07-31 16:55:42 -0500 (Thu, 31 Jul 2008) | 16 lines Blocked revisions 134976 via svnmerge ........ r134976 | tilghman | 2008-07-31 16:53:19 -0500 (Thu, 31 Jul 2008) | 9 lines Specify codecs in callfiles and manager, to allow video calls to be set up from callfiles and AMI. (closes issue #9531) Reported by: Geisj Patches: 20080715__bug9531__1.4.diff.txt uploaded by Corydon76 (license 14) 20080715__bug9531__1.6.0.diff.txt uploaded by Corydon76 (license 14) Tested by: Corydon76 ........ ................ * res/res_config_sqlite.c, /: Merged revisions 134977 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r134977 | tilghman | 2008-07-31 16:53:59 -0500 (Thu, 31 Jul 2008) | 2 lines Switch command order, to meet with current specs ........ 2008-07-31 19:54 +0000 [r134923] Steve Murphy * /, main/features.c: Merged revisions 134922 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r134922 | murf | 2008-07-31 13:48:08 -0600 (Thu, 31 Jul 2008) | 63 lines Merged revisions 134883 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r134883 | murf | 2008-07-31 13:23:42 -0600 (Thu, 31 Jul 2008) | 51 lines (closes issue #11849) Reported by: greyvoip Tested by: murf OK, a few days of debugging, a bunch of instrumentation in chan_sip, main/channel.c, main/pbx.c, etc. and 5 solid notebook pages of notes later, I have made the small tweek necc. to get the start time right on the second CDR when: A Calls B B answ. A hits Xfer button on sip phone, A dials C and hits the OK button, A hangs up C answers ringing phone B and C converse B and/or C hangs up But does not harm the scenario where: A Calls B B answ. B hits xfer button on sip phone, B dials C and hits the OK button, B hangs up C answers ringing phone A and C converse A and/or C hangs up The difference in start times on the second CDR is because of a Masquerade on the B channel when the xfer number is sent. It ends up replacing the CDR on the B channel with a duplicate, which ends up getting tossed out. We keep a pointer to the first CDR, and update *that* after the bridge closes. But, only if the CDR has changed. I hope this change is specific enough not to muck up any current CDR-based apps. In my defence, I assert that the previous information was wrong, and this change fixes it, and possibly other similar scenarios. I wonder if I should be doing the same thing for the channel, as I did for the peer, but I can't think of a scenario this might affect. I leave it, then, as an exersize for the users, to find the scenario where the chan's CDR changes and loses the proper start time. ........ ................ 2008-07-31 19:41 +0000 [r134918] Russell Bryant * /, apps/app_ices.c: Merged revisions 134917 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r134917 | russell | 2008-07-31 14:39:50 -0500 (Thu, 31 Jul 2008) | 17 lines Merged revisions 134915 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r134915 | russell | 2008-07-31 14:37:26 -0500 (Thu, 31 Jul 2008) | 9 lines Get app_ices working again (closes issue #12981) Reported by: dlogan Patches: 20080709__app_ices_v2_update_trunk.diff uploaded by bbryant (license 36) 20080709__app_ices_v2_update_14.diff uploaded by bbryant (license 36) Tested by: bbryant ........ ................ 2008-07-31 16:53 +0000 [r134816] Russell Bryant * channels/iax2-parser.c: Merged revisions 134815 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r134815 | russell | 2008-07-31 11:50:10 -0500 (Thu, 31 Jul 2008) | 15 lines Merged revisions 134814 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r134814 | russell | 2008-07-31 11:45:31 -0500 (Thu, 31 Jul 2008) | 7 lines In case we have some processing threads that free more frames than they allocate, do not let the frame cache grow forever. (closes issue #13160) Reported by: tavius Tested by: tavius, russell ........ ................ 2008-07-31 16:07 +0000 [r134760] Mark Michelson * /, apps/app_queue.c: Merged revisions 134759 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r134759 | mmichelson | 2008-07-31 11:05:12 -0500 (Thu, 31 Jul 2008) | 24 lines Merged revisions 134758 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r134758 | mmichelson | 2008-07-31 10:56:18 -0500 (Thu, 31 Jul 2008) | 16 lines Add more timeout checks into app_queue, specifically targeting areas where an unknown and potentially long time has just elapsed. Also added a check to try_calling() to return early if the timeout has elapsed instead of potentially setting a negative timeout for the call (thus making it have *no* timeout at all). (closes issue #13186) Reported by: miquel_cabrespina Patches: 13186.diff uploaded by putnopvut (license 60) Tested by: miquel_cabrespina ........ ................ 2008-07-30 22:41 +0000 [r134651-134707] Tilghman Lesher * main/sched.c, /, include/asterisk/sched.h: Merged revisions 134703 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r134703 | tilghman | 2008-07-30 17:38:58 -0500 (Wed, 30 Jul 2008) | 2 lines Oops, wrong define ........ * /, configure, configure.ac: Merged revisions 134650 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r134650 | tilghman | 2008-07-30 16:40:08 -0500 (Wed, 30 Jul 2008) | 12 lines Merged revisions 134649 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r134649 | tilghman | 2008-07-30 16:38:50 -0500 (Wed, 30 Jul 2008) | 4 lines Qwell pointed out, via IRC, that the previous fix only worked when explicitly set. When nothing is set, and the option is implied, it breaks, because configure sets the prefix to 'NONE'. Fixing. ........ ................ 2008-07-30 21:06 +0000 [r134599] Mark Michelson * /, apps/app_queue.c: Merged revisions 134598 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r134598 | mmichelson | 2008-07-30 16:05:37 -0500 (Wed, 30 Jul 2008) | 15 lines Merged revisions 134556 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r134556 | mmichelson | 2008-07-30 15:24:40 -0500 (Wed, 30 Jul 2008) | 7 lines Fix the parsing of the "reason" parameter in the Diversion: header. (closes issue #13195) Reported by: woodsfsg ........ ................ 2008-07-30 20:39 +0000 [r134597] Russell Bryant * /, pbx/pbx_dundi.c: Merged revisions 134596 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r134596 | russell | 2008-07-30 15:38:35 -0500 (Wed, 30 Jul 2008) | 14 lines Merged revisions 134595 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r134595 | russell | 2008-07-30 15:37:17 -0500 (Wed, 30 Jul 2008) | 6 lines Reduce stack consumption by 12.5% of the max stack size to fix a crash when compiled with LOW_MEMORY. (closes issue #13154) Reported by: edantie ........ ................ 2008-07-30 20:25 +0000 [r134561] Mark Michelson * /, channels/chan_sip.c: Merged revisions 134556 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r134556 | mmichelson | 2008-07-30 15:24:40 -0500 (Wed, 30 Jul 2008) | 7 lines Fix the parsing of the "reason" parameter in the Diversion: header. (closes issue #13195) Reported by: woodsfsg ........ 2008-07-30 19:56 +0000 [r134542] Russell Bryant * funcs/func_curl.c, /: Merged revisions 134541 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r134541 | russell | 2008-07-30 14:55:31 -0500 (Wed, 30 Jul 2008) | 12 lines Merged revisions 134540 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r134540 | russell | 2008-07-30 14:52:53 -0500 (Wed, 30 Jul 2008) | 4 lines Fix a memory leak in func_curl. Every thread that used this function leaked an allocation the size of a pointer. (reported by jmls in #asterisk-dev) ........ ................ 2008-07-30 19:49 +0000 [r134482-134539] Tilghman Lesher * /, configure, configure.ac: Merged revisions 134538 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r134538 | tilghman | 2008-07-30 14:48:37 -0500 (Wed, 30 Jul 2008) | 12 lines Merged revisions 134536 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r134536 | tilghman | 2008-07-30 14:47:16 -0500 (Wed, 30 Jul 2008) | 4 lines Only override sysconfdir and mandir when prefix=/usr (closes issue #13093) Reported by: pabelanger ........ ................ * /, apps/app_queue.c: Merged revisions 134483 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r134483 | tilghman | 2008-07-30 14:17:38 -0500 (Wed, 30 Jul 2008) | 4 lines Let "roundrobin" also reference rrmemory, for the 1.6 release (as described in UPGRADE-1.4.txt) (Closes issue #13181) ........ * /, res/res_agi.c: Merged revisions 134481 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r134481 | tilghman | 2008-07-30 14:05:35 -0500 (Wed, 30 Jul 2008) | 13 lines Merged revisions 134480 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r134480 | tilghman | 2008-07-30 14:03:44 -0500 (Wed, 30 Jul 2008) | 5 lines launch_netscript sometimes returns -1, which fails to set AGISTATUS. Map failure to -1, so that AGISTATUS is always set. (closes issue #13199) Reported by: smw1218 ........ ................ 2008-07-30 18:33 +0000 [r134477] Mark Michelson * /, main/app.c: Merged revisions 134476 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r134476 | mmichelson | 2008-07-30 13:33:12 -0500 (Wed, 30 Jul 2008) | 12 lines Merged revisions 134475 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r134475 | mmichelson | 2008-07-30 13:31:47 -0500 (Wed, 30 Jul 2008) | 4 lines Fix a spot where a function could return without bringing a channel out of autoservice. ........ ................ 2008-07-30 15:34 +0000 [r134356] Kevin P. Fleming * Makefile, /: Merged revisions 134355 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r134355 | kpfleming | 2008-07-30 10:32:14 -0500 (Wed, 30 Jul 2008) | 10 lines Merged revisions 134352 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r134352 | kpfleming | 2008-07-30 10:29:17 -0500 (Wed, 30 Jul 2008) | 2 lines use the proper method for building version.h ........ ................ 2008-07-29 22:29 +0000 [r134283] Kevin P. Fleming * apps/app_rpt.c, apps/app_dahdibarge.c, channels/chan_dahdi.c, /, apps/app_meetme.c, apps/app_dahdiscan.c, apps/app_dahdiras.c: Merged revisions 134260 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r134260 | kpfleming | 2008-07-29 17:22:13 -0500 (Tue, 29 Jul 2008) | 2 lines build against the now-typedef-free dahdi/user.h, and remove some #ifdefs for features that will always be present in DAHDI ........ 2008-07-28 22:16 +0000 [r134164] Tilghman Lesher * apps/app_voicemail.c, /: Merged revisions 134163 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r134163 | tilghman | 2008-07-28 17:07:12 -0500 (Mon, 28 Jul 2008) | 15 lines Merged revisions 134161 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r134161 | tilghman | 2008-07-28 16:50:50 -0500 (Mon, 28 Jul 2008) | 7 lines Detect when sox fails to raise the volume, because sox can't read the file. (closes issue #12939) Reported by: rickbradley Patches: 20080728__bug12939.diff.txt uploaded by Corydon76 (license 14) Tested by: rickbradley ........ ................ 2008-07-28 19:55 +0000 [r134126] Mark Michelson * /, configure, main/Makefile, configure.ac, CHANGES: Merged revisions 134125 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r134125 | mmichelson | 2008-07-28 14:53:56 -0500 (Mon, 28 Jul 2008) | 27 lines This commit compensates for buggy poll(2) implementations. Asterisk has, for a long time, had its own implementation of poll(2) which just used the input arguments to call select(2). In 1.4, this internal implementation was used for Darwin systems. This was removed in Asterisk trunk at some point, but it seems as though this was not the right move to make. On Mac OS X, it appears as though the poll used to gather CLI input does not respond properly when connecting via a remote Asterisk console. Reverting to the use of Asterisk's poll fixed the issue. Also, there is now an option for the configure script, --enable-internal-poll, which will allow for anyone to use Asterisk's internal poll implementation in case they suspect that their system's poll implementation is buggy. closes issue #11928) Reported by: adriavidal Patches: 1.6.0-configurev2.patch uploaded by putnopvut (license 60) ........ 2008-07-28 16:49 +0000 [r134087] Kevin P. Fleming * apps/app_parkandannounce.c, main/loader.c, sample.call, contrib/scripts/autosupport, build_tools/cflags.xml, main/channel.c, apps/app_dahdibarge.c, channels/chan_dahdi.c, configs/chan_dahdi.conf.sample, doc/ss7.txt, /, main/features.c, doc/osp.txt, main/file.c, pbx/pbx_config.c: Merged revisions 134086 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r134086 | kpfleming | 2008-07-28 11:42:00 -0500 (Mon, 28 Jul 2008) | 3 lines remove remaining Zaptel references in various places ........ 2008-07-28 16:13 +0000 [r134052] Mark Michelson * apps/app_chanspy.c, include/asterisk/options.h, main/asterisk.c, /, apps/app_meetme.c, apps/app_dahdiscan.c: Merged revisions 134050 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r134050 | mmichelson | 2008-07-28 11:00:19 -0500 (Mon, 28 Jul 2008) | 3 lines merging the zap_and_dahdi_trunk branch up to trunk ........ 2008-07-26 15:34 +0000 [r133942-133982] Russell Bryant * main/asterisk.c, include/asterisk/doxyref.h, /: Include the licensing page in 1.6.0 as well. Now, this page exists in 1.4, trunk, and 1.6.0. * /: unblock 133575 * /, main/devicestate.c: Merged revisions 133945-133946 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r133945 | russell | 2008-07-26 10:15:14 -0500 (Sat, 26 Jul 2008) | 6 lines ast_device_state() gets called in two different ways. The first way is when called from elsewhere in Asterisk to find the current state of a device. In that case, we want to use the cached value if it exists. The other way is when processing a device state change. In that case, we do not want to check the cache because returning the last known state is counter productive. ........ r133946 | russell | 2008-07-26 10:16:20 -0500 (Sat, 26 Jul 2008) | 1 line actually use the cache_cache argument ........ 2008-07-25 22:09 +0000 [r133863-133905] Tilghman Lesher * contrib/scripts/asterisk.ldif, /: Merged revisions 133902 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r133902 | tilghman | 2008-07-25 16:59:39 -0500 (Fri, 25 Jul 2008) | 6 lines Update version (closes issue #13163) Reported by: suretec Patches: asterisk.ldif uploaded by suretec (license 70) ........ 2008-07-25 19:37 +0000 [r133804-133806] Brandon Kruse * /: Blocking revert of code changes in r133770 * main/http.c: Include the http_decode function from trunk to replace the + with a space. 2008-07-25 17:33 +0000 [r133694] Brandon Kruse * /: Blocking a fix from trunk for the function http_decode. 1.6.0 does not have this function. 2008-07-25 17:26 +0000 [r133671] Tilghman Lesher * main/channel.c, /, channels/chan_agent.c, main/devicestate.c: Merged revisions 133665 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r133665 | tilghman | 2008-07-25 12:24:43 -0500 (Fri, 25 Jul 2008) | 16 lines Merged revisions 133649 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r133649 | tilghman | 2008-07-25 12:19:39 -0500 (Fri, 25 Jul 2008) | 8 lines Fix some errant device states by making the devicestate API more strict in terms of the device argument (only without the unique identifier appended). (closes issue #12771) Reported by: davidw Patches: 20080717__bug12771.diff.txt uploaded by Corydon76 (license 14) Tested by: davidw, jvandal, murf ........ ................ 2008-07-25 15:01 +0000 [r133576-133580] Russell Bryant * /, LICENSE: Merged revisions 133579 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r133579 | russell | 2008-07-25 10:00:49 -0500 (Fri, 25 Jul 2008) | 18 lines Merged revisions 133578 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r133578 | russell | 2008-07-25 10:00:31 -0500 (Fri, 25 Jul 2008) | 10 lines Merged revisions 133577 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r133577 | russell | 2008-07-25 10:00:13 -0500 (Fri, 25 Jul 2008) | 2 lines Fix the IAX2 URI for calling Digium ........ ................ ................ 2008-07-25 14:41 +0000 [r133571-133574] Mark Michelson * /, channels/chan_sip.c: Merged revisions 133573 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r133573 | mmichelson | 2008-07-25 09:40:52 -0500 (Fri, 25 Jul 2008) | 15 lines Merged revisions 133572 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r133572 | mmichelson | 2008-07-25 09:40:10 -0500 (Fri, 25 Jul 2008) | 7 lines We need to make sure to null-terminate the "name" portion of SIP URI parameters so that there are no bogus comparisons. Thanks to bbryant for pointing this out. ........ ................ 2008-07-25 13:25 +0000 [r133567-133569] Russell Bryant * /, channels/chan_sip.c: Merged revisions 133568 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r133568 | russell | 2008-07-25 08:01:59 -0500 (Fri, 25 Jul 2008) | 4 lines Minor coding guidelines tweaks ... - Use ast_strlen_zero in one place - check for successful string comparison the way most of Asterisk code does it ........ 2008-07-24 21:31 +0000 [r133524] Tilghman Lesher * /, channels/chan_sip.c: Merged revisions 133509 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r133509 | tilghman | 2008-07-24 16:27:06 -0500 (Thu, 24 Jul 2008) | 11 lines Merged revisions 133488 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r133488 | tilghman | 2008-07-24 16:17:55 -0500 (Thu, 24 Jul 2008) | 3 lines Fix rtautoclear and rtcachefriends (Closes issue #12707) ........ ................ 2008-07-24 20:41 +0000 [r133487] Russell Bryant * /, channels/chan_agent.c: Merged revisions 133486 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r133486 | russell | 2008-07-24 15:40:15 -0500 (Thu, 24 Jul 2008) | 3 lines I made this change from DEVICE_STATE to DEVICE_STATE_CHANGE, but I had it backwards, this is the right event to subscribe to ... ........ 2008-07-24 19:55 +0000 [r133449] Mark Michelson * /, main/logger.c: Merged revisions 133448 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r133448 | mmichelson | 2008-07-24 14:53:37 -0500 (Thu, 24 Jul 2008) | 12 lines Print the correct PID in log messages. Prior to this commit, only the logger thread's PID would be printed. (closes issue #13150) Reported by: atis Patches: log_pid.diff uploaded by putnopvut (license 60) Tested by: eliel ........ 2008-07-24 05:21 +0000 [r133392-133405] Tilghman Lesher * contrib/scripts/asterisk.logrotate, Makefile, /: Merged revisions 133400 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r133400 | tilghman | 2008-07-24 00:21:00 -0500 (Thu, 24 Jul 2008) | 3 lines Build the logrotate script according to paths (Closes issue #13147) ........ * Makefile, /: Merged revisions 133391 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r133391 | tilghman | 2008-07-23 23:51:42 -0500 (Wed, 23 Jul 2008) | 3 lines Optionally install logrotate file (Closes issue #13148) ........ 2008-07-23 22:07 +0000 [r133300] Steve Murphy * main/pbx.c, /: Merged revisions 133299 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r133299 | murf | 2008-07-23 16:03:48 -0600 (Wed, 23 Jul 2008) | 27 lines (closes issue #13144) Reported by: murf Tested by: murf For: J. Geis The 'data' field in the ast_exten struct was being 'moved' from the current dialplan to the replacement dialplan. This was not good, as the current dialplan could have problems in the time between the change and when the new dialplan is swapped in. So, I modified the merge_and_delete code to strdup the 'data' field (the args to the app call), and then it's freed as normal. I improved a few messages; I added code to limit the number of calls to the context_merge_incls_swits_igps_other_registrars() to one per context. I don't think having it called multiple times per context was doing anything bad, but it was inefficient. I hope this fixes the problems Mr. Geiss was noting in asterisk-users, see http://lists.digium.com/pipermail/asterisk-users/2008-July/215634.html ........ 2008-07-23 21:50 +0000 [r133297] Jason Parker * channels/chan_dahdi.c, /: Merged revisions 133296 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r133296 | qwell | 2008-07-23 16:50:20 -0500 (Wed, 23 Jul 2008) | 9 lines Merged revisions 133295 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r133295 | qwell | 2008-07-23 16:49:03 -0500 (Wed, 23 Jul 2008) | 1 line inbandrelease is gone - it's now inbanddisconnect ........ ................ 2008-07-23 20:39 +0000 [r133218] Brett Bryant * /, channels/chan_sip.c: Merged revisions 133197 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r133197 | bbryant | 2008-07-23 15:33:22 -0500 (Wed, 23 Jul 2008) | 2 lines Fix issue where tcp in sip is enabled by default, despite what it says in the config sample file. Also fix "sip show settings" for tcp connections. ........ 2008-07-23 19:50 +0000 [r133042-133172] Mark Michelson * apps/app_chanspy.c, include/asterisk/options.h, main/asterisk.c, /: Merged revisions 133171 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r133171 | mmichelson | 2008-07-23 14:48:03 -0500 (Wed, 23 Jul 2008) | 20 lines Merged revisions 133169 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r133169 | mmichelson | 2008-07-23 14:39:47 -0500 (Wed, 23 Jul 2008) | 12 lines As suggested by seanbright, the PSEUDO_CHAN_LEN in app_chanspy should be set at load time, not at compile time, since dahdi_chan_name is determined at load time. Also changed the next_unique_id_to_use to have the static qualifier. Also added the dahdi_chan_name_len variable so that strlen(dahdi_chan_name) isn't necessary. Thanks to seanbright for the suggestion. ........ ................ * apps/app_chanspy.c, /: Merged revisions 133106 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r133106 | mmichelson | 2008-07-23 14:07:56 -0500 (Wed, 23 Jul 2008) | 13 lines Merged revisions 133104 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r133104 | mmichelson | 2008-07-23 14:06:16 -0500 (Wed, 23 Jul 2008) | 5 lines Zap/pseudo is ten characters, but DAHDI/pseudo is twelve. The strncmp call in next_channel should account for this. ........ ................ * apps/app_chanspy.c, /: Merged revisions 133102 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r133102 | mmichelson | 2008-07-23 13:58:37 -0500 (Wed, 23 Jul 2008) | 14 lines Merged revisions 133101 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r133101 | mmichelson | 2008-07-23 13:57:17 -0500 (Wed, 23 Jul 2008) | 6 lines Update the "last" channel in next_channel in app_chanspy so that the same pseudo channel isn't constantly returned. related to issue #13124 ........ ................ * channels/chan_dahdi.c, /: Merged revisions 133041 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r133041 | mmichelson | 2008-07-23 12:54:03 -0500 (Wed, 23 Jul 2008) | 15 lines Merged revisions 133038 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r133038 | mmichelson | 2008-07-23 12:50:01 -0500 (Wed, 23 Jul 2008) | 7 lines Small cleanup. Move the declaration of the DAHDI_SPANINFO variable to the block where it is used. This allows one less #ifdef HAVE_PRI to clutter things up. Thanks to Tzafrir for pointing this out on #asterisk-dev ........ ................ 2008-07-23 17:21 +0000 [r132978-132983] Tilghman Lesher * /, channels/chan_iax2.c: Merged revisions 132981 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r132981 | tilghman | 2008-07-23 12:20:43 -0500 (Wed, 23 Jul 2008) | 6 lines Yet another conversion of '|' to ',' (closes issue #13137) Reported by: eliel Patches: chan_iax2trunk-IAXPEER.patch uploaded by eliel (license 64) ........ * contrib/scripts/asterisk.logrotate (added), /: Merged revisions 132977 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r132977 | tilghman | 2008-07-23 12:14:56 -0500 (Wed, 23 Jul 2008) | 6 lines Add logrotate script for Asterisk (closes issue #13085) Reported by: pabelanger Patches: logrotate uploaded by pabelanger (license 224) ........ 2008-07-23 16:42 +0000 [r132965-132967] Kevin P. Fleming * channels/misdn/isdn_lib.c, /: Merged revisions 132883,132966 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r132883 | crichter | 2008-07-23 07:07:15 -0500 (Wed, 23 Jul 2008) | 9 lines Merged revisions 132826 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r132826 | crichter | 2008-07-23 13:37:50 +0200 (Mi, 23 Jul 2008) | 1 line another Fix because of r119585, this commit has broken high frequented BRI Ports, there was a possibility that a channel, that was marked as in_use would be reused later, the corresponding port could got stuck then. So it is recommended to upgrade for chan_misdn users. ........ ................ r132966 | kpfleming | 2008-07-23 11:38:28 -0500 (Wed, 23 Jul 2008) | 2 lines use correct function name... please compile with --enable-dev-mode ................ * include/asterisk/stringfields.h, /, main/utils.c: Merged revisions 132964 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r132964 | kpfleming | 2008-07-23 11:30:18 -0500 (Wed, 23 Jul 2008) | 10 lines Merged revisions 132872 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r132872 | kpfleming | 2008-07-23 06:52:18 -0500 (Wed, 23 Jul 2008) | 2 lines minor optimization for stringfields: when a field is being set to a larger value than it currently contains and it happens to be the most recent field allocated from the currentl pool, it is possible to 'grow' it without having to waste the space it is currently using (or potentially even allocate a new pool) ........ ................ 2008-07-23 08:18 +0000 [r132824] Olle Johansson * /, channels/chan_sip.c: Merged revisions 132823 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r132823 | oej | 2008-07-23 10:13:07 +0200 (Ons, 23 Jul 2008) | 8 lines Well, the content of a channel variable may be longer than the size of a pointer... Thanks, eliel! Reported by: eliel Patches: chan_siptrunk.SIPPEER.patch uploaded by eliel (license 64) (closes issue #13135) ........ 2008-07-22 22:20 +0000 [r132797] Mark Michelson * /, channels/chan_sip.c: Merged revisions 132795 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r132795 | mmichelson | 2008-07-22 17:17:09 -0500 (Tue, 22 Jul 2008) | 11 lines Merged revisions 132777 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ Allow Spiraled INVITEs to work correctly within Asterisk. Prior to this change, a spiraled INVITE would cause a 482 Loop Detected to be sent to the caller. With this change, if a potential loop is detected, the Request-URI is inspected to see if it has changed from what was originally received. If pedantic mode is on, then this inspection is fully RFC 3261 compliant. If pedantic mode is not on, then a string comparison is used to test the equality of the two R-URIs. This has been tested by using OpenSER to rewrite the R-URI and send the INVITE back to Asterisk. (closes issue #7403) Reported by: stephen_dredge Modified: branches/1.4/channels/chan_sip.c ........ ................ 2008-07-22 22:15 +0000 [r132793] Kevin P. Fleming * channels/chan_dahdi.c, /: Merged revisions 132791 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r132791 | kpfleming | 2008-07-22 17:14:37 -0500 (Tue, 22 Jul 2008) | 2 lines correct fix made in r132777... the code *did* compile in dev-mode, as long as libpri was installed and enabled ........ 2008-07-22 21:59 +0000 [r132782] Olle Johansson * /, channels/chan_sip.c, doc/sip-retransmit.txt (added): Merged revisions 132703 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r132703 | oej | 2008-07-22 22:46:11 +0200 (Tis, 22 Jul 2008) | 17 lines Merged revisions 132645 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r132645 | oej | 2008-07-22 22:10:26 +0200 (Tis, 22 Jul 2008) | 9 lines The most common question on the #asterisk iRC channel and on mailing lists seems to be in regards to an error message when retransmit fails. This is frequently misunderstood as a failure of Asterisk, not a failure of the network to reach the other party. This document tries to assist the Asterisk user in sorting out these issues by explaining the logic and pointing at some possible causes. Hopefully, we will get other questions now :-) ........ ................ 2008-07-22 21:55 +0000 [r132780] Tilghman Lesher * configs/iax.conf.sample, /, channels/chan_iax2.c: Merged revisions 132778 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r132778 | tilghman | 2008-07-22 16:53:40 -0500 (Tue, 22 Jul 2008) | 18 lines Merged revisions 132713 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r132713 | tilghman | 2008-07-22 16:19:39 -0500 (Tue, 22 Jul 2008) | 10 lines Merged revisions 132711 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r132711 | tilghman | 2008-07-22 16:14:10 -0500 (Tue, 22 Jul 2008) | 2 lines Fixes for AST-2008-010 and AST-2008-011 ........ ................ ................ 2008-07-22 21:54 +0000 [r132779] Mark Michelson * channels/chan_dahdi.c, /: Merged revisions 132777 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r132777 | mmichelson | 2008-07-22 16:52:24 -0500 (Tue, 22 Jul 2008) | 3 lines Get chan_dahdi to compile in devmode ........ 2008-07-22 21:23 +0000 [r132574-132729] Kevin P. Fleming * channels/chan_dahdi.c, /: Merged revisions 132721 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r132721 | kpfleming | 2008-07-22 16:21:56 -0500 (Tue, 22 Jul 2008) | 14 lines Merged revisions 132712 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r132712 | kpfleming | 2008-07-22 16:17:23 -0500 (Tue, 22 Jul 2008) | 6 lines ensure that if any alarms exist at channel creation time, they are handled identically to if they occurred later, so that later alarm clearing will work properly and 'make sense' (closes issue #12160) Reported by: tzafrir ........ ................ * /, configure, configure.ac, acinclude.m4: Merged revisions 132705 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r132705 | kpfleming | 2008-07-22 15:54:07 -0500 (Tue, 22 Jul 2008) | 10 lines Merged revisions 132704 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r132704 | kpfleming | 2008-07-22 15:49:41 -0500 (Tue, 22 Jul 2008) | 2 lines make AST_C_COMPILE_CHECK able to print a 'pretty' description of what it is doing ........ ................ * channels/chan_dahdi.c, configs/chan_dahdi.conf.sample, /, configure, include/asterisk/autoconfig.h.in, configure.ac: Merged revisions 132643 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r132643 | kpfleming | 2008-07-22 14:59:10 -0500 (Tue, 22 Jul 2008) | 10 lines Merged revisions 132641 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r132641 | kpfleming | 2008-07-22 14:49:11 -0500 (Tue, 22 Jul 2008) | 2 lines use renamed libpri API call for controlling this feature (was improperly named before) ........ ................ * channels/chan_dahdi.c, /: Merged revisions 132573 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r132573 | kpfleming | 2008-07-21 17:51:16 -0500 (Mon, 21 Jul 2008) | 10 lines Merged revisions 132571 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r132571 | kpfleming | 2008-07-21 17:45:16 -0500 (Mon, 21 Jul 2008) | 2 lines teach chan_dahdi how to find the D-channel on BRI spans, and don't attempt to use channel 24 as a D-channel on spans of unexpected sizes ........ ................ 2008-07-21 21:13 +0000 [r132515] Brett Bryant * configs/features.conf.sample, configs/gtalk.conf.sample, /, configs/jingle.conf.sample, configs/manager.conf.sample: Merged revisions 132514 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r132514 | bbryant | 2008-07-21 16:12:51 -0500 (Mon, 21 Jul 2008) | 8 lines Update configuration files to add missing options for jingle, gtalk, manager.conf, and features.conf. (closes issue #13128) Reported by: caio1982 Patches: missing_options1.diff uploaded by caio1982 (license 22) ........ 2008-07-21 21:02 +0000 [r132512-132513] Tilghman Lesher * main/fskmodem.c (added), /, include/asterisk/fskmodem.h (added): Merged revisions 132511 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r132511 | tilghman | 2008-07-21 16:00:47 -0500 (Mon, 21 Jul 2008) | 2 lines (Step 2 of 2) ........ * main/fskmodem.c (removed), include/asterisk/fskmodem_int.h (added), build_tools/cflags.xml, main/fskmodem_float.c (added), /, main/tdd.c, include/asterisk/fskmodem.h (removed), main/fskmodem_int.c (added), main/callerid.c, include/asterisk/fskmodem_float.h (added): Merged revisions 132510 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r132510 | tilghman | 2008-07-21 15:59:03 -0500 (Mon, 21 Jul 2008) | 5 lines Optionally build integer-based routines for FSK tone decoding (but default to the more accurate float-based routines). (Closes issue #11679) (Step 1 of 2) ........ 2008-07-21 20:55 +0000 [r132467-132509] Brett Bryant * /, apps/app_sendtext.c: Merged revisions 132508 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r132508 | bbryant | 2008-07-21 15:54:09 -0500 (Mon, 21 Jul 2008) | 9 lines Fix a bug where SENDTEXTSTATUS isn't set properly when it isn't supported on a channel (yet _another_ useful patch by eliel). (closes issue #13081) Reported by: eliel Patches: app_sendtext.c.patch uploaded by eliel (license 64) Tested by: eliel ........ * /, channels/chan_sip.c: Merged revisions 132468 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r132468 | bbryant | 2008-07-21 12:42:45 -0500 (Mon, 21 Jul 2008) | 5 lines Fix bug where ast_parse_arg would inadvertantly enable sip tcp when parsing a tcpbindaddr if it was disabled. (closes issue #13117) Reported by: pj ........ * /, channels/chan_iax2.c: Merged revisions 132466 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r132466 | bbryant | 2008-07-21 12:22:02 -0500 (Mon, 21 Jul 2008) | 3 lines Fix an issue in iax2 where a call that's been rejected still kept an open channel on the side that attempted to make the call (not the side of the call that rejected the call). Changes were load tested and also approved by Russell. ........ 2008-07-21 15:34 +0000 [r132426] Jeff Peeler * channels/chan_dahdi.c, /: Merged revisions 132425 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r132425 | jpeeler | 2008-07-21 10:33:13 -0500 (Mon, 21 Jul 2008) | 2 lines make buffers config option (chan_dahdi.conf) parsing safer and added logging in case of failure ........ 2008-07-21 14:48 +0000 [r132389-132391] Russell Bryant * apps/app_jack.c, include/asterisk/libresample.h (removed), /, build_tools/menuselect-deps.in, configure, include/asterisk/autoconfig.h.in, main/Makefile, main/libresample (removed), configure.ac, codecs/codec_resample.c, makeopts.in: Merged revisions 132390 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r132390 | russell | 2008-07-21 09:47:41 -0500 (Mon, 21 Jul 2008) | 16 lines Remove libresample from the Asterisk source tree. It is now available in its own repository, and must be installed like any other library for Asterisk to use. The two modules that require it are codec_resample and app_jack. To install libresample: $ svn co http://svn.digium.com/svn/libresample/trunk libresample $ cd libresample $ ./configure $ make $ sudo make install This code is currently in our own repository because the build system did not include the appropriate targets for building a dynamic library or for installing the library. ........ * apps/app_jack.c, /, codecs/codec_resample.c: Merged revisions 132388 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r132388 | russell | 2008-07-21 08:51:05 -0500 (Mon, 21 Jul 2008) | 3 lines Enable higher quality resampling, as it doesn't have a noticeable performance impact on my machine .. ........ 2008-07-19 16:47 +0000 [r132313] Kevin P. Fleming * /, LICENSE: Merged revisions 132312 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r132312 | kpfleming | 2008-07-19 11:46:33 -0500 (Sat, 19 Jul 2008) | 10 lines Merged revisions 132311 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r132311 | kpfleming | 2008-07-19 11:45:52 -0500 (Sat, 19 Jul 2008) | 2 lines grant a license exception to allow distribution of Asterisk binaries that use the UW IMAP Toolkit (which is licensed under a non-GPL-compatible license) ........ ................ 2008-07-19 10:47 +0000 [r132278] Michiel van Baak * res/res_config_sqlite.c, /: Merged revisions 132277 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r132277 | mvanbaak | 2008-07-19 12:46:12 +0200 (Sat, 19 Jul 2008) | 7 lines fix a couple of comments in sqlite resource driver. (closes issue #13110) Reported by: gknispel_proformatique Patches: res_config_sqlite_comments.patch uploaded by gknispel (license 261) ........ 2008-07-18 22:20 +0000 [r132245] Brett Bryant * main/manager.c, /: Merged revisions 132242 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r132242 | bbryant | 2008-07-18 17:19:56 -0500 (Fri, 18 Jul 2008) | 4 lines Fixes problem where manager users loaded from users.conf would be removed early (before the routine to load the configuration was finished) because a variable wasn't initialized. ........ 2008-07-18 20:58 +0000 [r132114-132207] Tilghman Lesher * /, main/say.c: Merged revisions 132113 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r132113 | tilghman | 2008-07-18 14:09:39 -0500 (Fri, 18 Jul 2008) | 14 lines Merged revisions 132112 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r132112 | tilghman | 2008-07-18 14:06:10 -0500 (Fri, 18 Jul 2008) | 6 lines Fix for Taiwanese number syntax (closes issue #12319) Reported by: CharlesWang Patches: saynumber-tw-1.4.18.1.patch uploaded by CharlesWang (license 444) ........ ................ 2008-07-18 18:53 +0000 [r132111] Matthew Fredrickson * channels/chan_dahdi.c, /: Merged revisions 132108 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r132108 | mattf | 2008-07-18 13:50:00 -0500 (Fri, 18 Jul 2008) | 1 line Make sure we break the poll so that messages queued will be sent on the SS7 when using CLI commands for blocking and blocking of CICs and linksets. ........ 2008-07-18 18:51 +0000 [r132110] Tilghman Lesher * main/config.c, /: Merged revisions 132109 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r132109 | tilghman | 2008-07-18 13:50:37 -0500 (Fri, 18 Jul 2008) | 14 lines Merged revisions 132107 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r132107 | tilghman | 2008-07-18 13:47:50 -0500 (Fri, 18 Jul 2008) | 6 lines Textual clarification (closes issue #13106) Reported by: flefoll Patches: config.c.br14.120173.patch-unknown-directive uploaded by flefoll (license 244) ........ ................ 2008-07-18 17:56 +0000 [r132051] Brett Bryant * main/hashtab.c, /, cdr/cdr_radius.c: Merged revisions 132050 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r132050 | bbryant | 2008-07-18 12:55:41 -0500 (Fri, 18 Jul 2008) | 8 lines Fix magic Revision keywords in hashtab.c and change cdr_radius.c to use the same keyword as the other files (patch by eliel). (closes issue #13104) Reported by: eliel Patches: revision.patch uploaded by eliel (license 64) ........ 2008-07-18 17:40 +0000 [r131984-132047] Tilghman Lesher * main/sched.c, /: Merged revisions 131989 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r131989 | tilghman | 2008-07-18 12:10:34 -0500 (Fri, 18 Jul 2008) | 10 lines Merged revisions 131988 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r131988 | tilghman | 2008-07-18 12:10:01 -0500 (Fri, 18 Jul 2008) | 2 lines Oops ........ ................ * main/sched.c, /, include/asterisk/sched.h: Merged revisions 131986 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r131986 | tilghman | 2008-07-18 11:48:18 -0500 (Fri, 18 Jul 2008) | 10 lines Merged revisions 131985 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r131985 | tilghman | 2008-07-18 11:46:23 -0500 (Fri, 18 Jul 2008) | 2 lines Preserve ABI compatibility with last change ........ ................ * main/sched.c, /, include/asterisk/sched.h, channels/chan_iax2.c: Merged revisions 131982 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r131982 | tilghman | 2008-07-18 11:33:56 -0500 (Fri, 18 Jul 2008) | 10 lines Merged revisions 131970 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r131970 | tilghman | 2008-07-18 11:30:31 -0500 (Fri, 18 Jul 2008) | 2 lines Make the ast_assert call within ast_sched_del report something useful. ........ ................ 2008-07-18 16:16 +0000 [r131924] Kevin P. Fleming * main/dlfcn.c (removed), main/loader.c, /, main/Makefile, include/asterisk/dlfcn-compat.h (removed): Merged revisions 131923 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r131923 | kpfleming | 2008-07-18 11:16:12 -0500 (Fri, 18 Jul 2008) | 10 lines Merged revisions 131921 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r131921 | kpfleming | 2008-07-18 11:15:41 -0500 (Fri, 18 Jul 2008) | 2 lines remove the dlfcn compatibility stuff, because no platforms that Asterisk currently runs on it use it, and it doesn't build anyway ........ ................ 2008-07-18 15:39 +0000 [r131917] Brett Bryant * /, main/features.c: Merged revisions 131916 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r131916 | bbryant | 2008-07-18 10:38:22 -0500 (Fri, 18 Jul 2008) | 12 lines Merged revisions 131915 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r131915 | bbryant | 2008-07-18 10:34:42 -0500 (Fri, 18 Jul 2008) | 4 lines Fix a bug in blind transfers where the BLINDTRANSFER variable isn't always set to the other end of the blind transfer. (closes issue #12586) ........ ................ 2008-07-17 22:45 +0000 [r131869] Jeff Peeler * channels/chan_dahdi.c, /: Merged revisions 131868 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r131868 | jpeeler | 2008-07-17 17:40:00 -0500 (Thu, 17 Jul 2008) | 6 lines Add configuration option to chan_dahdi.conf to allow buffering policy and number of buffers to be configured per channel. Syntax: buffers=, Where the number of buffers is some non-negative integer and the policy is either "full", "half", or "immediate". ........ 2008-07-17 21:27 +0000 [r131830] Mark Michelson * /, apps/app_senddtmf.c: Merged revisions 131824 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r131824 | mmichelson | 2008-07-17 16:26:41 -0500 (Thu, 17 Jul 2008) | 10 lines Document that the duration of dtmf may be passed to the SendDTMF application. Also correct the default pause between digits. (closes issue #13102) Reported by: eliel Patches: app_senddtmf.c.patch uploaded by eliel (license 64) ........ 2008-07-17 20:38 +0000 [r131754-131792] Tilghman Lesher * channels/chan_dahdi.c, /: Merged revisions 131791 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r131791 | tilghman | 2008-07-17 15:37:14 -0500 (Thu, 17 Jul 2008) | 15 lines Merged revisions 131790 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r131790 | tilghman | 2008-07-17 15:35:44 -0500 (Thu, 17 Jul 2008) | 7 lines Revert part of issue #5620 (revision 6965) as it appears that it was in error. This should fix talk call progress on analog lines. (closes issue #12178) Reported by: michael-fig Patches: 20080717__bug12178.diff.txt uploaded by Corydon76 (license 14) ........ ................ * res/res_config_sqlite.c, /: Merged revisions 131753 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r131753 | tilghman | 2008-07-17 13:36:34 -0500 (Thu, 17 Jul 2008) | 6 lines Fix memory leaks (closes issue #13099) Reported by: gknispel_proformatique Patches: res_config_sqlite_leak_on_error.patch uploaded by gknispel (license 261) ........ 2008-07-17 18:15 +0000 [r131718] Brett Bryant * /, main/features.c: Merged revisions 131717 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r131717 | bbryant | 2008-07-17 13:14:42 -0500 (Thu, 17 Jul 2008) | 8 lines Fix a memory leak in register_group_feature when attempting to register a feature without specifying a group or feature to register. (closes issue #13101) Reported by: eliel Patches: features.c.patch uploaded by eliel (license 64) ........ 2008-07-17 15:46 +0000 [r131682] Tilghman Lesher * res/res_config_sqlite.c, /: Merged revisions 131681 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r131681 | tilghman | 2008-07-17 10:45:25 -0500 (Thu, 17 Jul 2008) | 4 lines Fix memory leak. (Closes issue #13096) Reported by gknispel_proformatique ........ 2008-07-16 23:56 +0000 [r131571] Steve Murphy * /: The commit from 131570 should not be applied to 1.6.0, as it is not as necessary, because log_show_lock in trunk is not available in 1.6.0, and is estimated to be the only function that might care about the lock_addr values. 2008-07-16 22:18 +0000 [r131493] Brett Bryant * /, channels/chan_iax2.c: Merged revisions 131492 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r131492 | bbryant | 2008-07-16 17:17:36 -0500 (Wed, 16 Jul 2008) | 14 lines Merged revisions 131491 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r131491 | bbryant | 2008-07-16 17:17:07 -0500 (Wed, 16 Jul 2008) | 6 lines Fix a bug in iax2 registration that allowed peers to register with case-insensitive names (user_cmp_cb and peer_cmp_cb are now both case-sensitive). (closes issue #13091) ........ ................ 2008-07-16 21:54 +0000 [r131455-131486] Brett Bryant * /, funcs/func_sysinfo.c: Merged revisions 131484 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r131484 | bbryant | 2008-07-16 16:54:08 -0500 (Wed, 16 Jul 2008) | 4 lines Fixes sysinfo operator issue also fixed elsewhere in r131445. (issue #13057) ........ * main/asterisk.c, /: Merged revisions 131445 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r131445 | bbryant | 2008-07-16 16:24:18 -0500 (Wed, 16 Jul 2008) | 9 lines Fixes an issue with "core show sysinfo" that used the wrong operator to calculate the number of bytes from a sysinfo structure. unsigned long. (closes issue #13057) Reported by: eliel Patches: asterisk.c.patch uploaded by eliel (license 64) ........ 2008-07-16 20:48 +0000 [r131423] Russell Bryant * /, channels/chan_iax2.c: Merged revisions 131422 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r131422 | russell | 2008-07-16 15:48:27 -0500 (Wed, 16 Jul 2008) | 15 lines Merged revisions 131421 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r131421 | russell | 2008-07-16 15:47:53 -0500 (Wed, 16 Jul 2008) | 7 lines Always ensure that the channel's tech_pvt reference is NULL after calling the destroy callback. (closes issue #13060) Reported by: jpgrayson Patches: chan_iax2_tech_pvt_crash.patch uploaded by jpgrayson (license 492) ........ ................ 2008-07-16 20:24 +0000 [r131301-131378] Mark Michelson * /, apps/app_queue.c: Merged revisions 131375 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r131375 | mmichelson | 2008-07-16 15:24:12 -0500 (Wed, 16 Jul 2008) | 22 lines Merged revisions 131369 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r131369 | mmichelson | 2008-07-16 15:23:02 -0500 (Wed, 16 Jul 2008) | 14 lines Move the init_queue call back to where it used to be (changed Sept 12 last year). It was moved then to prevent a memory leak. Since then, the same memory leak recurred and was fixed in a better way. Now it has been found that the placement of this init_queue call can cause problems if a realtime queue has values changed to an empty string. The problem is that the default value for that queue parameter would not be set. (closes issue #13084) Reported by: elbriga ........ ................ * res/res_config_sqlite.c, /: Merged revisions 131361 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r131361 | mmichelson | 2008-07-16 14:57:02 -0500 (Wed, 16 Jul 2008) | 9 lines Don't try to dereference the dbfile pointer if we know that it's NULL. (closes issue #13092) Reported by: gknispel_proformatique Patches: trunk_sqlite_check_vars_null.patch uploaded by gknispel (license 261) ........ * /, apps/app_queue.c: Merged revisions 131358 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r131358 | mmichelson | 2008-07-16 14:37:42 -0500 (Wed, 16 Jul 2008) | 14 lines Merged revisions 131357 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r131357 | mmichelson | 2008-07-16 14:37:08 -0500 (Wed, 16 Jul 2008) | 6 lines Apparently, "thread safety" is important, whatever that means. :P (Thanks Russell!) ........ ................ * /, apps/app_queue.c: Merged revisions 131300 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r131300 | mmichelson | 2008-07-16 13:59:27 -0500 (Wed, 16 Jul 2008) | 21 lines Merged revisions 131299 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r131299 | mmichelson | 2008-07-16 13:57:34 -0500 (Wed, 16 Jul 2008) | 13 lines Make absolutely certain that the transfer datastore is removed from the calling channel once the caller is finished in the queue. This could have weird con- sequences when dialing local queue members when multiple transfers occur on a single call. Also fixed a memory leak that would occur when an attended transfer occurred from a queue member. (closes issue #13047) Reported by: festr ........ ................ 2008-07-16 18:20 +0000 [r131248] Steve Murphy * res/ael/pval.c, /: Merged revisions 131243 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r131243 | murf | 2008-07-16 11:59:33 -0600 (Wed, 16 Jul 2008) | 27 lines Merged revisions 131242 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r131242 | murf | 2008-07-16 11:53:43 -0600 (Wed, 16 Jul 2008) | 19 lines (closes issue #13090) Reported by: murf The problem was that, esoteric as it is, because the hangerupper context immediately preceded the std-priv-extent macro, that the checking code accidentally would fall from traversing hangerupper into the std-priv-exten macro, where it would hit the hangerupper in the 'includes', and proceed into an infinite recursion. A small fix to traverse into the statements of the context instead of the context solves this issue. I also added some commented out printfs for debug, which were pretty handy in the face of a dorky gdb. This was a problem around since the package was first written; but evidently pretty rare in turning up in the field. ........ ................ 2008-07-16 15:04 +0000 [r131206] Luigi Rizzo * channels/chan_agent.c: add missing terminator argument for ast_event_subscribe(). Without it the function will randomly walk on the stack possibly causing a panic 2008-07-16 00:54 +0000 [r131168] Tilghman Lesher * /, main/logger.c: Merged revisions 131166 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r131166 | tilghman | 2008-07-15 19:52:48 -0500 (Tue, 15 Jul 2008) | 3 lines Fix rotate strategy (Closes issue #13086) ........ 2008-07-15 23:41 +0000 [r131131] Steve Murphy * main/pbx.c, /: Merged revisions 131129 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r131129 | murf | 2008-07-15 17:36:19 -0600 (Tue, 15 Jul 2008) | 21 lines (closes issue #12960) Reported by: mnicholson Spent most of the day on this bug, and the solution was so simple. Just had to find and understand the problem. The problem was, that the routine to copy the existing switches, includes, and ignorepats from the old context to the new one, wasn't getting called when the context is already existent. (In other words, if AEL is adding a new context to the mix, they get copied, but if pbx_config already defined a context, then the copy wasn't happening. This made no sense, so I moved the call to copy the includes & etc, no matter the case. ........ 2008-07-15 18:47 +0000 [r131073] Russell Bryant * /, res/res_agi.c: Merged revisions 131072 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r131072 | russell | 2008-07-15 13:46:40 -0500 (Tue, 15 Jul 2008) | 5 lines Fix a couple of places in res_agi where the agi_commands lock would not be released, causing a deadlock. (Reported by mvanbaak in #asterisk-dev, discovered by bbryant's change to the lock tracking code to yell at you if a thread exits with a lock still held) ........ 2008-07-15 18:29 +0000 [r131060] Tilghman Lesher * main/pbx.c, main/manager.c, /, channels/chan_sip.c: Merged revisions 131044 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r131044 | tilghman | 2008-07-15 13:25:34 -0500 (Tue, 15 Jul 2008) | 16 lines Merged revisions 130959 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r130959 | tilghman | 2008-07-15 12:19:13 -0500 (Tue, 15 Jul 2008) | 8 lines astman_send_error does not need a newline appended -- the API takes care of that for us. (closes issue #13068) Reported by: gknispel_proformatique Patches: asterisk_1_4_astman_send.patch uploaded by gknispel (license 261) asterisk_trunk_astman_send.patch uploaded by gknispel (license 261) ........ ................ 2008-07-15 18:00 +0000 [r131014] Michiel van Baak * main/cdr.c, /: Merged revisions 131013 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r131013 | mvanbaak | 2008-07-15 19:49:48 +0200 (Tue, 15 Jul 2008) | 15 lines Merged revisions 131012 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r131012 | mvanbaak | 2008-07-15 19:47:15 +0200 (Tue, 15 Jul 2008) | 7 lines remove 4 lines of redundant code. (closes issue #13080) Reported by: gknispel_proformatique Patches: trunk_ast_cdr_setapp.patch uploaded by gknispel (license 261) ........ ................ 2008-07-15 13:14 +0000 [r130946] Steve Murphy * utils/conf2ael.c, utils/Makefile, res/ael/pval.c, channels/chan_skinny.c, res/ael/ael.tab.c, main/features.c, pbx/pbx_ael.c, res/ael/ael_lex.c, res/ael/ael.tab.h, utils/ael_main.c, include/asterisk/pbx.h, utils/extconf.c, res/ael/ael.flex, pbx/pbx_config.c, apps/app_stack.c, apps/app_dial.c, main/pbx.c, include/asterisk/pval.h, /, channels/chan_sip.c, apps/app_meetme.c, res/ael/ael.y, channels/chan_iax2.c, apps/app_queue.c: Merged revisions 130145 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk Merging this rev from trunk to 1.6.0 was not simple. Why? Because we've enhanced trunk to do a [fast] merge-and-delete operation which also solved problems with contexts having entries from different registrars. Fast as in the amount of time the contexts are locked down. That *is* fast, but traversing the entire dialplan looking for priorities to delete takes more time overall. This particular fix involved pulling in those enhancements from trunk, along with all the various fixes and refinements made along the way. Merging all this from trunk into 1.6 involved: a. mergetrunk6 in the stuff from 130145; b. revert all but the prop changes c. catalog all revisions to pbx.c since 1.6.0 was forked (at rev 105596). d. catalog all revisions to pbx.c in trunk since 1.6.0 was forked, making special note of all revs that were not merged into 1.6.0. e. study each rev in trunk not applied to 1.6.0, and determine if it was involved in the merge_and_delete enhancements in trunk. 25 commits were done in 1.6.0, all but one (106306) was a merge from trunk. Trunk had 22 additional changes, of which 7 were involved in the merge_and_delete enhancements: 106757 108894 109169 116461 123358 130145 130297 f. Go to trunk and collect patches, one by one, of the changes made by each rev across the entire source tree, using svn diff -c > pfile g. Apply each patch in order to 1.6.0, and resolve all failures and compilation problems before proceding to the next patch. h. test the stuff. i. profit! ........ r130145 | murf | 2008-07-11 12:24:31 -0600 (Fri, 11 Jul 2008) | 40 lines (closes issue #13041) Reported by: eliel Tested by: murf (closes issue #12960) Reported by: mnicholson In this 'omnibus' fix, I **think** I solved both the problem in 13041, where unloading pbx_ael.so caused crashes, or incomplete removal of previous registrar'ed entries. And I added code to completely remove all includes, switches, and ignorepats that had a matching registrar entry, which should appease 12960. I also added a lot of seemingly useless brackets around single statement if's, which helped debug so much that I'm leaving them there. I added a routine to check the correlation between the extension tree lists and the hashtab tables. It can be amazingly helpful when you have lots of dialplan stuff, and need to narrow down where a problem is occurring. It's ifdef'd out by default. I cleaned up the code around the new CIDmatch code. It was leaving hanging extens with bad ptrs, getting confused over which objects to remove, etc. I tightened up the code and changed the call to remove_exten in the merge_and_delete code. I added more conditions to check for empty context worthy of deletion. It's not empty if there are any includes, switches, or ignorepats present. If I've missed anything, please re-open this bug, and be prepared to supply example dialplan code. ........ 2008-07-15 00:00 +0000 [r130891] Tilghman Lesher * /, channels/chan_iax2.c: Merged revisions 130890 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r130890 | tilghman | 2008-07-14 18:59:54 -0500 (Mon, 14 Jul 2008) | 16 lines Merged revisions 130889 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r130889 | tilghman | 2008-07-14 18:59:13 -0500 (Mon, 14 Jul 2008) | 8 lines Override the callerid in all cases when the callerid is set in the user, not just when a remote callerid is set. Also, if not set in the user, allow the remote CallerID to pass through. (closes issue #12875) Reported by: dimas Patches: 20080714__bug12875.diff.txt uploaded by Corydon76 (license 14) ........ ................ 2008-07-14 22:24 +0000 [r130795-130855] Mark Michelson * main/asterisk.c, /: Merged revisions 130854 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r130854 | mmichelson | 2008-07-14 17:22:57 -0500 (Mon, 14 Jul 2008) | 9 lines Fix a memory leak in the case that /dev/null cannot be opened when running startup commands from cli.conf (closes issue #13066) Reported by: eliel Patches: asterisk.c.patch uploaded by eliel (license 64) ........ * apps/app_dial.c, /: Merged revisions 130794 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r130794 | mmichelson | 2008-07-14 12:54:11 -0500 (Mon, 14 Jul 2008) | 16 lines Merged revisions 130792 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r130792 | mmichelson | 2008-07-14 12:50:21 -0500 (Mon, 14 Jul 2008) | 8 lines Add a check to the CAN_EARLY_BRIDGE macro in app_dial to be sure there are no audiohooks present on the channels involved. This fixed a one-way audio situation I had in my test setup. I couldn't find any open issues that suggested one-way audio with regards to mixmonitor (or other audiohook) usage, though. ........ ................ 2008-07-14 17:22 +0000 [r130752] Michiel van Baak * main/dnsmgr.c, /: Merged revisions 130744 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r130744 | mvanbaak | 2008-07-14 19:21:18 +0200 (Mon, 14 Jul 2008) | 18 lines Merged revisions 130735 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r130735 | mvanbaak | 2008-07-14 19:10:21 +0200 (Mon, 14 Jul 2008) | 10 lines notify the user that dnsmgr refresh wont work when dnsmgr is not enabled. Previously this command would automagically appear and disappear. This was confusing. (closes issue #12796) Reported by: chappell Patches: dnsmgr_refresh_3.diff uploaded by chappell (license 8) Tested by: russell, chappell, mvanbaak ........ ................ 2008-07-14 10:40 +0000 [r130636-130637] Russell Bryant * /, include/asterisk/astobj.h: Merged revisions 129987 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r129987 | russell | 2008-07-11 09:22:44 -0500 (Fri, 11 Jul 2008) | 10 lines Merged revisions 129970 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r129970 | russell | 2008-07-11 09:18:43 -0500 (Fri, 11 Jul 2008) | 2 lines add a simple ASTOBJ_TRYWRLOCK macro ... ........ ................ * /, main/audiohook.c: Merged revisions 130635 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r130635 | russell | 2008-07-14 05:39:23 -0500 (Mon, 14 Jul 2008) | 10 lines Merged revisions 130634 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r130634 | russell | 2008-07-14 05:38:14 -0500 (Mon, 14 Jul 2008) | 2 lines Bump up the debug level for a message. ........ ................ 2008-07-13 23:20 +0000 [r130575-130582] Michiel van Baak * /, doc/tex/Makefile, build_tools/prep_tarball, res/Makefile: Merged revisions 130578 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r130578 | mvanbaak | 2008-07-14 01:14:03 +0200 (Mon, 14 Jul 2008) | 15 lines Make all sed calls Posix sed compatible. To make sure nobody commits script-modified files we first make a backup of asterisk.tex, run the script, generate the pdf and / or html, and put the original asterisk.tex back. This will guard us for the stuff that happened before that someone committed a locally modified asterisk.tex, with changes done by this script. (closes issue #13062) Reported by: mvanbaak Patches: sed_without-i-v3.diff uploaded by mvanbaak (license 7) Tested by: mvanbaak Feedback from Corydon. Thanks for taking the time to go through this. ........ * main/manager.c, /: Merged revisions 130574 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r130574 | mvanbaak | 2008-07-14 00:50:31 +0200 (Mon, 14 Jul 2008) | 16 lines Merged revisions 130573 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r130573 | mvanbaak | 2008-07-14 00:48:51 +0200 (Mon, 14 Jul 2008) | 8 lines fix memory leak when originate from manager cannot create a thread (closes issue #13069) Reported by: gknispel_proformatique Patches: asterisk_trunk_action_originate.patch uploaded by gknispel (license 261) Tested by: gknispel_proformatique, mvanbaak ........ ................ 2008-07-13 17:59 +0000 [r130516] Tilghman Lesher * /, channels/chan_iax2.c: Merged revisions 130515 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r130515 | tilghman | 2008-07-13 12:58:47 -0500 (Sun, 13 Jul 2008) | 12 lines Merged revisions 130514 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r130514 | tilghman | 2008-07-13 12:56:10 -0500 (Sun, 13 Jul 2008) | 4 lines Reverting 2 changesets, as it breaks incoming IAX2 calls (Related to issue #12963) Reported by: mvanbaak ........ ................ 2008-07-13 15:00 +0000 [r130480] Michiel van Baak * doc/tex/asterisk.tex, /: Merged revisions 130479 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r130479 | mvanbaak | 2008-07-13 16:58:40 +0200 (Sun, 13 Jul 2008) | 3 lines restore ASTERISKVERSION marker to asterisk.tex. This got lost in commit 97634 ........ 2008-07-13 02:35 +0000 [r130445] Tilghman Lesher * /, channels/chan_agent.c: Merged revisions 130444 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r130444 | tilghman | 2008-07-12 21:34:32 -0500 (Sat, 12 Jul 2008) | 2 lines Unlock list before returning ........ 2008-07-11 21:39 +0000 [r130294] Matthew Fredrickson * channels/chan_dahdi.c, /: Merged revisions 130293 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r130293 | mattf | 2008-07-11 16:36:26 -0500 (Fri, 11 Jul 2008) | 1 line Support new TRANSPORT definitions in libss7 ........ 2008-07-11 20:04 +0000 [r130238] Mark Michelson * /, main/audiohook.c: Merged revisions 130237 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r130237 | mmichelson | 2008-07-11 15:03:55 -0500 (Fri, 11 Jul 2008) | 11 lines Merged revisions 130236 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r130236 | mmichelson | 2008-07-11 15:03:23 -0500 (Fri, 11 Jul 2008) | 3 lines Remove redundant logic ........ ................ 2008-07-11 19:57 +0000 [r130231-130235] Tilghman Lesher * channels/chan_dahdi.c, /, channels/chan_agent.c, utils/astman.c: Merged revisions 130230 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r130230 | tilghman | 2008-07-11 14:40:55 -0500 (Fri, 11 Jul 2008) | 2 lines Fix trunk breakage ........ 2008-07-11 19:14 +0000 [r130175] Mark Michelson * /, main/audiohook.c: Merged revisions 130174 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r130174 | mmichelson | 2008-07-11 14:14:15 -0500 (Fri, 11 Jul 2008) | 15 lines Merged revisions 130173 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r130173 | mmichelson | 2008-07-11 14:13:29 -0500 (Fri, 11 Jul 2008) | 7 lines Fix a typo in audiohook_read_frame_both. While this change has not been proven to fix any specific issue, it is incorrect and could cause unforeseen problems. ........ ................ 2008-07-11 18:53 +0000 [r130171] Tilghman Lesher * /, channels/chan_iax2.c: Merged revisions 130170 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r130170 | tilghman | 2008-07-11 13:52:42 -0500 (Fri, 11 Jul 2008) | 15 lines Merged revisions 130169 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r130169 | tilghman | 2008-07-11 13:51:56 -0500 (Fri, 11 Jul 2008) | 7 lines Ensure that a destination callno of 0 will not match for frames that do not start a dialog (new, lagrq, and ping). (closes issue #12963) Reported by: russellb Patches: chan_iax2_dup_new_fix4.patch uploaded by jpgrayson (license 492) ........ ................ 2008-07-11 18:33 +0000 [r130168] Sean Bright * /, channels/chan_sip.c: Merged revisions 130167 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r130167 | seanbright | 2008-07-11 14:32:26 -0400 (Fri, 11 Jul 2008) | 1 line Missed one. Formatting only. ........ 2008-07-11 18:14 +0000 [r130130] Brett Bryant * main/cli.c, channels/chan_jingle.c, channels/chan_dahdi.c, channels/chan_skinny.c, main/abstract_jb.c, apps/app_minivm.c, codecs/codec_resample.c, codecs/codec_dahdi.c, apps/app_chanspy.c, main/asterisk.c, apps/app_milliwatt.c, main/dsp.c, codecs/codec_g722.c, /, channels/chan_sip.c, main/threadstorage.c, utils/astman.c, main/utils.c, channels/chan_gtalk.c, pbx/dundi-parser.c: Merged revisions 130129 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r130129 | bbryant | 2008-07-11 13:09:35 -0500 (Fri, 11 Jul 2008) | 8 lines Janitor patch to change uses of sizeof to ARRAY_LEN (closes issue #13054) Reported by: pabelanger Patches: ARRAY_LEN.patch2 uploaded by pabelanger (license 224) Tested by: seanbright ........ 2008-07-11 17:30 +0000 [r130127] Tilghman Lesher * /, channels/chan_agent.c: Merged revisions 130126 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r130126 | tilghman | 2008-07-11 12:29:24 -0500 (Fri, 11 Jul 2008) | 17 lines Merged revisions 130102 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r130102 | tilghman | 2008-07-11 11:50:42 -0500 (Fri, 11 Jul 2008) | 9 lines Pass the devicestate from an underlying channel up through the Agent channel. This should make the Agent always report the correct device state, even when the underlying channel is used for other purposes. (closes issue #12773) Reported by: davidw Patches: 20080710__bug12773.diff.txt uploaded by Corydon76 (license 14) Tested by: davidw ........ ................ 2008-07-11 16:18 +0000 [r129936-130045] Kevin P. Fleming * doc/ss7.txt, /, contrib/utils/zones2indications.c, CHANGES: Merged revisions 130044 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r130044 | kpfleming | 2008-07-11 11:18:01 -0500 (Fri, 11 Jul 2008) | 2 lines clean up a bunch more Zaptel-related references ........ * channels/chan_dahdi.c, configs/chan_dahdi.conf.sample, /, configure, include/asterisk/autoconfig.h.in, configure.ac: Merged revisions 130040 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r130040 | kpfleming | 2008-07-11 10:57:17 -0500 (Fri, 11 Jul 2008) | 12 lines Merged revisions 130039 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r130039 | kpfleming | 2008-07-11 10:41:56 -0500 (Fri, 11 Jul 2008) | 4 lines add support for a configuration parameter for 'inband audio during RELEASE', which is currently mandatory in libpri-1.4.4 but will become configurable in libpri-1.4.5 later today (related to issue #13042) ........ ................ * /, main/astmm.c: Merged revisions 129968 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r129968 | kpfleming | 2008-07-11 09:16:15 -0500 (Fri, 11 Jul 2008) | 18 lines Merged revisions 129966 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r129966 | kpfleming | 2008-07-11 09:03:52 -0500 (Fri, 11 Jul 2008) | 5 lines fix a flaw found while experimenting with structure alignment and padding; low-fence checking would not work properly on 64-bit platforms, because the compiler was putting 4 bytes of padding between the fence field and the allocation memory block added a very obvious runtime warning if this condition reoccurs, so the developer who broke it can be chastised into fixing it :-) ........ r129967 | kpfleming | 2008-07-11 09:03:52 -0500 (Fri, 11 Jul 2008) | 5 lines simplify calculation ........ ................ * /, sounds/Makefile: Merged revisions 129916 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r129916 | kpfleming | 2008-07-11 07:21:29 -0500 (Fri, 11 Jul 2008) | 10 lines Merged revisions 129907 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r129907 | kpfleming | 2008-07-11 07:15:42 -0500 (Fri, 11 Jul 2008) | 2 lines don't attempt to set user/group ownership of extracted sound files (reported on asterisk-users) ........ ................ 2008-07-11 01:01 +0000 [r129865] Sean Bright * res/res_config_pgsql.c, /, res/res_config_ldap.c: Merged revisions 129864 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r129864 | seanbright | 2008-07-10 20:55:06 -0400 (Thu, 10 Jul 2008) | 1 line Fix some usages of snprintf, and clarify a couple variable names. ........ 2008-07-10 22:07 +0000 [r129764-129805] Tilghman Lesher * /, channels/chan_iax2.c: Merged revisions 129804 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r129804 | tilghman | 2008-07-10 17:06:07 -0500 (Thu, 10 Jul 2008) | 16 lines Merged revisions 129803 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r129803 | tilghman | 2008-07-10 16:57:05 -0500 (Thu, 10 Jul 2008) | 8 lines Correctly deal with duplicate NEW frames (due to retransmission). Also, fixup the destination call number matching to be more strict and reliable. (closes issue #12963) Reported by: jpgrayson Patches: chan_iax2_dup_new_fix3.patch uploaded by jpgrayson (license 492) Tested by: jpgrayson, Corydon76 ........ ................ * res/res_config_odbc.c, /: Merged revisions 129758 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r129758 | tilghman | 2008-07-10 16:23:23 -0500 (Thu, 10 Jul 2008) | 10 lines Merged revisions 129741 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r129741 | tilghman | 2008-07-10 16:19:48 -0500 (Thu, 10 Jul 2008) | 2 lines Oops ........ ................ 2008-07-10 21:05 +0000 [r129739] Terry Wilson * Makefile, /: Merged revisions 129738 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r129738 | twilson | 2008-07-10 15:56:20 -0500 (Thu, 10 Jul 2008) | 2 lines Move phoneprov config files to be installed with 'make samples' so changes aren't inadvertently lost on a 'make install' ........ 2008-07-10 19:14 +0000 [r129685] Brett Bryant * /, apps/app_queue.c: Merged revisions 129684 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r129684 | bbryant | 2008-07-10 14:13:12 -0500 (Thu, 10 Jul 2008) | 8 lines Fixes a bug where the interface for a queue member gets reloaded as the state_interface, if a state_interface was set, on reload because the state_interface isn't stored in the ast_db. (closes issue #13043) Reported by: jvandal Patches: app_queue.patch uploaded by jvandal (license 413) ........ 2008-07-10 18:20 +0000 [r129640-129647] Sean Bright * /, channels/chan_sip.c: Merged revisions 129642 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r129642 | seanbright | 2008-07-10 14:19:17 -0400 (Thu, 10 Jul 2008) | 1 line A couple more minor text changes ........ * /, channels/chan_sip.c: Merged revisions 129638 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r129638 | seanbright | 2008-07-10 14:16:21 -0400 (Thu, 10 Jul 2008) | 1 line Remove extraneous \n. Pointed out by eliel on #asterisk-dev. ........ 2008-07-10 16:13 +0000 [r129570] Russell Bryant * sample.call, /: Merged revisions 129569 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r129569 | russell | 2008-07-10 11:12:51 -0500 (Thu, 10 Jul 2008) | 11 lines Merged revisions 129567 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r129567 | russell | 2008-07-10 11:03:59 -0500 (Thu, 10 Jul 2008) | 3 lines Note that pbx_spool.so is the module used for call files (inspired by a question in #asterisk) ........ ................ 2008-07-10 14:09 +0000 [r129504-129507] Sean Bright * /, main/editline: Merged revisions 129503 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r129503 | seanbright | 2008-07-10 09:54:29 -0400 (Thu, 10 Jul 2008) | 2 lines Update svn:ignore ........ 2008-07-09 19:41 +0000 [r129438] Mark Michelson * main/rtp.c, /: Merged revisions 129437 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r129437 | mmichelson | 2008-07-09 14:40:30 -0500 (Wed, 09 Jul 2008) | 21 lines Merged revisions 129436 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r129436 | mmichelson | 2008-07-09 14:32:20 -0500 (Wed, 09 Jul 2008) | 13 lines Fix a problem where inbound rfc2833 audio would be sent to the core instead of being P2P bridged. When the core regenerated the rfc2833 packet for the outbound leg, the SSRC would be different than the RTP audio on the call leg causing DTMF detection issues on the far end. (closes issue #12955) Reported by: tonyredstone Patches: dynamic_rtp.patch uploaded by tsearle (license 373) Tested by: tonyredstone ........ ................ 2008-07-09 16:01 +0000 [r129400] Matthew Fredrickson * main/pbx.c, /: Merged revisions 129399 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r129399 | mattf | 2008-07-09 10:57:06 -0500 (Wed, 09 Jul 2008) | 1 line Add Proceeding() application (#13025) ........ 2008-07-09 13:46 +0000 [r129345] Sean Bright * main/editline/makelist (removed), main/editline/makelist.in (added), /, main/editline/configure, main/editline/Makefile.in, main/editline/configure.in: Merged revisions 129344 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r129344 | seanbright | 2008-07-09 09:44:43 -0400 (Wed, 09 Jul 2008) | 12 lines Merged revisions 129343 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r129343 | seanbright | 2008-07-09 09:41:21 -0400 (Wed, 09 Jul 2008) | 4 lines Look for the system installed awk instead of assuming it's at /usr/bin/awk. Pointed out by jmls via #asterisk-dev. ........ ................ 2008-07-08 22:56 +0000 [r129160-129271] Mark Michelson * apps/app_voicemail.c, /: Merged revisions 129270 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r129270 | mmichelson | 2008-07-08 17:56:12 -0500 (Tue, 08 Jul 2008) | 3 lines Fix compilation error when IMAP storage is enabled ........ 2008-07-08 21:04 +0000 [r129157] Brett Bryant * main/dns.c, main/srv.c, /: Merged revisions 129156 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r129156 | bbryant | 2008-07-08 16:00:01 -0500 (Tue, 08 Jul 2008) | 6 lines Fix a bug in SRV lookups where dnsmgr would discard everything but the first SRV result from DNS before processing weights and priorities and dns_parse_answer wouldn't report that there were no records in DNS unless a failure occured. Also fixed a bug where dnsmgr_refresh would report that a entry was being changed when ast_gethostbyname had failed. ........ 2008-07-08 20:31 +0000 [r129049-129153] Tilghman Lesher * apps/app_dial.c, /, channels/chan_sip.c, include/asterisk/causes.h: Merged revisions 129152 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r129152 | tilghman | 2008-07-08 15:30:29 -0500 (Tue, 08 Jul 2008) | 16 lines Merged revisions 129149 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r129149 | tilghman | 2008-07-08 15:27:47 -0500 (Tue, 08 Jul 2008) | 8 lines Cause SIP to return a 480 instead of a 404 when a sip peer exists, but is not registered. (closes issue #12885) Reported by: ibc Patches: 20080701__bug12885__2.diff.txt uploaded by Corydon76 (license 14) Tested by: ibc ........ ................ * /, channels/chan_iax2.c: Merged revisions 129048 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r129048 | tilghman | 2008-07-08 11:49:01 -0500 (Tue, 08 Jul 2008) | 15 lines Merged revisions 129047 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r129047 | tilghman | 2008-07-08 11:45:23 -0500 (Tue, 08 Jul 2008) | 7 lines Timestamp decoding for video mini-frames is bogus, because the timestamp only includes 15 bits, unlike voice frames, which contain a 16-bit timestamp. (closes issue #13013) Reported by: jpgrayson Patches: chan_iax2_unwrap_ts.patch uploaded by jpgrayson (license 492) ........ ................ 2008-07-08 16:41 +0000 [r129041-129046] Brett Bryant * main/rtp.c, main/channel.c, channels/chan_dahdi.c, main/manager.c, formats/format_pcm.c, main/logger.c, main/callerid.c, apps/app_parkandannounce.c, apps/app_adsiprog.c, main/pbx.c, main/frame.c, /, channels/chan_sip.c, apps/app_meetme.c, channels/h323/ast_h323.cxx, res/res_limit.c, main/acl.c, channels/iax2-provision.c, pbx/dundi-parser.c, channels/chan_iax2.c: Merged revisions 129045 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r129045 | bbryant | 2008-07-08 11:40:28 -0500 (Tue, 08 Jul 2008) | 7 lines Janitor project to convert sizeof to ARRAY_LEN macro. (closes issue #13002) Reported by: caio1982 Patches: janitor_arraylen5.diff uploaded by caio1982 (license 22) ........ * /, channels/chan_sip.c: Merged revisions 127621 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r127621 | bbryant | 2008-07-02 17:16:29 -0500 (Wed, 02 Jul 2008) | 1 line Update transport= in sip so that the option is not broken from a recent commit. ........ * /, channels/chan_sip.c: Merged revisions 127434 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r127434 | bbryant | 2008-07-02 12:27:36 -0500 (Wed, 02 Jul 2008) | 1 line Fix to sip_parse_host so that it passes the correct information to sip_registry. ........ 2008-07-08 14:18 +0000 [r129007] Russell Bryant * /, apps/app_fax.c: Merged revisions 129006 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r129006 | russell | 2008-07-08 09:17:37 -0500 (Tue, 08 Jul 2008) | 9 lines Update app_fax for better compatibility with spandsp 0.0.5. Add a call to t38_terminal_release, and make sure that the phase E handler gets called with proper status. (closes issue #13020) Reported by: dimas Patches: v1-appfax.patch uploaded by dimas (license 88) ........ 2008-07-08 10:06 +0000 [r128913-128952] Olle Johansson * /, channels/chan_sip.c: Merged revisions 128951 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r128951 | oej | 2008-07-08 12:02:12 +0200 (Tis, 08 Jul 2008) | 19 lines Merged revisions 128950 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r128950 | oej | 2008-07-08 11:52:21 +0200 (Tis, 08 Jul 2008) | 11 lines Don't hangup the call if we can't resolve the Contact if there's a proxy route set for the call. ---- This comment was added a while ago and today it hit me badly. /* OEJ: Possible issue that may need a check: If we have a proxy route between us and the device, should we care about resolving the contact or should we just send it? */ ........ ................ * /, channels/chan_sip.c: Merged revisions 128927 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r128927 | oej | 2008-07-08 11:26:37 +0200 (Tis, 08 Jul 2008) | 15 lines Merged revisions 128912 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r128912 | oej | 2008-07-08 11:06:08 +0200 (Tis, 08 Jul 2008) | 7 lines Fix issues where repeated messages where ignored, but retransmitted reliably instead of unreliably. Reported by: johan Patches: 12746.txt uploaded by oej (license 306) Tested by: johan (issue #12746) ........ ................ 2008-07-08 00:03 +0000 [r128855-128858] Tilghman Lesher * /: Merged revisions 128857 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r128857 | tilghman | 2008-07-07 19:02:11 -0500 (Mon, 07 Jul 2008) | 15 lines Merged revisions 128856 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r128856 | tilghman | 2008-07-07 19:01:30 -0500 (Mon, 07 Jul 2008) | 7 lines Check for non-NULL before stripping characters. (closes issue #12954) Reported by: bfsworks Patches: 20080701__bug12954.diff.txt uploaded by Corydon76 (license 14) Tested by: deti ........ ................ * apps/app_voicemail.c, /: Merged revisions 128830 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r128830 | tilghman | 2008-07-07 18:25:39 -0500 (Mon, 07 Jul 2008) | 10 lines Merged revisions 128812 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r128812 | tilghman | 2008-07-07 18:21:52 -0500 (Mon, 07 Jul 2008) | 2 lines Stop using deprecated method, as requested by Kevin. ........ ................ 2008-07-07 22:44 +0000 [r128797] Russell Bryant * /, channels/chan_iax2.c: Merged revisions 128796 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r128796 | russell | 2008-07-07 17:42:30 -0500 (Mon, 07 Jul 2008) | 16 lines Merged revisions 128795 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r128795 | russell | 2008-07-07 17:41:48 -0500 (Mon, 07 Jul 2008) | 8 lines Fix handling of when a pvt disappears. Properly return the pvt locked and don't hold the pvt lock while destroying the ast_channel. (closes issue #13014) Reported by: jpgrayson Patches: chan_iax2_ast_iax2_new2.patch uploaded by jpgrayson (license 492) ........ ................ 2008-07-07 20:51 +0000 [r128739] Sean Bright * /, channels/chan_iax2.c: Merged revisions 128738 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r128738 | seanbright | 2008-07-07 16:50:29 -0400 (Mon, 07 Jul 2008) | 17 lines Merged revisions 128737 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r128737 | seanbright | 2008-07-07 16:47:56 -0400 (Mon, 07 Jul 2008) | 9 lines Remove spurious trailing whitespace from log messages and fix a spelling error in a log message. (closes issue #13017) Reported by: jpgrayson Patches: chan_iax2_space_after_newline.patch uploaded by jpgrayson (license 492) chan_iax2_spelling.patch uploaded by jpgrayson (license 492) ........ ................ 2008-07-07 20:31 +0000 [r128601-128735] Mark Michelson * apps/app_voicemail.c, /: Merged revisions 128733 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r128733 | mmichelson | 2008-07-07 15:30:46 -0500 (Mon, 07 Jul 2008) | 3 lines Crap ........ * apps/app_voicemail.c, /: Merged revisions 128731 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r128731 | mmichelson | 2008-07-07 15:28:33 -0500 (Mon, 07 Jul 2008) | 7 lines If imapfolder=foo were set in voicemail.conf, then when calling VoiceMailMain, app_voicemail would attempt to play a file called vm-foo instead of playing vm-INBOX to play the "new" sound file. This commit fixes that issue. This may fix one of the problems reported in issue #12987 ........ * /, channels/chan_iax2.c: Merged revisions 128640 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r128640 | mmichelson | 2008-07-07 12:09:11 -0500 (Mon, 07 Jul 2008) | 18 lines Merged revisions 128639 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r128639 | mmichelson | 2008-07-07 12:02:28 -0500 (Mon, 07 Jul 2008) | 10 lines By using the iaxdynamicthreadcount to identify a thread, it was possible for thread identifiers to be duplicated. By using a globally-unique monotonically- increasing integer, this is now avoided. (closes issue #13009) Reported by: jpgrayson Patches: chan_iax2_dyn_threadnum.patch uploaded by jpgrayson (license 492) ........ ................ * configs/extensions.conf.sample, /, doc/tex/extensions.tex: Merged revisions 128599 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r128599 | mmichelson | 2008-07-07 09:35:27 -0500 (Mon, 07 Jul 2008) | 6 lines Update a few instances of "extensions reload" to "dialplan reload" in the documentation. Patch provided by caio1982 (license 22) ........ 2008-07-06 20:22 +0000 [r128288-128543] Olle Johansson * /, channels/chan_sip.c, configs/sip.conf.sample: Merged revisions 128524 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r128524 | oej | 2008-07-06 22:11:37 +0200 (Sön, 06 Jul 2008) | 5 lines - Fixing issues with "sip show settings" - Adding IP address for TCP and/or TLS too if auto-domain is enabled and binding to a different IP address - Fixing documentation in sip.conf.sample ........ * /, channels/chan_sip.c: Merged revisions 128491 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r128491 | oej | 2008-07-06 21:14:06 +0200 (Sön, 06 Jul 2008) | 3 lines - Remove unused variable "expiry" - Set global_outboundproxy.force at reload. ........ * doc/realtimetext.txt (added), /: The following patch with references to t140red removed, since it only exists in trunk. Merged revisions 128417 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r128417 | oej | 2008-07-06 12:13:45 +0200 (Sön, 06 Jul 2008) | 3 lines Adding documentation on the T.140 support in Asterisk. This is a function that we're the reference implementation on now. :-) ........ * /: Merged revisions 128343 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r128343 | oej | 2008-07-06 10:10:27 +0200 (Sön, 06 Jul 2008) | 2 lines Removing the CLI dumpdb command (see asterisk-dev discussion and decision) ........ * /, channels/chan_sip.c: Merged revisions 128290 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r128290 | oej | 2008-07-05 23:55:57 +0200 (Lör, 05 Jul 2008) | 5 lines Adding doxygen comments to missing parts, moving some #define ...trying to get my head around the thoughts behind the TCP/TLS stuff and figure out what needs to be done to make it useful... ........ * /, channels/chan_sip.c: Merged revisions 128287 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r128287 | oej | 2008-07-05 23:37:57 +0200 (Lör, 05 Jul 2008) | 3 lines Adding TCP and TLS to "sip show settings". TLS needs to have one configuration per configured domain at some point. ........ * /: Blocking changes in trunk. 2008-07-05 21:02 +0000 [r128238-128243] Olle Johansson * /: Keep the "sip-user" structure in 1.6.0, while testing new funky stuff in trunk. * /: Blocking the AGi changes from 1.6.0. Let's test them for a while in trunk before a release. * /, channels/chan_sip.c, configs/sip.conf.sample: Merged revisions 128237 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r128237 | oej | 2008-07-05 22:39:54 +0200 (Lör, 05 Jul 2008) | 2 lines Make TCP disabled by default (it's considered experimental) ........ * /, configs/sip.conf.sample: Merged revisions 128236 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r128236 | oej | 2008-07-05 22:37:53 +0200 (Lör, 05 Jul 2008) | 2 lines Reformatting the config sample ........ 2008-07-05 15:19 +0000 [r128161] Tilghman Lesher * contrib/scripts/asterisk.ldap-schema, contrib/scripts/asterisk.ldif, /: Merged revisions 128160 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r128160 | tilghman | 2008-07-05 10:17:51 -0500 (Sat, 05 Jul 2008) | 7 lines LDAP schema updates (closes issue #12860) Reported by: flyn Patches: asterisk.ldif uploaded by suretec (license 70) asterisk.schema uploaded by suretec (license 70) ........ 2008-07-05 03:40 +0000 [r128124-128127] Matthew Fredrickson * channels/chan_dahdi.c, /: Merged revisions 128125 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r128125 | mattf | 2008-07-04 22:39:07 -0500 (Fri, 04 Jul 2008) | 1 line It would help if we actually parsed the ss7_explicitacm option in the config file... ........ * channels/chan_dahdi.c, configs/chan_dahdi.conf.sample, /: Merged revisions 128122 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r128122 | mattf | 2008-07-04 22:26:42 -0500 (Fri, 04 Jul 2008) | 1 line Add option to wait to be able to explicitly send ACM via the Proceeding() application in the dialplan. Also minor documentation update explaining how to setup multiple signalling links within a linkset ........ 2008-07-04 16:12 +0000 [r128028-128031] Tilghman Lesher * main/pbx.c, /, include/asterisk/pbx.h, pbx/pbx_config.c: Merged revisions 128027 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r128027 | tilghman | 2008-07-04 11:06:34 -0500 (Fri, 04 Jul 2008) | 16 lines Merged revisions 127973 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r127973 | tilghman | 2008-07-03 22:30:30 -0500 (Thu, 03 Jul 2008) | 8 lines Fix the 'dialplan remove extension' logic, so that it a) works with cidmatch, and b) completes contexts correctly when the extension is ambiguous. (closes issue #12980) Reported by: licedey Patches: 20080703__bug12980.diff.txt uploaded by Corydon76 (license 14) Tested by: Corydon76 ........ ................ 2008-07-03 22:23 +0000 [r127905] Kevin P. Fleming * Makefile, /, apps/Makefile, main/editline/np/vis.c: Merged revisions 127903 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r127903 | kpfleming | 2008-07-03 17:23:04 -0500 (Thu, 03 Jul 2008) | 20 lines Merged revisions 127892,127895 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r127892 | kpfleming | 2008-07-03 17:18:38 -0500 (Thu, 03 Jul 2008) | 6 lines a couple of small Solaris-related fixes (closes issue #11885) Reported by: snuffy, asgaroth ........ r127895 | kpfleming | 2008-07-03 17:20:16 -0500 (Thu, 03 Jul 2008) | 3 lines remove this, it has been moved to the main Makefile ........ ................ 2008-07-03 19:12 +0000 [r127830] Steve Murphy * main/cdr.c, main/channel.c, channels/chan_dahdi.c, main/pbx.c, /, channels/chan_sip.c, main/features.c, include/asterisk/cdr.h: Merged revisions 127793 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r127793 | murf | 2008-07-03 11:16:44 -0600 (Thu, 03 Jul 2008) | 38 lines Merged revisions 127663 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r127663 | murf | 2008-07-02 18:16:25 -0600 (Wed, 02 Jul 2008) | 30 lines The CDRfix4/5/6 omnibus cdr fixes. (closes issue #10927) Reported by: murf Tested by: murf, deeperror (closes issue #12907) Reported by: falves11 Tested by: murf, falves11 (closes issue #11849) Reported by: greyvoip As to 11849, I think these changes fix the core problems brought up in that bug, but perhaps not the more global problems created by the limitations of CDR's themselves not being oriented around transfers. Reopen if necc, but bug reports are not the best medium for enhancement discussions. We need to start a second-generation CDR standardization effort to cover transfers. (closes issue #11093) Reported by: rossbeer Tested by: greyvoip, murf ........ ................ 2008-07-03 16:50 +0000 [r127790-127792] Olle Johansson * /, channels/chan_sip.c: Merged revisions 127791 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r127791 | oej | 2008-07-03 18:48:23 +0200 (Tor, 03 Jul 2008) | 5 lines Make sure we stop session timers as soon as we start hanging up an active call. May fix issue 12919. ........ * /, channels/chan_sip.c: Merged revisions 127779 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r127779 | oej | 2008-07-03 18:25:59 +0200 (Tor, 03 Jul 2008) | 4 lines Revert some logic for session timers. We do send in-dialog requests that should not have session-timer require headers, like MESSAGE and REFER. So in the future, only add them on requests and responses that are related to INVITEs and re-INVITEs. ........ 2008-07-03 16:24 +0000 [r127778] Kevin P. Fleming * /, configure, include/asterisk/autoconfig.h.in, configure.ac, acinclude.m4: Merged revisions 127767 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r127767 | kpfleming | 2008-07-03 11:22:02 -0500 (Thu, 03 Jul 2008) | 2 lines some minor fixes found while working on issue #12911 (and block the rev from 1.4 since the equivalent is already here) ........ 2008-07-02 21:10 +0000 [r127567] Mark Michelson * /, doc/janitor-projects.txt: Merged revisions 127566 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r127566 | mmichelson | 2008-07-02 16:09:18 -0500 (Wed, 02 Jul 2008) | 4 lines Add a janitor project to use ARRAY_LEN instead of in-line sizeof() and division. ........ 2008-07-02 20:49 +0000 [r127559-127563] Mark Michelson * /, channels/chan_agent.c: Merged revisions 127562 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r127562 | mmichelson | 2008-07-02 15:49:08 -0500 (Wed, 02 Jul 2008) | 11 lines Merged revisions 127560 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r127560 | mmichelson | 2008-07-02 15:47:38 -0500 (Wed, 02 Jul 2008) | 3 lines Fix thread-safety of some of the pbx_builtin_getvar_helper calls ........ ................ 2008-07-02 19:48 +0000 [r127467-127503] Tilghman Lesher * /, main/acl.c: Merged revisions 127466 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r127466 | tilghman | 2008-07-02 13:31:11 -0500 (Wed, 02 Jul 2008) | 6 lines Solaris fix (closes issue #12949) Reported by: snuffy Patches: bug_12949.diff uploaded by snuffy (license 35) ........ 2008-07-02 14:30 +0000 [r127396-127399] Sean Bright * cdr/cdr_tds.c, /: Merged revisions 127398 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r127398 | seanbright | 2008-07-02 10:30:09 -0400 (Wed, 02 Jul 2008) | 1 line Fix a bug I noticed while doing the previous merge ........ * cdr/cdr_tds.c, /, doc/tex/freetds.tex, configure, include/asterisk/autoconfig.h.in, configure.ac, UPGRADE.txt: Merged revisions 126226,126513 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r126226 | seanbright | 2008-06-28 17:28:16 -0400 (Sat, 28 Jun 2008) | 8 lines Merge in changes from my cdr-tds-conversion branch. This changes the internal implementation from using the volatile libtds, to using the db-lib front end. The unintended side effect of this is that we support (at least) versions 0.62 through 0.82 of the FreeTDS distribution without any #ifdef ugliness. (closes issue #12844) Reported by: jcollie ........ r126513 | seanbright | 2008-06-30 07:57:42 -0400 (Mon, 30 Jun 2008) | 4 lines Cast a few more strings to char *, so that we can compile cleanly against FreeTDS 0.60. Update the docs to reflect that we can now compile and run against all modern releases of FreeTDS (0.60 through 0.82) ........ * /: Unblock some revisions so I can merge the cdr_tds changes from trunk 2008-07-02 12:09 +0000 [r127364] Russell Bryant * doc/CODING-GUIDELINES, /: Merged revisions 127363 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r127363 | russell | 2008-07-02 07:08:33 -0500 (Wed, 02 Jul 2008) | 13 lines Add a locking section to the coding guidelines document. This section covers some locking fundamentals, as well as some information on locking as it is used in Asterisk. It describes some of the ways that are used and could be used to achieve deadlock avoidance. It also demonstrates the unfortunate conclusion that with the use of recursive locks, none of the constructs in use today are failsafe from deadlocks. Finally, it makes some recommendations for new code being written. As proper locking strategies is a complex subject, this section still has room for expansion and improvement. This is a result of collaboration between Luigi Rizzo and myself on the asterisk-dev mailing list. ........ 2008-07-02 02:49 +0000 [r127298] Tilghman Lesher * /, channels/chan_sip.c: Merged revisions 127297 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r127297 | tilghman | 2008-07-01 21:48:43 -0500 (Tue, 01 Jul 2008) | 12 lines Change the global timer B to be dependent on the value of the T1 timer, as recommended in RFC 3261, instead of being hardcoded to 32 seconds. This is important for LANs, as it allows autocongestion to occur much more quickly, if desired by the local PBX administrator. It also corrects a bug: if the T1 timer was increased beyond 500ms, then timer B would have been set at a much lower value than recommended. (closes issue #12544) Reported by: kactus Patches: 20080616__bug12544.diff.txt uploaded by Corydon76 (license 14) Tested by: kactus ........ 2008-07-01 23:39 +0000 [r127246] Mark Michelson * apps/app_voicemail.c, /: Merged revisions 127245 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r127245 | mmichelson | 2008-07-01 18:38:12 -0500 (Tue, 01 Jul 2008) | 13 lines Merged revisions 127244 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r127244 | mmichelson | 2008-07-01 18:36:40 -0500 (Tue, 01 Jul 2008) | 5 lines Add error message to failed open(2) calls inside the copy() function of app_voicemail. This idea came as part of my work in helping to resolve issue #12764. ........ ................ 2008-07-01 21:19 +0000 [r127163] Brett Bryant * /, channels/chan_sip.c, configs/sip.conf.sample: Merged revisions 127154 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r127154 | bbryant | 2008-07-01 16:03:52 -0500 (Tue, 01 Jul 2008) | 2 lines Add a configuration option so the global outboundproxy can use tcptls without it being defined by each sip user. ........ 2008-07-01 21:16 +0000 [r127156-127158] Mark Michelson * main/channel.c, /: Merged revisions 127157 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r127157 | mmichelson | 2008-07-01 16:16:00 -0500 (Tue, 01 Jul 2008) | 8 lines Place the delay in __ast_answer prior to the channel-specific answer callback. This change differs from commit 127113 in that now the channel is not set to AST_STATE_UP until after the answer callback. (closes issue #12924) Reported by: snyfer ........ * main/channel.c, /: Merging Revision 127113 from trunk 2008-07-01 20:52 +0000 [r127153] Jason Parker * Makefile, /: Merged revisions 127152 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r127152 | qwell | 2008-07-01 15:51:43 -0500 (Tue, 01 Jul 2008) | 7 lines Fix a typo that caused this asterisk.conf to not get correctly generated. (closes issue #12966) Reported by: ibc Patches: 12966.patch uploaded by bkruse (license 132) ........ 2008-07-01 20:29 +0000 [r127085-127149] Tilghman Lesher * build_tools/cflags.xml, /, channels/chan_iax2.c: Merged revisions 127143 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r127143 | tilghman | 2008-07-01 15:28:54 -0500 (Tue, 01 Jul 2008) | 10 lines Merged revisions 127133 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r127133 | tilghman | 2008-07-01 15:25:37 -0500 (Tue, 01 Jul 2008) | 2 lines Disable the old, slow search for matching callno in chan_iax2 (but allow it to be reenabled for debugging) ........ ................ * /, channels/chan_iax2.c: Merged revisions 127074 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r127074 | tilghman | 2008-07-01 14:20:25 -0500 (Tue, 01 Jul 2008) | 16 lines Merged revisions 127068 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r127068 | tilghman | 2008-07-01 13:52:53 -0500 (Tue, 01 Jul 2008) | 8 lines Change around how we schedule pings and lagrqs, and fix a reason why the jobs were not getting properly cancelled. (closes issue #12903) Reported by: stevedavies Patches: 20080620__bug12903__2.diff.txt uploaded by Corydon76 (license 14) Tested by: stevedavies ........ ................ 2008-07-01 16:53 +0000 [r127001] Tilghman Lesher * /, channels/chan_iax2.c: Merged revisions 127000 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r127000 | tilghman | 2008-07-01 11:52:29 -0500 (Tue, 01 Jul 2008) | 10 lines Merged revisions 126999 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r126999 | tilghman | 2008-07-01 11:50:46 -0500 (Tue, 01 Jul 2008) | 2 lines Suppress annoying warning by finding the remaining cases where the callno is not in the hash. ........ ................ 2008-07-01 15:05 +0000 [r126756-126904] Olle Johansson * /, channels/chan_sip.c: Merged revisions 126903 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r126903 | oej | 2008-07-01 17:03:59 +0200 (Tis, 01 Jul 2008) | 15 lines Merged revisions 126902 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r126902 | oej | 2008-07-01 16:59:31 +0200 (Tis, 01 Jul 2008) | 7 lines Use domain part of SIP uri in register= configuration as fromdomain. Reported by: one47 Patches: sip-reg-fromdom2.dpatch uploaded by one47 (license 23) (closes issue #12474) ........ ................ * /, channels/chan_sip.c: Merged revisions 126900 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r126900 | oej | 2008-07-01 16:32:15 +0200 (Tis, 01 Jul 2008) | 16 lines Merged revisions 126899 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r126899 | oej | 2008-07-01 16:27:33 +0200 (Tis, 01 Jul 2008) | 8 lines Handle escaped URI's in call pickups. Patch by oej and IgorG. Reported by: IgorG Patches: bug12299-11062-v2.patch uploaded by IgorG (license 20) Tested by: IgorG, oej (closes issue #12299) ........ ................ * /, configs/sip.conf.sample: Merged revisions 126845 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r126845 | oej | 2008-07-01 14:54:57 +0200 (Tis, 01 Jul 2008) | 14 lines Merged revisions 126844 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r126844 | oej | 2008-07-01 14:53:01 +0200 (Tis, 01 Jul 2008) | 5 lines Clear up documentation on "domain=" setting in sip.conf Reported by: davidw (closes issue #12413) ........ ................ * /, channels/chan_sip.c: Merged revisions 126790 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r126790 | oej | 2008-07-01 13:58:17 +0200 (Tis, 01 Jul 2008) | 14 lines Merged revisions 126789 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r126789 | oej | 2008-07-01 13:51:38 +0200 (Tis, 01 Jul 2008) | 6 lines Report 200 OK to all in-dialog OPTIONs requests (to confirm that the dialog exist). Don't bother checking the request URI. (closes issue #11264) Reported by: ibc ........ ................ * /, channels/chan_sip.c: Merged revisions 126755 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r126755 | oej | 2008-07-01 11:51:22 +0200 (Tis, 01 Jul 2008) | 15 lines Merged revisions 126735 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r126735 | oej | 2008-07-01 09:49:15 +0200 (Tis, 01 Jul 2008) | 7 lines Fix bad XML for hold notification. Reported by: gowen72 Patches: hold.patch uploaded by gowen72 (license 432) (closes issue #12942) ........ ................ 2008-06-30 22:34 +0000 [r126676] Jeff Peeler * configs/zapata.conf.sample (removed), configs/chan_dahdi.conf.sample (added), /: Merged revisions 126675 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r126675 | jpeeler | 2008-06-30 17:34:08 -0500 (Mon, 30 Jun 2008) | 1 line rename zapata.conf.sample to chan_dahdi.conf.sample ........ 2008-06-30 20:32 +0000 [r126638] Matthew Fredrickson * channels/chan_dahdi.c, /: Merged revisions 126637 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r126637 | mattf | 2008-06-30 15:25:46 -0500 (Mon, 30 Jun 2008) | 1 line Add support to see MTP2 down events when the link layer drops in SS7 ........ 2008-06-30 16:09 +0000 [r126575] Russell Bryant * /, include/asterisk/lock.h: Merged revisions 126574 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r126574 | russell | 2008-06-30 11:07:25 -0500 (Mon, 30 Jun 2008) | 18 lines Merged revisions 126573 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r126573 | russell | 2008-06-30 11:05:08 -0500 (Mon, 30 Jun 2008) | 10 lines Fix a typo in the non-DEBUG_THREADS version of the recently added DEADLOCK_AVOIDANCE() macro. This caused the lock to not actually be released, and as a result, not avoid deadlocks at all. This resolves the issues reported in the last while about Asterisk locking up all over the place (and most commonly, in chan_iax2). (closes issue #12927) (closes issue #12940) (closes issue #12925) (potentially closes others ...) ........ ................ 2008-06-30 13:07 +0000 [r126518] Olle Johansson * /, channels/chan_sip.c: Merged revisions 126517 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r126517 | oej | 2008-06-30 15:03:53 +0200 (MÃ¥n, 30 Jun 2008) | 20 lines The following patch with some changes for trunk... Merged revisions 126516 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r126516 | oej | 2008-06-30 14:50:55 +0200 (MÃ¥n, 30 Jun 2008) | 10 lines Send all responses to an INVITE reliably, so that we retransmit if we don't get an ACK and also fail if we don't get the very same precious ACK. Based on patch by tsearle, with my own additions. (closes issue #12951) Reported by: tsearle Patches: busy_retransmit.patch uploaded by tsearle (license 373) ........ ................ 2008-06-29 17:02 +0000 [r126362-126364] Kevin P. Fleming * apps/app_zapbarge.c (removed): finish converting this module * pbx/pbx_gtkconsole.c, /, configure, configure.ac, pbx/pbx_lua.c, pbx/Makefile: Merged revisions 126356 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r126356 | kpfleming | 2008-06-29 09:19:29 -0700 (Sun, 29 Jun 2008) | 9 lines various minor fixes created while i worked on getting *every* Asterisk module to build on laptop in dev mode: remove weird pre-setting of LUA paths; they are not necessary; also use the proper path for including the files in pbx_lua.c make the compiler shut up about some ignored function results in pbx_gtkconsole; this module is badly coded anyway ........ * apps/app_dahdibarge.c (added): don't know how this got missed in the DAHDI conversion of this branch 2008-06-29 13:20 +0000 [r126227-126322] Sean Bright * /, cdr/cdr_pgsql.c: Merged revisions 126274 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r126274 | seanbright | 2008-06-29 08:06:46 -0400 (Sun, 29 Jun 2008) | 6 lines Quote column names when inserting CDRs into postgres to avoid conflicts with reserved words. (closes issue #12947) Reported by: panolex ........ 2008-06-28 15:58 +0000 [r126155-126188] Kevin P. Fleming * Makefile, /: update this branch to use the trunk goodness version of menuselect 2008-06-27 22:43 +0000 [r126058-126112] Tilghman Lesher * /, channels/chan_sip.c: Merged revisions 126057 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r126057 | tilghman | 2008-06-27 17:10:34 -0500 (Fri, 27 Jun 2008) | 12 lines Merged revisions 126056 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r126056 | tilghman | 2008-06-27 17:01:09 -0500 (Fri, 27 Jun 2008) | 4 lines When we get a 408 Timeout, don't stop trying to re-register. (closes issue #12863) Reported by: ricvil ........ ................ 2008-06-27 21:00 +0000 [r126023] Mark Michelson * apps/app_queue.c: Port revisions 124661 and 123650 from trunk to 1.6.0 Thanks to Atis Lezdins for pointing this out on the asterisk-dev mailing list 2008-06-27 19:20 +0000 [r125994] Russell Bryant * /, doc/siptls.txt: Merged revisions 125988 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r125988 | russell | 2008-06-27 14:19:08 -0500 (Fri, 27 Jun 2008) | 3 lines Fix a typo. Someone on IRC copied this literally and then wondered why it wasn't working. :) ........ 2008-06-27 19:06 +0000 [r125981-125985] Matthew Fredrickson * channels/chan_dahdi.c, /: Merged revisions 125984 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r125984 | mattf | 2008-06-27 14:05:40 -0500 (Fri, 27 Jun 2008) | 1 line Revert this part of the fix. We'll fix it in libss7 ........ * channels/chan_dahdi.c, /: Merged revisions 125982 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r125982 | mattf | 2008-06-27 14:00:44 -0500 (Fri, 27 Jun 2008) | 1 line Obviously somebody didn't compile with libss7 support when doing the DAHDI conversion. ........ * channels/chan_dahdi.c, /: Merged revisions 125980 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r125980 | mattf | 2008-06-27 13:32:17 -0500 (Fri, 27 Jun 2008) | 1 line Add support for new commands to block/unblock all CICs on a linkset ........ 2008-06-27 17:36 +0000 [r125948] Brett Bryant * /, channels/chan_sip.c: Merged revisions 125947 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r125947 | bbryant | 2008-06-27 12:35:41 -0500 (Fri, 27 Jun 2008) | 1 line Small error in the function that converts peer transports to a string. ........ 2008-06-27 16:29 +0000 [r125892] Brett Bryant * /, channels/chan_sip.c, configs/sip.conf.sample: Merged revisions 125891 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r125891 | bbryant | 2008-06-27 11:28:06 -0500 (Fri, 27 Jun 2008) | 6 lines Change the way that the transport option works for sip users. transport will now take multiple arguments, the first one listed will be the one used for new dialogs, and the rest listed will be acceptable ways for that peer to contact us. This fixes a minor bug where, because SIP TCP/UDP run on the same port, could cause a TCP peer to be saved in the ast_db. There will also be warnings when a transport is changed for an unexpected reason. (issue #12799) ........ 2008-06-27 16:19 +0000 [r125859-125863] Mark Michelson * /, apps/app_queue.c: Merged revisions 125855 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r125855 | mmichelson | 2008-06-27 11:16:13 -0500 (Fri, 27 Jun 2008) | 5 lines Ensure the thread-safety of the monexec variable in app_queue. Thanks to Russell for pointing out the problem ........ 2008-06-27 16:01 +0000 [r125854] Tilghman Lesher * channels/chan_local.c, /: Merged revisions 125853 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r125853 | tilghman | 2008-06-27 11:00:05 -0500 (Fri, 27 Jun 2008) | 3 lines Revert half of the fix, as this part may have been unnecessary (related to issue #12914) Requested here: http://lists.digium.com/pipermail/asterisk-dev/2008-June/033658.html ........ 2008-06-27 14:57 +0000 [r125800-125852] Mark Michelson * main/asterisk.c, main/channel.c, channels/chan_iax2.c: Make sure to only include dahdi/user.h if we have installed DAHDI. * channels/chan_iax2.c: I accidentally committed a change to chan_iax2.c in addition to a change to app_queue.c. Reverting the change to chan_iax2.c, even though it may turn out that this change is necessary. * utils/Makefile, /: Merged revisions 125799 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r125799 | mmichelson | 2008-06-27 09:14:09 -0500 (Fri, 27 Jun 2008) | 3 lines Remove an unneeded target from the Makefile ........ 2008-06-27 14:09 +0000 [r125742-125797] Tilghman Lesher * /, main/utils.c, include/asterisk/lock.h: Merged revisions 125794 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r125794 | tilghman | 2008-06-27 08:54:13 -0500 (Fri, 27 Jun 2008) | 10 lines Merged revisions 125793 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r125793 | tilghman | 2008-06-27 08:45:03 -0500 (Fri, 27 Jun 2008) | 2 lines In this debugging function, copy to a buffer instead of using potentially unsafe pointers. ........ ................ * channels/chan_local.c, /: Merged revisions 125741 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r125741 | tilghman | 2008-06-27 07:28:38 -0500 (Fri, 27 Jun 2008) | 15 lines Merged revisions 125740 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r125740 | tilghman | 2008-06-27 07:19:39 -0500 (Fri, 27 Jun 2008) | 7 lines Add proper deadlock avoidance. (closes issue #12914) Reported by: ozan Patches: 20080625__bug12914.diff.txt uploaded by Corydon76 (license 14) Tested by: ozan ........ ................ 2008-06-27 07:41 +0000 [r125704] Philippe Sultan * /, include/asterisk/jabber.h, res/res_jabber.c: Merged revisions 125703 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r125703 | phsultan | 2008-06-27 09:28:17 +0200 (Fri, 27 Jun 2008) | 1 line Fix a compile time error that occurs if OpenSSL is not installed. Reported by Noel Morais on the users mailing list ........ 2008-06-27 01:09 +0000 [r125648-125684] Mark Michelson * apps/app_queue.c, channels/chan_iax2.c: SENTINEL is not defined in 1.6.0 * /, apps/app_queue.c: Merged revisions 125666 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r125666 | mmichelson | 2008-06-26 19:22:03 -0500 (Thu, 26 Jun 2008) | 3 lines Make this compile with dev-mode on ........ * /, apps/app_queue.c: Merged revisions 125649 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r125649 | mmichelson | 2008-06-26 19:15:54 -0500 (Thu, 26 Jun 2008) | 15 lines The monitor-join option for queues was deprecated in favor of using MixMonitor to mix audio. However, it was pointed out to me that because of this, the command set for the MONITOR_EXEC variable is ignored as well. This means that people can't do their own custom mixing commands at the end of recordings in order to make, for instance, stereo recordings of calls. With this patch, app_queue will set the "joinfiles" variable for the channel's monitor if MONITOR_EXEC is not zero-length. This means that for normal audio mixing, MixMonitor is still the preferred choice, but we allow custom mixing to be done with the two Monitor streams if desired. (closes issue #12923) Reported by: snyfer ........ 2008-06-26 23:06 +0000 [r125592] Mark Michelson * /, apps/app_queue.c: Merged revisions 125591 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r125591 | mmichelson | 2008-06-26 18:06:18 -0500 (Thu, 26 Jun 2008) | 3 lines Fix a really stupid mistake ........ 2008-06-26 23:05 +0000 [r125590] Jason Parker * /, main/utils.c: Merged revisions 125589 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r125589 | qwell | 2008-06-26 18:04:18 -0500 (Thu, 26 Jun 2008) | 9 lines Merged revisions 125587 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r125587 | qwell | 2008-06-26 18:03:15 -0500 (Thu, 26 Jun 2008) | 1 line Make sure to unlock the lock_info lock (huh?). Possible deadlock? ........ ................ 2008-06-26 23:04 +0000 [r125588] Mark Michelson * /, apps/app_queue.c: Merged revisions 125586 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r125586 | mmichelson | 2008-06-26 18:01:02 -0500 (Thu, 26 Jun 2008) | 19 lines Merged revisions 125585 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r125585 | mmichelson | 2008-06-26 17:52:39 -0500 (Thu, 26 Jun 2008) | 11 lines Add the interface of a queue member to the output of the "queue show" command so that it can easily be associated with a queue member's name. This helps so that the appropriate queue member can be removed or paused since the interface is required, not the member's name. (closes issue #12783) Reported by: davevg Patches: app_queue.diff uploaded by davevg (license 209) with small mod from me ........ ................ 2008-06-26 22:50 +0000 [r125584] Tilghman Lesher * /, contrib/scripts/astcli: Merged revisions 125583 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r125583 | tilghman | 2008-06-26 17:49:16 -0500 (Thu, 26 Jun 2008) | 2 lines Don't hang if the command is blank ........ 2008-06-26 22:06 +0000 [r125478-125532] Mark Michelson * /, apps/app_queue.c: Merged revisions 125477 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r125477 | mmichelson | 2008-06-26 15:57:41 -0500 (Thu, 26 Jun 2008) | 19 lines Merged revisions 125476 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r125476 | mmichelson | 2008-06-26 15:56:01 -0500 (Thu, 26 Jun 2008) | 11 lines Prior to this patch, the "queue show" command used cached information for realtime queues instead of giving up-to-date info. Now realtime is queried for the latest and greatest in queue info. (closes issue #12858) Reported by: bcnit Patches: queue_show.patch uploaded by putnopvut (license 60) ........ ................ 2008-06-26 17:07 +0000 [r125388] Olle Johansson * /, channels/chan_sip.c: Merged revisions 125385 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r125385 | oej | 2008-06-26 18:54:22 +0200 (Tor, 26 Jun 2008) | 12 lines Merged revisions 125384 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r125384 | oej | 2008-06-26 18:32:08 +0200 (Tor, 26 Jun 2008) | 3 lines Add support for peer realm based auth (a few missing lines, the rest is well documented but never worked) ........ ................ 2008-06-26 15:52 +0000 [r125280-125334] Kevin P. Fleming * channels/chan_dahdi.c, /: Merged revisions 125333 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r125333 | kpfleming | 2008-06-26 10:50:07 -0500 (Thu, 26 Jun 2008) | 13 lines Merged revisions 125327 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r125327 | kpfleming | 2008-06-26 10:30:33 -0500 (Thu, 26 Jun 2008) | 5 lines ensure that (whenever possible) if we generate a log message because an ioctl() call to DAHDI/Zaptel failed, that we include the reason it failed by including the stringified error number (issue AST-80) ........ ................ * /, res/res_musiconhold.c: Merged revisions 125279 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r125279 | kpfleming | 2008-06-26 07:09:24 -0500 (Thu, 26 Jun 2008) | 2 lines fix compile failure found by buildbot (go, buildbot!) ........ 2008-06-26 11:08 +0000 [r125192-125278] Tilghman Lesher * main/rtp.c, /: Merged revisions 125277 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r125277 | tilghman | 2008-06-26 06:02:11 -0500 (Thu, 26 Jun 2008) | 15 lines Merged revisions 125276 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r125276 | tilghman | 2008-06-26 06:01:21 -0500 (Thu, 26 Jun 2008) | 7 lines Check for rtcp structure before trying to delete schedule. (closes issue #12872) Reported by: destiny6628 Patches: 20080621__bug12872.diff.txt uploaded by Corydon76 (license 14) Tested by: destiny6628 ........ ................ * configs/agents.conf.sample, /: Merged revisions 125223 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r125223 | tilghman | 2008-06-25 20:25:16 -0500 (Wed, 25 Jun 2008) | 12 lines Merged revisions 125218 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r125218 | tilghman | 2008-06-25 20:24:26 -0500 (Wed, 25 Jun 2008) | 4 lines Document ackcall=always. (closes issue #12852) Reported by: davidw ........ ................ * configs/http.conf.sample, /: Merged revisions 125191 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r125191 | tilghman | 2008-06-25 20:11:43 -0500 (Wed, 25 Jun 2008) | 6 lines Update sample configuration to match what are now the defaults for the prefix. (closes issue #12838, related to issue #12198) Reported by: pabelanger Patches: http.conf.diff2 uploaded by pabelanger (license 224) ........ 2008-06-25 23:20 +0000 [r125146] Kevin P. Fleming * main/channel.c, channels/chan_dahdi.c, apps/app_flash.c, configure, codecs/codec_dahdi.c, apps/app_rpt.c, main/asterisk.c, /, apps/app_meetme.c, main/Makefile, apps/app_dahdiscan.c, apps/app_dahdiras.c, configure.ac, include/asterisk/dahdi.h (removed), res/res_musiconhold.c, channels/chan_iax2.c: Merged revisions 125138 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r125138 | kpfleming | 2008-06-25 18:05:28 -0500 (Wed, 25 Jun 2008) | 18 lines Merged revisions 125132 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r125132 | kpfleming | 2008-06-25 17:21:30 -0500 (Wed, 25 Jun 2008) | 10 lines allow tonezone to live in a different place than DAHDI/Zaptel, since dahdi-tools and dahdi-linux are now separate packages and can be installed in different places don't include tonezone.h in dahdi_compat.h, because only a couple of modules need it get app_rpt building again after the DAHDI changes (closes issue #12911) Reported by: tzafrir ........ ................ 2008-06-25 01:13 +0000 [r124964-124967] Tilghman Lesher * channels/chan_dahdi.c, /, include/asterisk/lock.h: Merged revisions 124966 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r124966 | tilghman | 2008-06-24 20:08:37 -0500 (Tue, 24 Jun 2008) | 15 lines Merged revisions 124965 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r124965 | tilghman | 2008-06-24 19:46:24 -0500 (Tue, 24 Jun 2008) | 7 lines Pvt deadlock causes some channels to get stuck in Reserved status. (closes issue #12621) Reported by: fabianoheringer Patches: 20080612__bug12621.diff.txt uploaded by Corydon76 (license 14) Tested by: fabianoheringer ........ ................ * apps/app_voicemail.c, /: Merged revisions 124912 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r124912 | tilghman | 2008-06-24 16:18:52 -0500 (Tue, 24 Jun 2008) | 16 lines Merged revisions 124910 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r124910 | tilghman | 2008-06-24 16:08:52 -0500 (Tue, 24 Jun 2008) | 8 lines Occasionally control characters find their way into CallerID. These need to be stripped prior to placing CallerID in the headers of an email. (closes issue #12759) Reported by: RobH Patches: 20080602__bug12759__2.diff.txt uploaded by Corydon76 (license 14) Tested by: RobH ........ ................ 2008-06-24 17:52 +0000 [r124871-124873] Philippe Sultan * /, res/res_jabber.c: Merged revisions 124872 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r124872 | phsultan | 2008-06-24 19:50:22 +0200 (Tue, 24 Jun 2008) | 6 lines Subscribe to buddy's presence only if we really need to. That is, if the corresponding roster item has a subscription value set to "none" or "from". Make the code more readable. ........ * /, res/res_jabber.c: Merged revisions 124870 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r124870 | phsultan | 2008-06-24 19:28:39 +0200 (Tue, 24 Jun 2008) | 1 line Code simplification ........ 2008-06-23 15:44 +0000 [r124708] Dwayne M. Hubbard * /: blocked revision 124707, taskprocessors are not in 1.6.0 2008-06-22 03:18 +0000 [r124542] Steve Murphy * apps/app_forkcdr.c, /: Merged revisions 124541 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r124541 | murf | 2008-06-21 20:58:06 -0600 (Sat, 21 Jun 2008) | 17 lines Merged revisions 124540 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r124540 | murf | 2008-06-21 20:54:52 -0600 (Sat, 21 Jun 2008) | 9 lines (closes issue #12910) Reported by: chris-mac Sorry, my testing did not contain the simple case of forkCDR(v), I am much embarrassed to admit. If I had, I would have more solidly initialized the opts element for varset. ........ ................ 2008-06-21 12:54 +0000 [r124397-124506] Tilghman Lesher * /, res/res_config_ldap.c: Merged revisions 124505 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r124505 | tilghman | 2008-06-21 07:53:48 -0500 (Sat, 21 Jun 2008) | 4 lines Reduce warning to debug, otherwise we flood the log when we (legitimately) can't find a record. (Closes issue #12908) ........ * apps/app_rpt.c, /: Merged revisions 124451 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r124451 | tilghman | 2008-06-20 18:13:21 -0500 (Fri, 20 Jun 2008) | 14 lines Merged revisions 124450 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r124450 | tilghman | 2008-06-20 18:12:33 -0500 (Fri, 20 Jun 2008) | 6 lines usleep with a value over 1,000,000 is nonportable. Changing to use sleep() instead. (closes issue #12814) Reported by: pputman Patches: app_rtp_sleep.patch uploaded by pputman (license 81) ........ ................ * /, main/app.c: Merged revisions 124396 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r124396 | tilghman | 2008-06-20 17:04:37 -0500 (Fri, 20 Jun 2008) | 11 lines Merged revisions 124395 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r124395 | tilghman | 2008-06-20 17:02:55 -0500 (Fri, 20 Jun 2008) | 3 lines If the last character in a string to be parsed is the delimiter, then we should count that final empty string as an additional argument. ........ ................ 2008-06-20 21:48 +0000 [r124394] Jeff Gehlbach * doc/asterisk-mib.txt, /, doc/digium-mib.txt: Merged revisions 124392-124393 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r124392 | jeffg | 2008-06-20 17:36:01 -0400 (Fri, 20 Jun 2008) | 9 lines Merged revisions 124372 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r124372 | jeffg | 2008-06-20 17:14:40 -0400 (Fri, 20 Jun 2008) | 1 line Fix issues in digium-mib.txt and asterisk-mib.txt to placate smilint - bug 12905 ........ ................ r124393 | jeffg | 2008-06-20 17:43:18 -0400 (Fri, 20 Jun 2008) | 12 lines (Missed committing . on previous commit.....) Merged revisions 124372 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r124372 | jeffg | 2008-06-20 17:14:40 -0400 (Fri, 20 Jun 2008) | 1 line Fix issues in digium-mib.txt and asterisk-mib.txt to placate smilint - bug 12905 ........ ................ ................ 2008-06-20 20:18 +0000 [r124317] Tilghman Lesher * channels/chan_local.c, /: Merged revisions 124316 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r124316 | tilghman | 2008-06-20 15:17:04 -0500 (Fri, 20 Jun 2008) | 16 lines Merged revisions 124315 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r124315 | tilghman | 2008-06-20 15:16:02 -0500 (Fri, 20 Jun 2008) | 8 lines When using a Local channel, started by a call file, with a destination of an AGI script, the AGI script does not always get notified of a hangup if the underlying channel hangs up early. (closes issue #11833) Reported by: IgorG Patches: local_hangup-v1.diff uploaded by IgorG (license 20) ........ ................ 2008-06-20 16:31 +0000 [r124244-124279] Mark Michelson * main/ast_expr2.fl, include/asterisk/doxyref.h, /, main/ast_expr2f.c: Merged revisions 124278 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r124278 | mmichelson | 2008-06-20 11:30:18 -0500 (Fri, 20 Jun 2008) | 6 lines Change references to doc/channelvariables.txt to doc/tex/channelvariables.tex. This issue came up on the asterisk-dev mailing list. ........ * /, channels/chan_sip.c: Merged revisions 124243 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r124243 | mmichelson | 2008-06-20 10:20:11 -0500 (Fri, 20 Jun 2008) | 9 lines Add a missing "ChannelType" header to one of the "PeerStatus" manager events in chan_sip (closes issue #12904) Reported by: eliel Patches: chan_sip.c.patch uploaded by eliel (license 64) ........ 2008-06-19 23:02 +0000 [r124184] Tilghman Lesher * channels/chan_dahdi.c, /: Merged revisions 124183 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r124183 | tilghman | 2008-06-19 17:59:41 -0500 (Thu, 19 Jun 2008) | 15 lines Merged revisions 124182 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r124182 | tilghman | 2008-06-19 17:53:22 -0500 (Thu, 19 Jun 2008) | 7 lines It's possible for a hangup to be received, even just after the initial cid spill. (closes issue #12453) Reported by: Alex728 Patches: 20080604__bug12453.diff.txt uploaded by Corydon76 (license 14) ........ ................ 2008-06-19 20:32 +0000 [r124124] Mark Michelson * apps/app_voicemail.c, /: Merged revisions 124121 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r124121 | mmichelson | 2008-06-19 15:30:23 -0500 (Thu, 19 Jun 2008) | 16 lines Merged revisions 124112 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r124112 | mmichelson | 2008-06-19 15:28:41 -0500 (Thu, 19 Jun 2008) | 8 lines Fix IMAP forwarding so that messages are sent to the proper mailbox. (closes issue #12897) Reported by: jaroth Patches: destination_forward.patch uploaded by jaroth (license 50) ........ ................ 2008-06-19 19:49 +0000 [r124065] Brett Bryant * /, main/utils.c: Merged revisions 124064 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r124064 | bbryant | 2008-06-19 14:48:26 -0500 (Thu, 19 Jun 2008) | 2 lines Add errors that report any locks held by threads when they are being closed. ........ 2008-06-19 18:57 +0000 [r124026] Brett Bryant * /, channels/chan_sip.c: Merged revisions 124024 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r124024 | bbryant | 2008-06-19 13:57:04 -0500 (Thu, 19 Jun 2008) | 2 lines Fix bug in sip registration that sets the default port to 5060 for tls. ........ 2008-06-19 17:58 +0000 [r123871-123989] Tilghman Lesher * /, res/res_config_ldap.c: Merged revisions 123952 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r123952 | tilghman | 2008-06-19 12:22:27 -0500 (Thu, 19 Jun 2008) | 6 lines Don't change pointers that need to be later passed back for deallocation. (closes issue #12572) Reported by: flyn Patches: 20080613__bug12572.diff.txt uploaded by Corydon76 (license 14) ........ * main/channel.c, /: Merged revisions 123931 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r123931 | tilghman | 2008-06-19 12:02:54 -0500 (Thu, 19 Jun 2008) | 13 lines Merged revisions 123930 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r123930 | tilghman | 2008-06-19 11:58:19 -0500 (Thu, 19 Jun 2008) | 5 lines Change informative messages to use the _multiple variant when multiple formats are possible. (Closes issue #12848) Reported by klaus3000 ........ ................ * /, build_tools/strip_nonapi: Merged revisions 123913 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r123913 | tilghman | 2008-06-19 11:26:50 -0500 (Thu, 19 Jun 2008) | 13 lines Merged revisions 123909 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r123909 | tilghman | 2008-06-19 11:26:03 -0500 (Thu, 19 Jun 2008) | 5 lines Only process 40 arguments (20 files) at once with xargs, because some older shells may force xargs to separate on an odd boundary. (Closes issue #12883) Reported by Nik Soggia ........ ................ * /, configs/sip.conf.sample: Merged revisions 123887 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r123887 | tilghman | 2008-06-19 11:21:32 -0500 (Thu, 19 Jun 2008) | 12 lines Merged revisions 123883 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r123883 | tilghman | 2008-06-19 11:20:41 -0500 (Thu, 19 Jun 2008) | 4 lines Correct description of notifyringing option. (Closes issue #12890) Reported by gminet ........ ................ * main/asterisk.c, /: Merged revisions 123870 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r123870 | tilghman | 2008-06-19 11:08:29 -0500 (Thu, 19 Jun 2008) | 14 lines Merged revisions 123869 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r123869 | tilghman | 2008-06-19 11:07:23 -0500 (Thu, 19 Jun 2008) | 6 lines The RDTSC instruction was introduced on the Pentium line of microprocessors, and is not compatible with certain 586 clones, like Cyrix. Hence, asking for i386 compatibility was always incorrect. See http://en.wikipedia.org/wiki/RDTSC (Closes issue #12886) Reported by tecnoxarxa ........ ................ 2008-06-18 22:18 +0000 [r123718-123772] Tilghman Lesher * /, main/say.c, doc/lang (added), doc/lang/hebrew.ods: Merged revisions 123770 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r123770 | tilghman | 2008-06-18 17:17:17 -0500 (Wed, 18 Jun 2008) | 16 lines Merged revisions 123769 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r123769 | tilghman | 2008-06-18 17:08:30 -0500 (Wed, 18 Jun 2008) | 8 lines Add support for saying numbers in Hebrew. (closes issue #11662) Reported by: greenfieldtech Patches: say.c.patch-12042008 uploaded by greenfieldtech (license 369) Hebrew-Sounds.ods uploaded by greenfieldtech (with signficant changes to the spreadsheet by me) ........ ................ * pbx/pbx_spool.c, /: Merged revisions 123715 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r123715 | tilghman | 2008-06-18 15:23:58 -0500 (Wed, 18 Jun 2008) | 15 lines Merged revisions 123710 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r123710 | tilghman | 2008-06-18 15:22:42 -0500 (Wed, 18 Jun 2008) | 7 lines Set the variables top-down, so that if a script sets a variable more than once, the last one will take precedence. (closes issue #12673) Reported by: phber Patches: 20080519__bug12673.diff.txt uploaded by Corydon76 (license 14) ........ ................ 2008-06-18 20:08 +0000 [r123693] Brett Bryant * main/tcptls.c, /: Merged revisions 123692 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r123692 | bbryant | 2008-06-18 15:07:56 -0500 (Wed, 18 Jun 2008) | 2 lines Fix a crash in tcp and tls connections related to reference counts. ........ 2008-06-18 15:09 +0000 [r123651-123653] Mark Michelson * /, apps/app_queue.c: Merged revisions 123652 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r123652 | mmichelson | 2008-06-18 10:08:56 -0500 (Wed, 18 Jun 2008) | 7 lines A portion of the code which handled the 'c' queue option had been removed. No telling when it happened. Anyway, it's back in now and works properly. (Based on issue reported on mailing list) ........ 2008-06-18 12:34 +0000 [r123646-123647] Russell Bryant * apps/app_fax.c: don't use trunk only API for frame data (closes issue #12881) * apps/app_fax.c (added): re-add app_fax ... it got accidentally removed (closes issue #12881) 2008-06-17 21:57 +0000 [r123547] Brett Bryant * main/tcptls.c, main/manager.c, /, channels/chan_sip.c, main/http.c, include/asterisk/tcptls.h: Merged revisions 123546 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r123546 | bbryant | 2008-06-17 16:46:57 -0500 (Tue, 17 Jun 2008) | 5 lines Updates all usages of ast_tcptls_session_instance to be managed by reference counts so that they only get destroyed when all threads are done using them, and memory does not get free'd causing strange issues with SIP. This code was originally written by russellb in the team/group/issue_11972/ branch. ........ 2008-06-17 21:34 +0000 [r123487-123542] Mark Michelson * /, channels/chan_sip.c: Merged revisions 123486 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r123486 | mmichelson | 2008-06-17 15:28:47 -0500 (Tue, 17 Jun 2008) | 12 lines Merged revisions 123485 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r123485 | mmichelson | 2008-06-17 15:26:38 -0500 (Tue, 17 Jun 2008) | 4 lines Make chan_sip build under dev mode with compilers >= GCC 4.2 Thanks to jpeeler for alerting me of this ........ ................ 2008-06-17 20:23 +0000 [r123473] Steve Murphy * /: block 123448 from trunk; it doesn't apply here. 2008-06-17 19:01 +0000 [r123394] Tilghman Lesher * /, channels/chan_iax2.c: Merged revisions 123392 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r123392 | tilghman | 2008-06-17 13:57:45 -0500 (Tue, 17 Jun 2008) | 11 lines Merged revisions 123391 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r123391 | tilghman | 2008-06-17 13:56:53 -0500 (Tue, 17 Jun 2008) | 3 lines Fix 3 more places where failure to lock the structure could cause the wrong lock to be unlocked. (Closes issue #12795) ........ ................ 2008-06-17 18:28 +0000 [r123382-123387] Jeff Peeler * channels/chan_dahdi.c, /: Merged revisions 123238 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r123238 | jpeeler | 2008-06-16 18:05:18 -0500 (Mon, 16 Jun 2008) | 1 line Fix some (more) variables that were forgotten to be renamed, related to 117658 ........ 2008-06-17 18:10 +0000 [r123335] Mark Michelson * /, channels/chan_sip.c: Merged revisions 123334 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r123334 | mmichelson | 2008-06-17 13:09:54 -0500 (Tue, 17 Jun 2008) | 19 lines Merged revisions 123333 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r123333 | mmichelson | 2008-06-17 13:09:16 -0500 (Tue, 17 Jun 2008) | 11 lines Cisco BTS sends SIP responses with a tab between the Cseq number and SIP request method in the Cseq: header. Asterisk did not handle this properly, but with this patch, all is well. (closes issue #12834) Reported by: tobias_e Patches: 12834.patch uploaded by putnopvut (license 60) Tested by: tobias_e ........ ................ 2008-06-17 18:08 +0000 [r123332] Jeff Peeler * doc/tex/configuration.tex, configs/zapata.conf.sample, Makefile, doc/janitor-projects.txt, configs/vpb.conf.sample, doc/sms.txt, contrib/scripts/loadtest.tcl, codecs/codec_dahdi.c (added), configs/smdi.conf.sample, pbx/pbx_config.c, apps/app_chanspy.c, main/asterisk.c, configs/users.conf.sample, doc/ss7.txt, apps/app_meetme.c, configs/rpt.conf.sample, doc/backtrace.txt, doc/tex/queues-with-callback-members.tex, include/asterisk/dahdi.h (added), configs/extensions.ael.sample, res/res_musiconhold.c, configs/meetme.conf.sample, codecs/codec_zap.c (removed), contrib/init.d/rc.mandrake.zaptel, cdr/cdr_csv.c, main/channel.c, doc/tex/manager.tex, doc/tex/sla.tex, include/asterisk/dsp.h, doc/tex/localchannel.tex, apps/app_rpt.c, channels/chan_mgcp.c, contrib/scripts/autosupport, doc/manager_1_1.txt, channels/chan_zap.c (removed), doc/asterisk.8, doc/tex/ael.tex, doc/tex/channelvariables.tex, apps/app_getcpeid.c, doc/tex/enum.tex, apps/app_queue.c, configs/sla.conf.sample, doc/tex/security.tex, include/asterisk/zapata.h (removed), doc/tex/privacy.tex, build_tools/menuselect-deps.in, apps/app_flash.c, main/file.c, doc/osp.txt, contrib/utils/zones2indications.c, utils/extconf.c, makeopts.in, configs/extensions.conf.sample, doc/asterisk.sgml, README, contrib/init.d/rc.mandrake.asterisk, /, include/asterisk/autoconfig.h.in, apps/app_dahdiscan.c (added), apps/app_chanisavail.c, channels/chan_iax2.c, configs/muted.conf.sample, main/loader.c, channels/chan_dahdi.c (added), include/asterisk/doxyref.h, configure, doc/tex/backtrace.tex, apps/app_zapscan.c (removed), doc/tex/app-sms.tex, apps/app_zapras.c (removed), configs/extensions.lua.sample, include/asterisk/options.h, contrib/init.d/rc.suse.asterisk, apps/app_dial.c, apps/app_page.c, doc/tex/hardware.tex, apps/app_fax.c (removed), apps/app_dahdiras.c (added), configure.ac, configs/queues.conf.sample, include/asterisk/channel.h: Goodbye Zaptel, hello DAHDI. Removes Zaptel driver support with DAHDI. Configuration file and dialplan backwards compatability has been put in place where appropiate. Release announcement to follow. 2008-06-17 15:58 +0000 [r123276] Mark Michelson * /, apps/app_queue.c: Merged revisions 123275 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r123275 | mmichelson | 2008-06-17 10:57:43 -0500 (Tue, 17 Jun 2008) | 20 lines Merged revisions 123274 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r123274 | mmichelson | 2008-06-17 10:56:55 -0500 (Tue, 17 Jun 2008) | 12 lines davidw pointed out that the holdtime calculation used by app_queue does not use "boxcar" filtering as the comments say. The term "boxcar" means that the number of samples used to calculate stays constant, with new samples replacing the oldest ones. The queue holdtime calculation uses all holdtime samples collected since the queue was loaded, so the comment has been changed to be accurate. (closes issue #12781) Reported by: davidw ........ ................ 2008-06-17 15:52 +0000 [r123273] Russell Bryant * main/astobj2.c, /: Merged revisions 123272 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r123272 | russell | 2008-06-17 10:52:13 -0500 (Tue, 17 Jun 2008) | 12 lines Merged revisions 123271 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r123271 | russell | 2008-06-17 10:48:31 -0500 (Tue, 17 Jun 2008) | 4 lines Fix a memory leak in astobj2 that was pointed out by seanbright. When a container got destroyed, the underlying bucket list entry for each object that was in the container at that time did not get free'd. ........ ................ 2008-06-16 21:20 +0000 [r123178] Jeff Peeler * channels/chan_zap.c: Fix some variables that were forgotten to be renamed, related to 117658. Couldn't merge from trunk since the chan_dahdi transition has not occurred here yet 2008-06-16 21:19 +0000 [r123173] Steve Murphy * apps/app_stack.c, apps/app_dial.c, main/pbx.c, /, main/features.c, include/asterisk/pbx.h, apps/app_queue.c: Merged revisions 123165 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r123165 | murf | 2008-06-16 14:43:46 -0600 (Mon, 16 Jun 2008) | 19 lines (closes issue #12689) Reported by: ys Many thanks to ys for doing the research on this problem. I didn't think it would be best to unlock the contexts and then relock them after the remove_extension2() call, so I added an extra arg to remove_extension2() and set it appropriately in each call. There were not that many. I considered forcing the code to lock the contexts before the call to remove_extension2(), but that would require a slightly greater degree of changes, especially since the find_context_locked is local to pbx.c I did a simple sanity test to make sure the code doesn't mess things up in general. ........ 2008-06-16 20:03 +0000 [r123112-123116] Tilghman Lesher * channels/chan_mgcp.c, /, channels/chan_sip.c, channels/chan_skinny.c, channels/chan_h323.c, channels/chan_iax2.c: Merged revisions 123114 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r123114 | tilghman | 2008-06-16 14:57:05 -0500 (Mon, 16 Jun 2008) | 10 lines Merged revisions 123113 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r123113 | tilghman | 2008-06-16 14:50:12 -0500 (Mon, 16 Jun 2008) | 2 lines Port "hasvoicemail" change from SIP to other channel drivers ........ ................ * /, channels/chan_sip.c: Merged revisions 123111 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r123111 | tilghman | 2008-06-16 14:23:51 -0500 (Mon, 16 Jun 2008) | 16 lines Merged revisions 123110 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r123110 | tilghman | 2008-06-16 14:21:58 -0500 (Mon, 16 Jun 2008) | 8 lines People expect that if "hasvoicemail" is set in users.conf, even if "mailbox" isn't set, that SIP will detect a mailbox. (closes issue #12855) Reported by: PLL Patches: 20080614__bug12855__2.diff.txt uploaded by Corydon76 (license 14) Tested by: PLL ........ ................ 2008-06-16 17:29 +0000 [r123075] Chris Tooley * apps/app_externalivr.c: Fixes and closes bug number 12804 2008-06-16 12:32 +0000 [r122871-122921] Joshua Colp * /, channels/chan_sip.c: Merged revisions 122920 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r122920 | file | 2008-06-16 09:32:02 -0300 (Mon, 16 Jun 2008) | 14 lines Merged revisions 122919 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r122919 | file | 2008-06-16 09:31:09 -0300 (Mon, 16 Jun 2008) | 6 lines Only compare the first 15 characters so that even if the charset is specified we still accept it as SDP. (closes issue #12803) Reported by: lanzaandrea Patches: chan_sip.c.diff uploaded by lanzaandrea (license 496) ........ ................ * /, channels/chan_sip.c: Merged revisions 122870 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r122870 | file | 2008-06-16 09:09:54 -0300 (Mon, 16 Jun 2008) | 14 lines Merged revisions 122869 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r122869 | file | 2008-06-16 09:08:28 -0300 (Mon, 16 Jun 2008) | 6 lines Don't send a BYE on a dialog that is already gone during a REFER. (closes issue #12865) Reported by: flefoll Patches: chan_sip.c.br14.121495.patch-ALREADYGONE uploaded by flefoll (license 244) ........ ................ 2008-06-13 21:47 +0000 [r122715] Mark Michelson * main/autoservice.c, /: Merged revisions 122714 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r122714 | mmichelson | 2008-06-13 16:45:21 -0500 (Fri, 13 Jun 2008) | 17 lines Merged revisions 122713 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r122713 | mmichelson | 2008-06-13 16:44:53 -0500 (Fri, 13 Jun 2008) | 9 lines Short circuit the loop in autoservice_run if there are no channels to poll. If we continued, then the result would be calling poll() with a NULL pollfd array. While this is fine with POSIX's poll(2) system call, those who use Asterisk's internal poll mechanism (Darwin systems) would have a failed assertion occur when poll is called. (related to issue #10342) ........ ................ 2008-06-13 14:15 +0000 [r122558] Tilghman Lesher * main/dial.c, /: Merged revisions 122557 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r122557 | tilghman | 2008-06-13 09:15:07 -0500 (Fri, 13 Jun 2008) | 7 lines Convert one more delimiter to use comma. (closes issue #12850) Reported by: bcnit Patches: 20080613__bug12850.diff.txt uploaded by Corydon76 (license 14) Tested by: bcnit ........ 2008-06-13 00:18 +0000 [r122467] Jeff Peeler * apps/app_parkandannounce.c, /, main/features.c: Merged revisions 122433 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r122433 | jpeeler | 2008-06-12 18:08:37 -0500 (Thu, 12 Jun 2008) | 4 lines (closes issue 0012193) Reported by: davidw Patch by: Corydon76, modified by me to work properly with ParkAndAnnounce app ........ 2008-06-12 18:54 +0000 [r122313] Mark Michelson * /, apps/app_queue.c: Merged revisions 122312 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r122312 | mmichelson | 2008-06-12 13:53:17 -0500 (Thu, 12 Jun 2008) | 17 lines Merged revisions 122311 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r122311 | mmichelson | 2008-06-12 13:50:58 -0500 (Thu, 12 Jun 2008) | 9 lines Properly play a holdtime message if the announce-holdtime option is set to "once." (closes issue #12842) Reported by: ramonpeek Patches: patch001.diff uploaded by ramonpeek (license 266) ........ ................ 2008-06-12 18:24 +0000 [r122242-122266] Russell Bryant * /, channels/chan_iax2.c: Merged revisions 122262 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r122262 | russell | 2008-06-12 13:23:54 -0500 (Thu, 12 Jun 2008) | 11 lines Merged revisions 122259 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r122259 | russell | 2008-06-12 13:22:44 -0500 (Thu, 12 Jun 2008) | 3 lines Fix some race conditions that cause ast_assert() to report that chan_iax2 tried to remove an entry that wasn't in the scheduler ........ ................ 2008-06-12 15:27 +0000 [r122132-122180] Tilghman Lesher * /, apps/app_meetme.c: Merged revisions 122174 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r122174 | tilghman | 2008-06-12 10:26:07 -0500 (Thu, 12 Jun 2008) | 16 lines Merged revisions 122137 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r122137 | tilghman | 2008-06-12 10:18:39 -0500 (Thu, 12 Jun 2008) | 8 lines Flipflop the sections for two options, since the section for 'X' (exit context) may otherwise absorb keypresses meant for 's' (admin/user menu). (closes issue #12836) Reported by: blitzrage Patches: 20080611__bug12836.diff.txt uploaded by Corydon76 (license 14) Tested by: blitzrage ........ ................ * main/channel.c, /: Merged revisions 122131 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r122131 | tilghman | 2008-06-12 10:14:37 -0500 (Thu, 12 Jun 2008) | 12 lines Merged revisions 122130 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r122130 | tilghman | 2008-06-12 10:11:30 -0500 (Thu, 12 Jun 2008) | 4 lines Occasionally, the alertpipe loses its nonblocking status, so detect and correct that situation before it causes a deadlock. (Reported and tested by ctooley via #asterisk-dev) ........ ................ 2008-06-12 15:01 +0000 [r122126-122129] Steve Murphy * main/cdr.c, apps/app_forkcdr.c, /, CHANGES: Merged revisions 122128 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r122128 | murf | 2008-06-12 08:56:26 -0600 (Thu, 12 Jun 2008) | 9 lines Merged revisions 122127 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r122127 | murf | 2008-06-12 08:51:44 -0600 (Thu, 12 Jun 2008) | 1 line Arkadia tried to warn me, but the code added to ast_cdr_busy, _failed, and _noanswer was redundant. Didn't spot it until I was resolving conflicts in trunk. Ugh. Redundant code removed. It wasn't harmful. Just dumb. ........ ................ * main/cdr.c, apps/app_forkcdr.c, /, funcs/func_cdr.c, include/asterisk/cdr.h, CHANGES: Merged revisions 122091 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r122091 | murf | 2008-06-12 08:28:01 -0600 (Thu, 12 Jun 2008) | 45 lines Merged revisions 122046 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r122046 | murf | 2008-06-12 07:47:34 -0600 (Thu, 12 Jun 2008) | 37 lines (closes issue #10668) Reported by: arkadia Tested by: murf, arkadia Options added to forkCDR() app and the CDR() func to remove some roadblocks for CDR applications. The "show application ForkCDR" output was upgraded to more fully explain the inner workings of forkCDR. The A option was added to forkCDR to force the CDR system to NOT change the disposition on the original CDR, after the fork. This involves ast_cdr_answer, _busy, _failed, and so on. The T option was added to forkCDR to force obedience of the cdr LOCKED flag in the ast_cdr_end, all the disposition changing funcs (ast_cdr_answer, etc), and in the ast_cdr_setvar func. The CHANGES file was updated to explain ALL the new options added to satisfy this bug report (and some requests made verbally and via email, irc, etc, over the past months/year) The 's' option was added to the CDR() func, to force it to skip LOCKED cdr's in the chain. Again, the new options should be totally transparent to existing apps! Current behavior of CDR, forkCDR, and the rest of the CDR system should not change one little bit. Until you add the new options, at least! ........ ................ 2008-06-11 18:57 +0000 [r121915] Matthew Fredrickson * channels/chan_zap.c, /: Merged revisions 121914 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r121914 | mattf | 2008-06-11 13:53:10 -0500 (Wed, 11 Jun 2008) | 1 line Fix pseudo channel allocation errors on startup when using SS7 ........ 2008-06-11 18:20 +0000 [r121872] Tilghman Lesher * main/sched.c, main/channel.c, /, channels/chan_agent.c, main/abstract_jb.c: Merged revisions 121867 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r121867 | tilghman | 2008-06-11 13:19:24 -0500 (Wed, 11 Jun 2008) | 11 lines Merged revisions 121861 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r121861 | tilghman | 2008-06-11 13:18:16 -0500 (Wed, 11 Jun 2008) | 3 lines Make calls to ast_assert() actually test something, so that the error message printed is not nonsensical (reported by mvanbaak via #asterisk-bugs). ........ ................ 2008-06-11 17:59 +0000 [r121858] Matthew Fredrickson * channels/chan_zap.c, /: Merged revisions 121857 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r121857 | mattf | 2008-06-11 12:50:17 -0500 (Wed, 11 Jun 2008) | 1 line Make sure we hangup any calls we have and NULL out the ss7call value when we get a reset circuit message. Fixes crash bug ........ 2008-06-11 17:45 +0000 [r121856] Tilghman Lesher * contrib/scripts/realtime_pgsql.sql, /, UPGRADE.txt, include/asterisk/cdr.h: Merged revisions 121855 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r121855 | tilghman | 2008-06-11 12:44:39 -0500 (Wed, 11 Jun 2008) | 3 lines Expand CDR uniqueid field to 150 chars, to account for maximum systemname. (Closes issue #12831) ........ 2008-06-11 16:13 +0000 [r121806] Jeff Peeler * /, doc/backtrace.txt: Merged revisions 121805 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r121805 | jpeeler | 2008-06-11 11:11:40 -0500 (Wed, 11 Jun 2008) | 9 lines Merged revisions 121804 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r121804 | jpeeler | 2008-06-11 11:11:09 -0500 (Wed, 11 Jun 2008) | 1 line add instructions for logging gdb output via set logging on ........ ................ 2008-06-10 18:36 +0000 [r121598] Sean Bright * /, contrib/scripts/get_ilbc_source.sh: Merged revisions 121597 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r121597 | seanbright | 2008-06-10 14:35:37 -0400 (Tue, 10 Jun 2008) | 14 lines Merged revisions 121596 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r121596 | seanbright | 2008-06-10 14:34:45 -0400 (Tue, 10 Jun 2008) | 6 lines Fixes a problem with some buggy versions of GNU awk (3.1.3) not liking carriage returns in scripts. (closes issue #12749) Reported by: alinux Tested by: Laureano (on #asterisk-dev), juggie ........ ................ 2008-06-10 12:55 +0000 [r121445] Joshua Colp * main/channel.c, /: Merged revisions 121444 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r121444 | file | 2008-06-10 09:54:39 -0300 (Tue, 10 Jun 2008) | 12 lines Merged revisions 121442 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r121442 | file | 2008-06-10 09:52:06 -0300 (Tue, 10 Jun 2008) | 4 lines Update BRIDGEPEER variable before we do a generic bridge in case we just broke out of a native bridge and fell through to generic. (closes issue #12815) Reported by: ramonpeek ........ ................ 2008-06-10 00:53 +0000 [r121404-121408] Russell Bryant * /, channels/chan_iax2.c: Merged revisions 121407 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r121407 | russell | 2008-06-09 19:52:46 -0500 (Mon, 09 Jun 2008) | 2 lines Bump up the debug level of a couple of messages ........ 2008-06-09 16:37 +0000 [r121283] Russell Bryant * main/channel.c, /: Merged revisions 121282 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r121282 | russell | 2008-06-09 11:37:08 -0500 (Mon, 09 Jun 2008) | 18 lines Merged revisions 121280 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r121280 | russell | 2008-06-09 11:35:40 -0500 (Mon, 09 Jun 2008) | 10 lines Do not attempt to do emulation if an END digit is received and the length is less than the defined minimum digit length, and the other end only wants END digits (SIP INFO, for example). (closes issue #12778) Reported by: tsearle Patches: 12778.rev1.txt uploaded by russell (license 2) Tested by: tsearle ........ ................ 2008-06-09 16:36 +0000 [r121281] Tilghman Lesher * main/pbx.c, /: Merged revisions 121279 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r121279 | tilghman | 2008-06-09 11:35:06 -0500 (Mon, 09 Jun 2008) | 6 lines Implement FINDLABEL matching for the new extension matching engine. (closes issue #12800) Reported by: chris-mac Patches: 20080608__bug12800.diff.txt uploaded by Corydon76 (license 14) ........ 2008-06-09 15:10 +0000 [r121231] Mark Michelson * /, channels/chan_agent.c: Merged revisions 121230 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r121230 | mmichelson | 2008-06-09 10:08:58 -0500 (Mon, 09 Jun 2008) | 27 lines Merged revisions 121229 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 (Note that this is being merged to trunk/1.6.0 because it may affect non-callback agents with ackcall set) ........ r121229 | mmichelson | 2008-06-09 10:02:37 -0500 (Mon, 09 Jun 2008) | 16 lines A unique situation of timeouts brought forth a failure situation for autologoff in chan_agent. If using AgentCallbackLogin-style agents, then if the timeout specified by the Dial() to reach the agent's phone was shorter than the timeout specified in queues.conf, then autologoff would only work if the caller hung up while the agent's phone was ringing. This patch allows autologoff to work in this situation when the call in queue transfers to the next available agent (as it would have if the timeout in queues.conf were less than the timeout in the Dial()). (closes issue #12754) Reported by: Rodrigo Patches: 12754.patch uploaded by putnopvut (license 60) Tested by: Rodrigo ........ ................ 2008-06-08 01:43 +0000 [r121138-121164] Jeff Peeler * /, channels/chan_console.c: Merged revisions 121163 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r121163 | jpeeler | 2008-06-07 20:41:59 -0500 (Sat, 07 Jun 2008) | 4 lines This was accidentally reverted. Fixes a bug where if a stream monitor thread was not created (caused from failure of opening or starting the stream) pthread_cancel was called with an invalid thread ID. ........ * apps/app_parkandannounce.c, /: Merged revisions 121131 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r121131 | jpeeler | 2008-06-07 20:16:25 -0500 (Sat, 07 Jun 2008) | 2 lines Fixes segfault when using ParkAndAnnounce. Also, loop made more efficient as announce template only needs to be checked until the number of colon separated arguments run out, not the entire pointer storage array. Was done in a similiar fashion in 1.4, but here we're using less variables. ........ 2008-06-07 14:19 +0000 [r121080] Russell Bryant * channels/chan_local.c, /, channels/chan_agent.c: Merged revisions 121079 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r121079 | russell | 2008-06-07 09:18:44 -0500 (Sat, 07 Jun 2008) | 15 lines Merged revisions 121078 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r121078 | russell | 2008-06-07 09:10:56 -0500 (Sat, 07 Jun 2008) | 7 lines Don't run LIST_HEAD_DESTROY on a STATIC list (closes issue #12807) Reported by: ys Patches: chan_agent_local.diff uploaded by ys (license 281) ........ ................ 2008-06-06 20:25 +0000 [r121011-121047] Tilghman Lesher * main/pbx.c, /: Merged revisions 121010 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r121010 | tilghman | 2008-06-06 14:55:08 -0500 (Fri, 06 Jun 2008) | 6 lines Make extension match characters case-insensitive. (closes issue #12777) Reported by: jsmith Patches: lower_case_patterns-trunk-v1.patch uploaded by jsmith (license 15) ........ 2008-06-06 18:31 +0000 [r120907-120961] Jeff Peeler * /, channels/chan_sip.c: Merged revisions 120960 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r120960 | jpeeler | 2008-06-06 13:30:17 -0500 (Fri, 06 Jun 2008) | 9 lines Merged revisions 120959 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r120959 | jpeeler | 2008-06-06 13:29:14 -0500 (Fri, 06 Jun 2008) | 1 line add another LOW_MEMORY define I forgot ........ ................ * /, channels/chan_sip.c: Merged revisions 120909 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r120909 | jpeeler | 2008-06-06 13:06:06 -0500 (Fri, 06 Jun 2008) | 9 lines Merged revisions 120908 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r120908 | jpeeler | 2008-06-06 13:05:15 -0500 (Fri, 06 Jun 2008) | 1 line only define thread storage variable if necessary for LOW_MEMORY ........ ................ * channels/chan_sip.c: Merged revisions 120906 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r120906 | jpeeler | 2008-06-06 12:50:05 -0500 (Fri, 06 Jun 2008) | 16 lines Merged revisions 120863,120885 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r120863 | jpeeler | 2008-06-06 10:33:15 -0500 (Fri, 06 Jun 2008) | 3 lines This fixes a crash when LOW_MEMORY is turned on. Two allocations of the ast_rtp struct that were previously allocated on the stack have been modified to use thread local storage instead. ........ r120885 | jpeeler | 2008-06-06 11:39:20 -0500 (Fri, 06 Jun 2008) | 2 lines Correction to commmit 120863, make sure proper destructor function is called as well define two thread storage local variables. ........ ................ 2008-06-06 17:35 +0000 [r120864-120905] Tilghman Lesher * /, apps/app_exec.c: Merged revisions 120904 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r120904 | tilghman | 2008-06-06 12:34:21 -0500 (Fri, 06 Jun 2008) | 3 lines For the purpose of making the changed syntax to ExecIf easier to transition, allow the deprecated syntax (fixed for jmls on -dev). ........ 2008-06-05 21:39 +0000 [r120829] Steve Murphy * main/pbx.c, /: Merged revisions 120828 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r120828 | murf | 2008-06-05 15:34:42 -0600 (Thu, 05 Jun 2008) | 1 line a small fix for a crash that occurs when compiling AEL with global vars ........ 2008-06-05 17:17 +0000 [r120677] Philippe Sultan * /, res/res_jabber.c: Merged revisions 120676 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r120676 | phsultan | 2008-06-05 19:02:39 +0200 (Thu, 05 Jun 2008) | 10 lines Merged revisions 120675 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r120675 | phsultan | 2008-06-05 18:56:15 +0200 (Thu, 05 Jun 2008) | 2 lines Ignore appended resource when comparing JIDs. ........ ................ 2008-06-05 16:42 +0000 [r120643-120674] Brett Bryant 2008-06-05 16:01 +0000 [r120566-120603] Tilghman Lesher * apps/app_stack.c, main/loader.c, /, res/res_agi.c: Merged revisions 120602 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r120602 | tilghman | 2008-06-05 10:58:11 -0500 (Thu, 05 Jun 2008) | 4 lines Conditionally load the AGI command gosub, depending on whether or not res_agi has been loaded, fix a return value in the loader, and ensure that the help workhorse header does not print on load. ........ * /, UPGRADE.txt: Merged revisions 120567 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r120567 | tilghman | 2008-06-05 09:35:47 -0500 (Thu, 05 Jun 2008) | 2 lines Add info on the [compat] section of asterisk.conf. ........ * apps/app_fax.c: Fix frame API for 1.6.0 (Closes issue #12793) 2008-06-04 22:08 +0000 [r120515] Mark Michelson * /, apps/app_queue.c: Merged revisions 120514 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r120514 | mmichelson | 2008-06-04 17:07:37 -0500 (Wed, 04 Jun 2008) | 14 lines Merged revisions 120513 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r120513 | mmichelson | 2008-06-04 17:05:33 -0500 (Wed, 04 Jun 2008) | 6 lines Make sure that the string we set will survive the unref of the queue member. Thanks to Russell, who pointed this out. ........ ................ 2008-06-04 20:35 +0000 [r120478] Tilghman Lesher * main/pbx.c, /: Merged revisions 120477 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r120477 | tilghman | 2008-06-04 15:34:52 -0500 (Wed, 04 Jun 2008) | 2 lines MSet doesn't necessarily need chan to be set ........ 2008-06-04 15:38 +0000 [r120338] Joshua Colp * /, pbx/pbx_config.c: Merged revisions 120337 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r120337 | file | 2008-06-04 12:38:00 -0300 (Wed, 04 Jun 2008) | 2 lines We like tabs. ........ 2008-06-04 14:13 +0000 [r120287] Mark Michelson * /, apps/app_queue.c: Merged revisions 120286 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r120286 | mmichelson | 2008-06-04 09:12:45 -0500 (Wed, 04 Jun 2008) | 15 lines Merged revisions 120285 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r120285 | mmichelson | 2008-06-04 09:11:12 -0500 (Wed, 04 Jun 2008) | 7 lines Tab completion when removing a member should give the member's interface, not the name, since the interface is what is expected for the command. (closes issue #12783) Reported by: davevg ........ ................ 2008-06-04 13:34 +0000 [r120284] Joshua Colp * /, pbx/pbx_config.c: Merged revisions 120283 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r120283 | file | 2008-06-04 10:33:59 -0300 (Wed, 04 Jun 2008) | 14 lines Merged revisions 120282 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r120282 | file | 2008-06-04 10:31:09 -0300 (Wed, 04 Jun 2008) | 6 lines Fix a log message and add a message for when the dialplan is done reloading. (closes issue #12716) Reported by: chappell Patches: dialplan_reload_2.diff uploaded by chappell (license 8) ........ ................ 2008-06-03 23:18 +0000 [r120228-120234] Tilghman Lesher * pbx/pbx_loopback.c, /: Merged revisions 120227 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r120227 | tilghman | 2008-06-03 17:42:03 -0500 (Tue, 03 Jun 2008) | 16 lines Merged revisions 120226 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r120226 | tilghman | 2008-06-03 17:41:04 -0500 (Tue, 03 Jun 2008) | 8 lines Due to incorrect use of the AST_LIST_INSERT_HEAD() macro the loopback switch cannot perform any translation on the extension number before searching for it in the target context. (closes issue #12473) Reported by: chappell Patches: pbx_loopback.c.diff uploaded by chappell (license 8) ........ ................ 2008-06-03 22:18 +0000 [r120178] Jeff Peeler * main/config.c: Merged revisions 120174 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r120174 | jpeeler | 2008-06-03 17:17:07 -0500 (Tue, 03 Jun 2008) | 14 lines Merged revisions 120173 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r120173 | jpeeler | 2008-06-03 17:15:33 -0500 (Tue, 03 Jun 2008) | 6 lines (closes issue #11594) Reported by: yem Tested by: yem This change decreases the buffer size allocated on the stack substantially in config_text_file_load when LOW_MEMORY is turned on. This change combined with the fix from revision 117462 (making mkintf not copy the zt_chan_conf structure) was enough to prevent the crash. ........ ................ 2008-06-03 22:08 +0000 [r120172] Tilghman Lesher * include/asterisk/options.h, main/asterisk.c, Makefile, main/pbx.c, /, res/res_agi.c, pbx/pbx_realtime.c, configs/pbx_realtime.conf (removed): Merged revisions 120171 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r120171 | tilghman | 2008-06-03 17:05:16 -0500 (Tue, 03 Jun 2008) | 5 lines Move compatibility options into asterisk.conf, default them to on for upgrades, and off for new installations. This includes the translation from pipes to commas for pbx_realtime and the EXEC command for AGI, as well as the change to the Set application not to support multiple variables at once. ........ 2008-06-03 21:35 +0000 [r120170] Russell Bryant * /, channels/chan_iax2.c: Merged revisions 120169 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r120169 | russell | 2008-06-03 16:35:11 -0500 (Tue, 03 Jun 2008) | 12 lines Merged revisions 120168 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r120168 | russell | 2008-06-03 16:34:55 -0500 (Tue, 03 Jun 2008) | 4 lines Fix another place where peer->callno could change at a very bad time, and also fix a place where a peer was used after the reference was released. (inspired by rev 120001) ........ ................ 2008-06-03 16:24 +0000 [r120034] Tilghman Lesher * /, channels/chan_iax2.c: Merged revisions 120012 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r120012 | tilghman | 2008-06-03 11:19:35 -0500 (Tue, 03 Jun 2008) | 17 lines Merged revisions 120001 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r120001 | tilghman | 2008-06-03 11:10:53 -0500 (Tue, 03 Jun 2008) | 9 lines Save the callno when we're poking, because our peer structure could change during destruction (and thus we unlock the wrong callno, causing a cascade failure). (closes issue #12717) Reported by: gewfie Patches: 20080525__bug12717.diff.txt uploaded by Corydon76 (license 14) Tested by: gewfie ........ ................ 2008-06-03 15:57 +0000 [r119931-120000] Steve Murphy * pbx/ael/ael-test/ref.ael-test8, pbx/ael/ael-test/ref.ael-test18, pbx/ael/ael-test/ref.ael-vtest21, pbx/ael/ael-test/ref.ael-test19, pbx/ael/ael-test/ref.ael-vtest13, pbx/ael/ael-test/ref.ael-vtest17, pbx/ael/ael-test/ref.ael-ntest10, pbx/ael/ael-test/ref.ael-test1, pbx/ael/ael-test/ref.ael-test3, pbx/ael/ael-test/ref.ael-test5, pbx/ael/ael-test/ref.ael-test15: Merged revisions 119998 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r119998 | murf | 2008-06-03 09:49:34 -0600 (Tue, 03 Jun 2008) | 16 lines Merged revisions 119966 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r119966 | murf | 2008-06-03 09:26:56 -0600 (Tue, 03 Jun 2008) | 8 lines Updated the regressions on AEL. Hadn't updated this for the changes I made to preserve ${EXTEN} in switches, which affected several tests because it adds extra priorities, and at least one needed to be updated because of the removal of the empty extension warning message. ........ ................ * res/ael/pval.c, /: Merged revisions 119930 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r119930 | murf | 2008-06-03 09:07:20 -0600 (Tue, 03 Jun 2008) | 24 lines Merged revisions 119929 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r119929 | murf | 2008-06-03 08:49:46 -0600 (Tue, 03 Jun 2008) | 16 lines as per http://lists.digium.com/pipermail/asterisk-users/2008-June/212934.html, which is a message from Philipp Kempgen, requesting that the WARNING that an extension is empty be reduced to a NOTICE or less, as empty extensions are syntactically possible, and no big deal. With which I agree, and have removed that WARNING message entirely. I think it is not necessary to see this message. It didn't state that a NoOp() was inserted automatically on your behalf, and really, as users, who cares? Why freak out dialplan writers with unnecessary warnings? The details of the machinations a compiler goes thru to produce working assembly code is of little interest to most programmers-- we will follow the unix principal of doing our work silently. ........ ................ 2008-06-03 14:48 +0000 [r119928] Joshua Colp * /, channels/chan_sip.c: Merged revisions 119927 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r119927 | file | 2008-06-03 11:47:54 -0300 (Tue, 03 Jun 2008) | 10 lines Merged revisions 119926 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r119926 | file | 2008-06-03 11:46:24 -0300 (Tue, 03 Jun 2008) | 2 lines Treat ECONNREFUSED as an error that will stop further retransmissions. (issue #AST-58, patch from Switchvox) ........ ................ 2008-06-03 13:30 +0000 [r119745-119893] Russell Bryant * /, main/logger.c: Merged revisions 119892 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r119892 | russell | 2008-06-03 08:29:16 -0500 (Tue, 03 Jun 2008) | 9 lines Do a deep copy of file and function strings to avoid a potential crash when modules are unloaded. (closes issue #12780) Reported by: ys Patches: logger.diff uploaded by ys (license 281) -- modified by me for coding guidelines ........ * /, channels/chan_iax2.c: Merged revisions 119839 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r119839 | russell | 2008-06-02 15:08:24 -0500 (Mon, 02 Jun 2008) | 15 lines Merged revisions 119838 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r119838 | russell | 2008-06-02 15:08:04 -0500 (Mon, 02 Jun 2008) | 7 lines Revert a change made for issue #12479. This change caused a regression such that a dial string such as (IAX2/foo) did not automatically fall back to dialing the 's' extension anymore. (closes issue #12770) Reported by: dagmoller ........ ................ * /, apps/app_fax.c (added): Merged revisions 119801 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r119801 | russell | 2008-06-02 11:14:15 -0500 (Mon, 02 Jun 2008) | 4 lines Add app_fax from asterisk-addons, with some additional changes to resolve compiler warnings, as well as update to the APIs in spandsp 0.0.5. Spandsp 0.0.5 is being distributed under the LGPL, so we can move this module into the main tree. ........ * /, configure, include/asterisk/autoconfig.h.in, configure.ac: Merged revisions 119799 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r119799 | russell | 2008-06-02 10:57:43 -0500 (Mon, 02 Jun 2008) | 4 lines After determining that the version of spandsp installed is an acceptable version, do a build and link test to ensure that the library is usable, and that libtiff is also available ........ * /, build_tools/menuselect-deps.in, configure, include/asterisk/autoconfig.h.in, configure.ac, makeopts.in: Merged revisions 119795 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r119795 | russell | 2008-06-02 10:43:40 -0500 (Mon, 02 Jun 2008) | 2 lines Add a configure script check for spandsp ........ * main/manager.c, /: Merged revisions 119744 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r119744 | russell | 2008-06-02 09:41:55 -0500 (Mon, 02 Jun 2008) | 13 lines Merged revisions 119742 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r119742 | russell | 2008-06-02 09:39:45 -0500 (Mon, 02 Jun 2008) | 5 lines Improve CLI command blacklist checking for the command manager action. Previously, it did not handle case or whitespace properly. This made it possible for blacklisted commands to get executed anyway. (closes issue #12765) ........ ................ 2008-06-02 14:40 +0000 [r119743] Philippe Sultan * channels/chan_jingle.c, /, channels/chan_gtalk.c, res/res_jabber.c: Merged revisions 119741 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r119741 | phsultan | 2008-06-02 16:35:24 +0200 (Mon, 02 Jun 2008) | 13 lines Do not link the guest account with any configured XMPP client (in jabber.conf). The actual connection is made when a call comes in Asterisk. Apply this fix to Jingle too. Fix the ast_aji_get_client function that was not able to retrieve an XMPP client from its JID. (closes issue #12085) Reported by: junky Tested by: phsultan ........ 2008-06-02 12:32 +0000 [r119532-119690] Russell Bryant * channels/misdn/isdn_lib.c, channels/misdn/isdn_lib.h, /: Merged revisions 119586,119637 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r119586 | crichter | 2008-06-02 03:46:23 -0500 (Mon, 02 Jun 2008) | 9 lines Merged revisions 119585 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r119585 | crichter | 2008-06-02 10:35:28 +0200 (Mo, 02 Jun 2008) | 1 line Added counter for unhandled_bmsg Print, this prevents the logs to be flooded to fast and save CPU in this error scenario. Added 'last_used' element to bc structure, when a bchannel changes from used to free this exact time will be marked in last_used. When a new channel is requested the find_free_chan function will check if the new empty channel was used within the last second, if yes it will search for the next channel, if no it will return this channel. This simple mechanism has prooven to prevent race conditions where the NT and TE tried to allocate the exact same channel at the same time (RELEASE cause: 44). ........ ................ r119637 | crichter | 2008-06-02 04:35:04 -0500 (Mon, 02 Jun 2008) | 9 lines Merged revisions 119636 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r119636 | crichter | 2008-06-02 11:29:21 +0200 (Mo, 02 Jun 2008) | 1 line fixed compile issue when dev-mode is enabled ........ ................ * /, channels/chan_iax2.c: Merged revisions 119688 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r119688 | russell | 2008-06-02 07:30:42 -0500 (Mon, 02 Jun 2008) | 11 lines Merged revisions 119687 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r119687 | russell | 2008-06-02 07:30:17 -0500 (Mon, 02 Jun 2008) | 3 lines Even of the first PING or LAGRQ doesn't get sent because it comes up too soon, make sure to reschedule so it gets sent later. ........ ................ * /, channels/chan_iax2.c: Merged revisions 119534 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r119534 | russell | 2008-06-01 20:08:16 -0500 (Sun, 01 Jun 2008) | 10 lines Merged revisions 119533 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r119533 | russell | 2008-06-01 20:06:09 -0500 (Sun, 01 Jun 2008) | 2 lines Change a debug message to an actual debug message ........ ................ * apps/app_dial.c, /: Merged revisions 119531 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r119531 | russell | 2008-06-01 20:04:01 -0500 (Sun, 01 Jun 2008) | 10 lines Merged revisions 119530 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r119530 | russell | 2008-06-01 20:03:22 -0500 (Sun, 01 Jun 2008) | 2 lines Fix another typo in documentation ........ ................ 2008-06-01 21:59 +0000 [r119529] Michiel van Baak * apps/app_dial.c, /: Merged revisions 119479 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r119479 | mvanbaak | 2008-06-01 23:06:27 +0200 (Sun, 01 Jun 2008) | 10 lines Merged revisions 119478 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r119478 | mvanbaak | 2008-06-01 22:47:55 +0200 (Sun, 01 Jun 2008) | 2 lines small typo fix 'retires' => 'retries' ........ ................ 2008-05-30 21:24 +0000 [r119420] Tilghman Lesher * /, apps/app_queue.c: Merged revisions 119419 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r119419 | tilghman | 2008-05-30 16:23:14 -0500 (Fri, 30 May 2008) | 14 lines Merged revisions 119404 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r119404 | tilghman | 2008-05-30 16:17:45 -0500 (Fri, 30 May 2008) | 6 lines When joinempty=strict, it only failed on join if there were busy members. If all members were logged out OR paused, then it (incorrectly) let callers join the queue. (closes issue #12451) Reported by: davidw ........ ................ 2008-05-30 19:48 +0000 [r119356] Joshua Colp * main/autoservice.c, /: Merged revisions 119355 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r119355 | file | 2008-05-30 16:47:30 -0300 (Fri, 30 May 2008) | 10 lines Merged revisions 119354 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r119354 | file | 2008-05-30 16:46:37 -0300 (Fri, 30 May 2008) | 2 lines Fix a bug I found while testing for another issue. ........ ................ 2008-05-30 17:13 +0000 [r119304] Tilghman Lesher * apps/app_stack.c: Oops, broke 1.6 (thanks MattF) 2008-05-30 16:57 +0000 [r119303] Michiel van Baak * contrib/scripts/safe_asterisk, contrib/init.d/rc.suse.asterisk, contrib/init.d/rc.debian.asterisk, contrib/init.d/rc.mandrake.asterisk, /, contrib/init.d/rc.redhat.asterisk, contrib/init.d/rc.gentoo.asterisk, contrib/init.d/rc.slackware.asterisk: Merged revisions 119302 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r119302 | mvanbaak | 2008-05-30 18:47:24 +0200 (Fri, 30 May 2008) | 22 lines Merged revisions 119301 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r119301 | mvanbaak | 2008-05-30 18:44:39 +0200 (Fri, 30 May 2008) | 14 lines dont use a bashism way to check the $VERSION variable. The rc/init.d scripts, and safe_asterisk work on normal sh now again. Tested on: OpenBSD 4.2 (me) Debian etch (me) Ubuntu Hardy (me and loloski) FC9 (loloski) (closes issue #12687) Reported by: loloski Patches: 20080529-12687-safe_asterisk-fixversion.diff.txt uploaded by mvanbaak (license 7) Tested by: loloski, mvanbaak ........ ................ 2008-05-30 16:40 +0000 [r119297-119300] Tilghman Lesher * apps/app_stack.c, /: Merged revisions 119299 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r119299 | tilghman | 2008-05-30 11:40:13 -0500 (Fri, 30 May 2008) | 2 lines Suppress warning about pbx structure already existing ........ * apps/app_stack.c, apps/app_dial.c, include/asterisk/agi.h, /, CHANGES: Merged revisions 119296 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r119296 | tilghman | 2008-05-30 11:10:46 -0500 (Fri, 30 May 2008) | 8 lines Add native AGI command GOSUB, as invoking Gosub with EXEC does not work properly. (closes issue #12760) Reported by: Corydon76 Patches: 20080530__bug12760.diff.txt uploaded by Corydon76 (license 14) Tested by: tim_ringenbach, Corydon76 ........ 2008-05-30 13:01 +0000 [r119158-119240] Russell Bryant * /, channels/chan_iax2.c: Merged revisions 119239 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r119239 | russell | 2008-05-30 07:59:11 -0500 (Fri, 30 May 2008) | 23 lines Merged revisions 119238 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r119238 | russell | 2008-05-30 07:55:36 -0500 (Fri, 30 May 2008) | 15 lines Merged revisions 119237 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r119237 | russell | 2008-05-30 07:49:39 -0500 (Fri, 30 May 2008) | 7 lines - Instead of only enforcing destination call number checking on an ACK, check all full frames except for PING and LAGRQ, which may be sent by older versions too quickly to contain the destination call number. (As suggested by Tim Panton on the asterisk-dev list) - Merge changes from team/russell/iax2-frame-race, which prevents PING and LAGRQ from being sent before the destination call number is known. ........ ................ ................ * main/autoservice.c, /: Merged revisions 119157 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r119157 | russell | 2008-05-29 17:28:50 -0500 (Thu, 29 May 2008) | 18 lines Merged revisions 119156 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r119156 | russell | 2008-05-29 17:24:29 -0500 (Thu, 29 May 2008) | 10 lines Fix a race condition in channel autoservice. There was still a small window of opportunity for a DTMF frame, or some other deferred frame type, to come in and get dropped. (closes issue #12656) (closes issue #12656) Reported by: dimas Patches: v3-12656.patch uploaded by dimas (license 88) -- with some modifications by me ........ ................ 2008-05-29 20:26 +0000 [r119073] Tilghman Lesher * channels/chan_zap.c, /: Merged revisions 119072 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r119072 | tilghman | 2008-05-29 15:25:33 -0500 (Thu, 29 May 2008) | 15 lines Merged revisions 119071 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r119071 | tilghman | 2008-05-29 15:24:11 -0500 (Thu, 29 May 2008) | 7 lines Call waiting tone occurs too often, because it's getting serviced by both subchannels. (closes issue #11354) Reported by: cahen Patches: 20080512__bug11354.diff.txt uploaded by Corydon76 (license 14) ........ ................ 2008-05-29 19:06 +0000 [r118960-119014] Russell Bryant * apps/app_milliwatt.c, /: Merged revisions 119013 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r119013 | russell | 2008-05-29 14:05:33 -0500 (Thu, 29 May 2008) | 12 lines Merged revisions 119012 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r119012 | russell | 2008-05-29 14:04:52 -0500 (Thu, 29 May 2008) | 4 lines - Fix a typo in the argument to Playtones - use ast_safe_sleep() instead of calling the wait application (thanks to tilghman for pointing these out!) ........ ................ * /, channels/chan_iax2.c: Merged revisions 119010 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r119010 | russell | 2008-05-29 13:54:11 -0500 (Thu, 29 May 2008) | 24 lines Merged revisions 119009 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r119009 | russell | 2008-05-29 13:49:12 -0500 (Thu, 29 May 2008) | 16 lines Merged revisions 119008 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r119008 | russell | 2008-05-29 13:45:21 -0500 (Thu, 29 May 2008) | 7 lines Merge changes from team/russell/iax2-another-fix-to-the-fix As described in the following post to the asterisk-dev mailing list, only enforce destination call numbers when processing an ACK. http://lists.digium.com/pipermail/asterisk-dev/2008-May/033217.html (closes issue #12631) ........ ................ ................ * apps/app_milliwatt.c, /: Merged revisions 118962 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r118962 | russell | 2008-05-29 12:52:00 -0500 (Thu, 29 May 2008) | 11 lines Merged revisions 118961 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r118961 | russell | 2008-05-29 12:51:29 -0500 (Thu, 29 May 2008) | 3 lines - Mark app_milliwatt dependent on res_indications (thanks to jsmith) - fix a typo in a log message (thanks to qwell) ........ ................ * apps/app_milliwatt.c, /: Merged revisions 118959 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r118959 | russell | 2008-05-29 12:46:04 -0500 (Thu, 29 May 2008) | 11 lines Merged revisions 118956 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r118956 | russell | 2008-05-29 12:38:38 -0500 (Thu, 29 May 2008) | 3 lines Change milliwatt to use the proper tone by default (1004 Hz) instead of 1000 Hz. An option is there to use 1000 Hz for anyone that might want it. ........ ................ 2008-05-29 17:42 +0000 [r118958] Tilghman Lesher * channels/chan_mgcp.c, channels/chan_zap.c, /, channels/chan_agent.c, channels/chan_alsa.c, main/utils.c, include/asterisk/lock.h, channels/chan_iax2.c: Merged revisions 118955,118957 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r118955 | tilghman | 2008-05-29 12:35:19 -0500 (Thu, 29 May 2008) | 11 lines Merged revisions 118953 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r118953 | tilghman | 2008-05-29 12:20:16 -0500 (Thu, 29 May 2008) | 3 lines Add some debugging code that ensures that when we do deadlock avoidance, we don't lose the information about how a lock was originally acquired. ........ ................ r118957 | tilghman | 2008-05-29 12:39:50 -0500 (Thu, 29 May 2008) | 10 lines Merged revisions 118954 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r118954 | tilghman | 2008-05-29 12:33:01 -0500 (Thu, 29 May 2008) | 2 lines Define also when not DEBUG_THREADS ........ ................ 2008-05-29 04:11 +0000 [r118909] Steve Murphy * main/cdr.c, apps/app_forkcdr.c, /: Merged revisions 118880 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r118880 | murf | 2008-05-28 19:29:09 -0600 (Wed, 28 May 2008) | 54 lines Merged revisions 118858 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r118858 | murf | 2008-05-28 18:25:28 -0600 (Wed, 28 May 2008) | 46 lines (closes issue #10668) (closes issue #11721) (closes issue #12726) Reported by: arkadia Tested by: murf These changes: 1. revert the changes made via bug 10668; I should have known that such changes, even tho they made sense at the time, seemed like an omission, etc, were actually integral to the CDR system via forkCDR. It makes sense to me now that forkCDR didn't natively end any CDR's, but rather depended on natively closing them all at hangup time via traversing and closing them all, whether locked or not. I still don't completely understand the benefits of setvar and answer operating on locked cdrs, but I've seen enough to revert those changes also, and stop messing up users who depended on that behavior. bug 12726 found reverting the changes fixed his changes, and after a long review and working on forkCDR, I can see why. 2. Apply the suggested enhancements proposed in 10668, but in a completely compatible way. ForkCDR will behave exactly as before, but now has new options that will allow some actions to be taken that will slightly modify the outcome and side-effects of forkCDR. Based on conversations I've had with various people, these small tweaks will allow some users to get the behavior they need. For instance, users executing forkCDR in an AGI script will find the answer time set, and DISPOSITION set, a situation not covered when the routines were first written. 3. A small problem in the cdr serializer would output answer and end times even when they were not set. This is now fixed. ........ ................ 2008-05-28 18:07 +0000 [r118781] Michiel van Baak * /, channels/chan_skinny.c: Merged revisions 118750 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r118750 | mvanbaak | 2008-05-28 19:58:21 +0200 (Wed, 28 May 2008) | 2 lines remove unused astobj.h header file from chan_skinny.c ........ 2008-05-28 14:31 +0000 [r118648] Joshua Colp * /, channels/chan_sip.c, configs/sip.conf.sample, CHANGES: Merged revisions 118647 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r118647 | file | 2008-05-28 11:29:01 -0300 (Wed, 28 May 2008) | 12 lines Merged revisions 118646 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r118646 | file | 2008-05-28 11:23:34 -0300 (Wed, 28 May 2008) | 4 lines Add an option to use the source IP address of RTP as the destination IP address of UDPTL when a specific option is enabled. If the remote side is properly configured (ports forwarded) then UDPTL will flow. (closes issue #10417) Reported by: cstadlmann ........ ................ 2008-05-28 14:13 +0000 [r118615-118645] Philippe Sultan * channels/chan_jingle.c, /, include/asterisk/jingle.h: Merged revisions 118644 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r118644 | phsultan | 2008-05-28 16:10:48 +0200 (Wed, 28 May 2008) | 10 lines Changed to temporary namespaces to match with latest XEPs. As soon as Jingle is completely standardized, we can set those namespaces to their final values. Added two attributes to the jingle_pvt struct to store the content name attributes. Reported by Robert McQueen on Telepathy's framework mailing list : http://lists.freedesktop.org/archives/telepathy/2008-May/001971.html Keeping working on our Jingle stack! ........ * channels/chan_jingle.c, /: Merged revisions 118614 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r118614 | phsultan | 2008-05-28 10:39:10 +0200 (Wed, 28 May 2008) | 1 line Code simplification ........ 2008-05-27 19:35 +0000 [r118561] Joshua Colp * /, channels/chan_sip.c: Merged revisions 118560 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r118560 | file | 2008-05-27 16:34:14 -0300 (Tue, 27 May 2008) | 12 lines Merged revisions 118558 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r118558 | file | 2008-05-27 16:32:38 -0300 (Tue, 27 May 2008) | 4 lines Fix an issue where codec preferences were not set on dialogs that were not authenticated via a user or peer and allow framing to work without rtpmap in the SDP. (closes issue #12501) Reported by: slimey ........ ................ 2008-05-27 19:28 +0000 [r118557] Russell Bryant * /, include/asterisk/compat.h: Merged revisions 118556 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r118556 | russell | 2008-05-27 14:27:48 -0500 (Tue, 27 May 2008) | 6 lines Add printf format attribute for vasprintf(). (closes issue #12729) Reported by: snuffy Patches: bug_12729.diff uploaded by snuffy (license 35) ........ 2008-05-27 19:22 +0000 [r118555] Tilghman Lesher * main/cli.c, /: Merged revisions 118554 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r118554 | tilghman | 2008-05-27 14:21:03 -0500 (Tue, 27 May 2008) | 14 lines Merged revisions 118551 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r118551 | tilghman | 2008-05-27 14:15:27 -0500 (Tue, 27 May 2008) | 6 lines When showing an error message for a command, don't shorten the command output, as it tends to confuse the user (it's fine for suggesting other commands, however). Reported by: seanbright (on #asterisk-dev) Fixed by: me ........ ................ 2008-05-27 19:09 +0000 [r118518] Mark Michelson * apps/app_chanspy.c, /: Merged revisions 118514 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r118514 | mmichelson | 2008-05-27 14:08:24 -0500 (Tue, 27 May 2008) | 19 lines Merged revisions 118509 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r118509 | mmichelson | 2008-05-27 14:07:26 -0500 (Tue, 27 May 2008) | 11 lines Russell noted to me that in the case that separate threads use their own addressing system, the fix I made for issue 12376 does not guarantee uniqueness to the datastores' uids. Though I know of no system that works this way, I am going to change this right now to prevent trying to track down some future bug that may occur and cause untold hours of debugging time to track down. The change involves using a global counter which increases with each new chanspy_ds which is created. This guarantees uniqueness. ........ ................ 2008-05-27 18:59 +0000 [r118471] Tilghman Lesher * main/asterisk.c, /: Merged revisions 118466 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r118466 | tilghman | 2008-05-27 13:59:06 -0500 (Tue, 27 May 2008) | 16 lines Merged revisions 118465 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r118465 | tilghman | 2008-05-27 13:58:09 -0500 (Tue, 27 May 2008) | 8 lines NULL character should terminate only commands back to the core, not log messages to the console. (closes issue #12731) Reported by: seanbright Patches: 20080527__bug12731.diff.txt uploaded by Corydon76 (license 14) Tested by: seanbright ........ ................ 2008-05-27 17:25 +0000 [r118418] Michiel van Baak * apps/app_voicemail.c: small update to the g() option of app_voicemail to note that gain changes only work on zap channels right now. issue #12578 shows it's not clear right now. 2008-05-27 16:48 +0000 [r118378-118382] Mark Michelson * apps/app_chanspy.c, /: Merged revisions 118371 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r118371 | mmichelson | 2008-05-27 11:43:36 -0500 (Tue, 27 May 2008) | 22 lines Merged revisions 118365 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r118365 | mmichelson | 2008-05-27 11:38:38 -0500 (Tue, 27 May 2008) | 14 lines Add a unique id to the datastore allocated in app_chanspy since it is possible that multiple spies may be listening to the same channel. (closes issue #12376) Reported by: DougUDI Patches: 12376_chanspy_uid.diff uploaded by putnopvut (license 60) Tested by: destiny6628 (closes issue #12243) Reported by: atis ........ ................ * /: Hmm, I apparently forgot to commit the block of revision 118175. Now I'm doing it. 2008-05-27 15:47 +0000 [r118360] Tilghman Lesher * /, configs/queues.conf.sample: Merged revisions 118359 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r118359 | tilghman | 2008-05-27 10:46:58 -0500 (Tue, 27 May 2008) | 11 lines Merged revisions 118358 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r118358 | tilghman | 2008-05-27 10:45:37 -0500 (Tue, 27 May 2008) | 3 lines Add a note that pbx_config.so is needed for Local channels. (Closes issue #12671) ........ ................ 2008-05-27 14:51 +0000 [r118331] Russell Bryant * /, include/asterisk/compat.h: Merged revisions 118328 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r118328 | russell | 2008-05-27 09:51:13 -0500 (Tue, 27 May 2008) | 2 lines Add printf attribute to asprintf ........ 2008-05-27 13:30 +0000 [r118301-118303] Tilghman Lesher * /, res/res_config_ldap.c: Merged revisions 118302 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r118302 | tilghman | 2008-05-27 08:30:10 -0500 (Tue, 27 May 2008) | 6 lines When binding anonymously, credentials are still needed. (closes issue #12601) Reported by: suretec Patches: res_config_ldap.c.patch uploaded by suretec (license 70) ........ * /, pbx/pbx_realtime.c: Merged revisions 118300 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r118300 | tilghman | 2008-05-27 08:13:17 -0500 (Tue, 27 May 2008) | 4 lines In compat14 mode, don't translate pipes inside expressions, as they aren't argument delimiters, but rather 'or' symbols. (Closes issue #12723) ........ 2008-05-25 16:20 +0000 [r118253] Tilghman Lesher * /, channels/chan_sip.c: Merged revisions 118252 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r118252 | tilghman | 2008-05-25 11:17:05 -0500 (Sun, 25 May 2008) | 20 lines Merged revisions 118251 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r118251 | tilghman | 2008-05-25 11:02:04 -0500 (Sun, 25 May 2008) | 12 lines Realtime flag affects construction in multiple ways, so consulting whether rtcachefriends was set was done too soon (needed to be done inside build_peer, not just as a flag to build_peer). Also, fullcontact needed to be reconstructed, because realtime separates the embedded ';' into multiple fields. (closes issue #12722) Reported by: barthpbx Patches: 20080525__bug12722.diff.txt uploaded by Corydon76 (license 14) Tested by: barthpbx (Much of the discussion happened on #asterisk-dev for diagnosing this issue) ........ ................ 2008-05-24 01:15 +0000 [r118177-118179] Jeff Peeler * doc/api-1.6.0-changes.odt (added), /: Merged revisions 118178 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r118178 | jpeeler | 2008-05-23 20:14:41 -0500 (Fri, 23 May 2008) | 1 line add document describing API changes from 1.4.0 to 1.6.0 ........ 2008-05-23 21:37 +0000 [r118168] Brett Bryant * main/manager.c, /, main/http.c, include/asterisk/manager.h: Merged revisions 118161 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r118161 | bbryant | 2008-05-23 16:19:42 -0500 (Fri, 23 May 2008) | 3 lines Add new functionality to http server that requires manager authentication for any path that includes a directory named 'private'. This patch also requires manager authentication for any POST's being sent to the server as well to help secure uploads. ........ 2008-05-23 21:31 +0000 [r118165] Jeff Peeler * channels/chan_zap.c: Merged revisions 118164 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r118164 | jpeeler | 2008-05-23 16:26:39 -0500 (Fri, 23 May 2008) | 9 lines Merged revisions 118163 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r118163 | jpeeler | 2008-05-23 16:21:35 -0500 (Fri, 23 May 2008) | 1 line Fix a few things I missed to ensure zt_chan_conf structure is not modified in mkintf ........ ................ 2008-05-23 18:15 +0000 [r118130] Tilghman Lesher * res/res_odbc.c, /: Merged revisions 118129 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r118129 | tilghman | 2008-05-23 13:09:14 -0500 (Fri, 23 May 2008) | 3 lines Protect the object from changing while the 'odbc show' CLI command is running (Closes issue #12704) ........ 2008-05-23 13:00 +0000 [r118054] Tilghman Lesher * doc/cli.txt (added), /: Merged revisions 118053 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r118053 | tilghman | 2008-05-23 08:00:10 -0500 (Fri, 23 May 2008) | 11 lines Merged revisions 118052 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r118052 | tilghman | 2008-05-23 07:59:16 -0500 (Fri, 23 May 2008) | 3 lines Add information on using the Asterisk console, including tab command line completion. (Closes issue #12681) ........ ................ 2008-05-23 12:37 +0000 [r118050] Russell Bryant * include/asterisk/utils.h, /, main/utils.c: Merged revisions 118049 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r118049 | russell | 2008-05-23 07:37:31 -0500 (Fri, 23 May 2008) | 17 lines Merged revisions 118048 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r118048 | russell | 2008-05-23 07:30:53 -0500 (Fri, 23 May 2008) | 9 lines Don't declare a function that takes variable arguments as inline, because it's not valid, and on some compilers, will emit a warning. http://gcc.gnu.org/onlinedocs/gcc/Inline.html#Inline (closes issue #12289) Reported by: francesco_r Patches by Tilghman, final patch by me ........ ................ 2008-05-23 11:02 +0000 [r118021] Philippe Sultan * /, channels/chan_gtalk.c, res/res_jabber.c: Merged revisions 118020 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r118020 | phsultan | 2008-05-23 12:33:21 +0200 (Fri, 23 May 2008) | 15 lines - remove whitespaces between tags in received XML packets before giving them to the parser ; - report Gtalk error messages from a buddy to the console. This patch makes Asterisk "Google Jingle" (chan_gtalk) implementation work with Empathy. Note that this is only true for audio streams, not video. Thank you to PH for his great help! (closes issue #12647) Reported by: PH Patches: trunk-12647-1.diff uploaded by phsultan (license 73) Tested by: phsultan, PH ........ 2008-05-22 21:43 +0000 [r117984-117987] Tilghman Lesher * /, pbx/pbx_realtime.c, configs/pbx_realtime.conf (added): Merged revisions 117986 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r117986 | tilghman | 2008-05-22 16:42:50 -0500 (Thu, 22 May 2008) | 2 lines Add a compatibility option for upgrading realtime extensions ........ 2008-05-22 18:55 +0000 [r117901] Tilghman Lesher * main/asterisk.c, /: Merged revisions 117900 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r117900 | tilghman | 2008-05-22 13:54:41 -0500 (Thu, 22 May 2008) | 10 lines Merged revisions 117899 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r117899 | tilghman | 2008-05-22 13:53:53 -0500 (Thu, 22 May 2008) | 2 lines Also remove preamble from asynchronous events (reported by jsmith on #asterisk-dev) ........ ................ 2008-05-22 15:51 +0000 [r117793] Sean Bright * /, configs/jabber.conf.sample: Merged revisions 117792 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r117792 | seanbright | 2008-05-22 11:49:17 -0400 (Thu, 22 May 2008) | 1 line Minor text fix. roster -> resource. ........ 2008-05-22 13:41 +0000 [r117757] Russell Bryant * main/asterisk.c, /, build_tools/make_buildopts_h: Merged revisions 117756 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r117756 | russell | 2008-05-22 08:40:52 -0500 (Thu, 22 May 2008) | 5 lines Store build-time options as a string in AST_BUILDOPTS in buildopts.h. Also, display this information in the "core show settings" CLI command. This is useful if you want to verify that you're running a build with DONT_OPTIMIZE, DEBUG_THREADS, etc. ........ 2008-05-21 22:01 +0000 [r117659-117660] Jeff Peeler * channels/chan_zap.c: Merged revisions 117658 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r117658 | jpeeler | 2008-05-21 16:31:17 -0500 (Wed, 21 May 2008) | 10 lines Merged revisions 117582 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r117582 | jpeeler | 2008-05-21 15:11:14 -0500 (Wed, 21 May 2008) | 2 lines Ensure that passed in zt_chan_conf structure is not modified in mkintf. ........ ................ * channels/chan_zap.c, /: Merged revisions 117628 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r117628 | jpeeler | 2008-05-21 15:44:04 -0500 (Wed, 21 May 2008) | 12 lines Merged revisions 117462 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r117462 | jpeeler | 2008-05-21 11:58:40 -0500 (Wed, 21 May 2008) | 3 lines Pass a pointer for the conf parameter to the function mkintf rather than the whole zt_chan_conf structure. Another commit is following to make sure the zt_chan_conf structure is not modified. ........ ................ 2008-05-21 19:45 +0000 [r117576] Joshua Colp * /, channels/chan_sip.c: Merged revisions 117575 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r117575 | file | 2008-05-21 16:39:42 -0300 (Wed, 21 May 2008) | 10 lines Merged revisions 117574 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r117574 | file | 2008-05-21 16:38:28 -0300 (Wed, 21 May 2008) | 2 lines Apply the autoframing setting to dialogs that do not get matched against a user or peer. ........ ................ 2008-05-21 18:44 +0000 [r117522] Tilghman Lesher * main/asterisk.c, /: Merged revisions 117520 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r117520 | tilghman | 2008-05-21 13:43:26 -0500 (Wed, 21 May 2008) | 11 lines Merged revisions 117519 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r117519 | tilghman | 2008-05-21 13:40:14 -0500 (Wed, 21 May 2008) | 3 lines Strip the preamble from the output also when -rx is not being used (Related to issue #12702) ........ ................ 2008-05-21 18:29 +0000 [r117486-117516] Russell Bryant * main/asterisk.c, /: Merged revisions 117515 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r117515 | russell | 2008-05-21 13:29:05 -0500 (Wed, 21 May 2008) | 12 lines Merged revisions 117514 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r117514 | russell | 2008-05-21 13:28:46 -0500 (Wed, 21 May 2008) | 4 lines Don't filter the magic character in the network verboser. It gets filtered once it reaches the client. (related to issue #12702, pointed out by tilghman) ........ ................ * main/asterisk.c, pbx/pbx_gtkconsole.c, /: Merged revisions 117508 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r117508 | russell | 2008-05-21 13:20:11 -0500 (Wed, 21 May 2008) | 15 lines Merged revisions 117507 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r117507 | russell | 2008-05-21 13:19:34 -0500 (Wed, 21 May 2008) | 7 lines 1) Don't print the verbose marker in front of every message from ast_verbose() being sent to remote consoles. 2) Fix pbx_gtkconsole to filter out the verbose marker. (related to issue #12702) ........ ................ * main/asterisk.c, /: Merged revisions 117481 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r117481 | russell | 2008-05-21 13:12:19 -0500 (Wed, 21 May 2008) | 14 lines Merged revisions 117479 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r117479 | russell | 2008-05-21 13:11:51 -0500 (Wed, 21 May 2008) | 6 lines Don't display the verbose marker for calls to ast_verbose() that do not include a VERBOSE_PREFIX in front of the message. (closes issue #12702) Reported by: johnlange Patched by me ........ ................ 2008-05-21 02:21 +0000 [r117368] Mark Michelson * main/config.c, /: Merged revisions 117367 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r117367 | mmichelson | 2008-05-20 21:20:31 -0500 (Tue, 20 May 2008) | 19 lines Be sure that we cache included files for each source file which loads a configuration file. As it was, only the first did so. This led to a problem if the included file was changed (but not the configuration file which includes it) and the second source file attempted to reload the configuration. It would not see that the included file had changed. In this particular example, res_phoneprov and chan_sip both loaded sip.conf, which included a file call sip.peers.conf. Since res_phoneprov was the first to load sip.conf, only it cached the fact that sip.conf included sip.peers.conf. If sip.peers.conf were changed and sip.conf were not and a sip reload were issued (meaning that chan_sip attempts to reload sip.conf only if it and its included files have changed) the changes made to sip.peers.conf would not be seen and therefore no action would be taken. (closes issue #12693) Reported by: marsosa ........ 2008-05-21 01:20 +0000 [r117365] Steve Murphy * /, utils/ael_main.c: Merged revisions 117335 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r117335 | murf | 2008-05-20 19:00:28 -0600 (Tue, 20 May 2008) | 10 lines These changes were made via the comments atis_work made at 4:30am (Mountain Time zone- US) in #asterisk-dev on 20 May 2008. He noted that a backslash was being inserted before commas in app call arguments in the extensions.conf.aeldump file that you get from aelparse with the -w arg. This was being generated from code left over from 1.4, where commas were substituted with '|', and any remaining commas needed to be escaped. Many thanks to atis for his comment; please let us know if these changes break anything! ........ 2008-05-19 16:58 +0000 [r117134-117137] Joshua Colp * res/res_smdi.c, /: Merged revisions 117136 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r117136 | file | 2008-05-19 13:53:33 -0300 (Mon, 19 May 2008) | 14 lines Merged revisions 117135 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r117135 | file | 2008-05-19 13:50:52 -0300 (Mon, 19 May 2008) | 6 lines Use the right pthread lock and condition when waiting. (closes issue #12664) Reported by: tomo1657 Patches: res_smdi.c.patch uploaded by tomo1657 (license 484) ........ ................ 2008-05-19 16:07 +0000 [r117089] Tilghman Lesher * include/asterisk/utils.h, /: Merged revisions 117088 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r117088 | tilghman | 2008-05-19 11:07:09 -0500 (Mon, 19 May 2008) | 10 lines Merged revisions 117086 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r117086 | tilghman | 2008-05-19 11:05:05 -0500 (Mon, 19 May 2008) | 2 lines The addition of usleep(2) within ast_assert requires the inclusion of the unistd.h header ........ ................ 2008-05-19 16:05 +0000 [r117083-117087] Joshua Colp * /, main/logger.c: Merged revisions 117085 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r117085 | file | 2008-05-19 13:03:33 -0300 (Mon, 19 May 2008) | 4 lines The logger closes the files it is logging to when reloading so we have to read in the logger configuration even if it has not changed so that the logs get opened again. (closes issue #12665) Reported by: DennisD ........ * /, channels/h323/ast_h323.cxx: Merged revisions 117082 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r117082 | file | 2008-05-19 12:24:44 -0300 (Mon, 19 May 2008) | 14 lines Merged revisions 117081 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r117081 | file | 2008-05-19 12:22:10 -0300 (Mon, 19 May 2008) | 6 lines Make chan_h323 work with pwlib 1.12.0 (closes issue #12682) Reported by: bamby Patches: pwlib_nopipe.diff uploaded by bamby (license 430) ........ ................ 2008-05-19 03:44 +0000 [r116980] Russell Bryant * /, channels/chan_iax2.c: Merged revisions 116979 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r116979 | russell | 2008-05-18 22:44:28 -0500 (Sun, 18 May 2008) | 12 lines Merged revisions 116978 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r116978 | russell | 2008-05-18 22:44:04 -0500 (Sun, 18 May 2008) | 4 lines Avoid access of uninitialized memory. This caused a bunch of crashes for me while doing load testing of development branch where I'm working on some performance improvements. ........ ................ 2008-05-18 21:18 +0000 [r116949] Tilghman Lesher * /, utils/astcanary.c: Merged revisions 116948 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r116948 | tilghman | 2008-05-18 16:15:58 -0500 (Sun, 18 May 2008) | 4 lines Add a set of text to the file astcanary uses to communicate back the main Asterisk process, which explains the purpose for the file being there. This should assist people who find the file and wonder why it exists. ........ 2008-05-18 19:59 +0000 [r116922] Russell Bryant * /, channels/chan_sip.c: Merged revisions 116919 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r116919 | russell | 2008-05-18 14:58:10 -0500 (Sun, 18 May 2008) | 3 lines Remove duplicate colon on Reason header (closes issue #12678) ........ 2008-05-17 19:40 +0000 [r116849-116885] Joshua Colp * /, channels/chan_skinny.c: Merged revisions 116800 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r116800 | file | 2008-05-16 17:30:24 -0300 (Fri, 16 May 2008) | 12 lines Merged revisions 116799 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r116799 | file | 2008-05-16 17:28:11 -0300 (Fri, 16 May 2008) | 4 lines Check to make sure an RTP structure exists before calling ast_rtp_new_source on it. (closes issue #12669) Reported by: sbisker ........ ................ 2008-05-16 20:03 +0000 [r116798] Matthew Fredrickson * channels/chan_zap.c, /: Merged revisions 116797 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r116797 | mattf | 2008-05-16 15:00:04 -0500 (Fri, 16 May 2008) | 1 line Try to see if we can make our ringback situation a little better ........ 2008-05-15 22:07 +0000 [r116636-116695] Tilghman Lesher * include/asterisk/utils.h, /, include/asterisk/strings.h: Merged revisions 116694 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r116694 | tilghman | 2008-05-15 17:05:47 -0500 (Thu, 15 May 2008) | 4 lines Add an extra check in ast_strlen_zero, and make ast_assert() not print the file, line, and function name twice. (Closes issue #12650) ........ * cdr/cdr_csv.c, /: Merged revisions 116631 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r116631 | tilghman | 2008-05-15 12:58:22 -0500 (Thu, 15 May 2008) | 3 lines Don't unload config on reload, when config has not changed. (Closes issue #12652) ........ 2008-05-14 21:41 +0000 [r116470] Russell Bryant * main/rtp.c, main/sched.c, main/channel.c, main/udptl.c, include/asterisk/utils.h, /, channels/chan_agent.c, main/abstract_jb.c, include/asterisk/channel.h: Merged revisions 116469 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r116469 | russell | 2008-05-14 16:40:43 -0500 (Wed, 14 May 2008) | 12 lines Merged revisions 116463 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r116463 | russell | 2008-05-14 16:32:00 -0500 (Wed, 14 May 2008) | 4 lines Add ast_assert(), which can be used to handle fatal errors. It is only compiled in if dev-mode is enabled, and only aborts if DO_CRASH is defined. (inspired by issue #12650) ........ ................ 2008-05-14 21:39 +0000 [r116468] Tilghman Lesher * /, res/res_agi.c: Merged revisions 116467 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r116467 | tilghman | 2008-05-14 16:39:06 -0500 (Wed, 14 May 2008) | 15 lines Merged revisions 116466 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r116466 | tilghman | 2008-05-14 16:38:09 -0500 (Wed, 14 May 2008) | 7 lines Avoid zombies when the channel exits before the AGI. (closes issue #12648) Reported by: gkloepfer Patches: 20080514__bug12648.diff.txt uploaded by Corydon76 (license 14) Tested by: gkloepfer ........ ................ 2008-05-14 20:43 +0000 [r116408-116411] Jason Parker * /, configs/voicemail.conf.sample: Merged revisions 116410 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r116410 | qwell | 2008-05-14 15:43:26 -0500 (Wed, 14 May 2008) | 9 lines Merged revisions 116409 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r116409 | qwell | 2008-05-14 15:43:08 -0500 (Wed, 14 May 2008) | 1 line Document exitcontext in app_voicemail sample config ........ ................ * apps/app_voicemail.c, /: Merged revisions 116407 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r116407 | qwell | 2008-05-14 15:36:55 -0500 (Wed, 14 May 2008) | 9 lines Voicemail "* exit" should not require an exitcontext to be specified. The behavior in 1.4 was that it would use the current context if an exitcontext existed. (closes issue #12605) Reported by: kenjreno Patches: 12605-starexit.diff uploaded by qwell (license 4) Tested by: file ........ 2008-05-14 18:54 +0000 [r116351-116354] Joshua Colp * /, main/Makefile: Merged revisions 116353 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r116353 | file | 2008-05-14 15:54:16 -0300 (Wed, 14 May 2008) | 12 lines Merged revisions 116352 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r116352 | file | 2008-05-14 15:53:39 -0300 (Wed, 14 May 2008) | 4 lines Add linux-gnueabi in. (closes issue #12529) Reported by: tzafrir ........ ................ * /, res/res_config_ldap.c: Merged revisions 116350 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r116350 | file | 2008-05-14 15:25:54 -0300 (Wed, 14 May 2008) | 4 lines Make the ldap version setting work without having both version and protocol set. (closes issue #12613) Reported by: suretec ........ 2008-05-14 17:01 +0000 [r116319] Tilghman Lesher * /, apps/app_externalivr.c: Merged revisions 116298 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r116298 | tilghman | 2008-05-14 11:53:23 -0500 (Wed, 14 May 2008) | 15 lines Merged revisions 116296 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r116296 | tilghman | 2008-05-14 11:46:48 -0500 (Wed, 14 May 2008) | 2 lines Detect another way for a connection to have gone away. (closes issue #12618) Reported by: ctooley Patches: 1.4-externalivr-test_fd.diff uploaded by ctooley (license 136) trunk-externalivr-test_fd.diff uploaded by ctooley (license 136) ........ ................ 2008-05-14 Russell Bryant * Asterisk 1.6.0-beta9 released. 2008-05-14 13:13 +0000 [r116236] Olle Johansson * /, channels/chan_sip.c: Merged revisions 116234 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r116234 | oej | 2008-05-14 15:05:15 +0200 (Ons, 14 Maj 2008) | 11 lines Merged revisions 116230 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r116230 | oej | 2008-05-14 14:51:06 +0200 (Ons, 14 Maj 2008) | 3 lines Accept text messages even with Content-Type: text/plain;charset=Södermanländska ........ ................ 2008-05-14 00:20 +0000 [r116096-116139] Mark Michelson * main/channel.c, /, include/asterisk/lock.h: Merged revisions 116089 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r116089 | mmichelson | 2008-05-13 18:54:01 -0500 (Tue, 13 May 2008) | 20 lines Merged revisions 116088 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r116088 | mmichelson | 2008-05-13 18:47:49 -0500 (Tue, 13 May 2008) | 12 lines A change to the way channel locks are handled when DEBUG_CHANNEL_LOCKS is defined. After debugging a deadlock, it was noticed that when DEBUG_CHANNEL_LOCKS is enabled in menuselect, the actual origin of channel locks is obscured by the fact that all channel locks appear to happen in the function ast_channel_lock(). This code change redefines ast_channel_lock to be a macro which maps to __ast_channel_lock(), which then relays the proper file name, line number, and function name information to the core lock functions so that this information will be displayed in the case that there is some sort of locking error or core show locks is issued. ........ ................ 2008-05-13 21:19 +0000 [r116020-116040] Russell Bryant * channels/chan_local.c, /: Merged revisions 116039 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r116039 | russell | 2008-05-13 16:18:55 -0500 (Tue, 13 May 2008) | 32 lines Merged revisions 116038 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r116038 | russell | 2008-05-13 16:17:23 -0500 (Tue, 13 May 2008) | 24 lines Fix a deadlock involving channel autoservice and chan_local that was debugged and fixed by mmichelson and me. We observed a system that had a bunch of threads stuck in ast_autoservice_stop(). The reason these threads were waiting around is because this function waits to ensure that the channel list in the autoservice thread gets rebuilt before the stop() function returns. However, the autoservice thread was also locked, so the autoservice channel list was never getting rebuilt. The autoservice thread was stuck waiting for the channel lock on a local channel. However, the local channel was locked by a thread that was stuck in the autoservice stop function. It turned out that the issue came down to the local_queue_frame() function in chan_local. This function assumed that one of the channels passed in as an argument was locked when called. However, that was not always the case. There were multiple cases in which this channel was not locked when the function was called. We fixed up chan_local to indicate to this function whether this channel was locked or not. The previous assumption had caused local_queue_frame() to improperly return with the channel locked, where it would then never get unlocked. (closes issue #12584) (related to issue #12603) ........ ................ * main/autoservice.c, /: Merged revisions 116001 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r116001 | russell | 2008-05-13 16:07:59 -0500 (Tue, 13 May 2008) | 13 lines Merged revisions 115990 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r115990 | russell | 2008-05-13 16:05:57 -0500 (Tue, 13 May 2008) | 5 lines Fix an issue that I noticed in autoservice while mmichelson and I were debugging a different problem. I noticed that it was theoretically possible for two threads to attempt to start the autoservice thread at the same time. This change makes the process of starting the autoservice thread, thread-safe. ........ ................ 2008-05-13 20:30 +0000 [r115946] Joshua Colp * /, channels/chan_alsa.c: Merged revisions 115945 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r115945 | file | 2008-05-13 17:29:27 -0300 (Tue, 13 May 2008) | 12 lines Merged revisions 115944 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r115944 | file | 2008-05-13 17:28:23 -0300 (Tue, 13 May 2008) | 4 lines Use the right flag to open the audio in non-blocking. (closes issue #12616) Reported by: nicklewisdigiumuser ........ ................ 2008-05-13 20:19 +0000 [r115940-115942] Matthew Fredrickson * channels/chan_zap.c, /: Merged revisions 115941 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r115941 | mattf | 2008-05-13 15:18:04 -0500 (Tue, 13 May 2008) | 1 line Need to clear calling_party_cat variable after we retrieve it ........ * channels/chan_zap.c, /: Merged revisions 115939 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r115939 | mattf | 2008-05-13 15:11:20 -0500 (Tue, 13 May 2008) | 1 line Add support for receiving calling party category ........ 2008-05-13 18:38 +0000 [r115887] Tilghman Lesher * main/asterisk.c, /: Merged revisions 115886 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r115886 | tilghman | 2008-05-13 13:38:11 -0500 (Tue, 13 May 2008) | 11 lines Merged revisions 115884 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r115884 | tilghman | 2008-05-13 13:36:13 -0500 (Tue, 13 May 2008) | 3 lines If the socket dies (read returns 0=EOF), return immediately. (Closes issue #12637) ........ ................ 2008-05-13 17:48 +0000 [r115848-115851] Russell Bryant * res/res_smdi.c, /: Merged revisions 115847 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r115847 | russell | 2008-05-13 12:14:22 -0500 (Tue, 13 May 2008) | 2 lines Initialize the start time in smdi_msg_wait. Somehow this code got lost in trunk. ........ 2008-05-12 17:57 +0000 [r115738] Mark Michelson * main/utils.c: Merged revisions 115737 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r115737 | mmichelson | 2008-05-12 12:55:08 -0500 (Mon, 12 May 2008) | 15 lines Merged revisions 115735 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r115735 | mmichelson | 2008-05-12 12:51:14 -0500 (Mon, 12 May 2008) | 7 lines If a thread holds no locks, do not print any information on the thread when issuing a core show locks command. This will help to de-clutter output somewhat. Russell said it would be fine to place this improvement in the 1.4 branch, so that's why it's going here too. ........ ................ 2008-05-12 16:36 +0000 [r115706] Jason Parker * /, apps/app_queue.c: Merged revisions 115705 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r115705 | qwell | 2008-05-12 11:35:50 -0500 (Mon, 12 May 2008) | 1 line Correctly document state interface for AddQueueMember. Discovered while looking at issue #12626. ........ 2008-05-12 15:18 +0000 [r115672] Brett Bryant * /, channels/chan_iax2.c: Merged revisions 115669 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r115669 | bbryant | 2008-05-12 10:17:32 -0500 (Mon, 12 May 2008) | 3 lines A small change to fix iax2 native bridging. ........ 2008-05-11 03:27 +0000 [r115599-115601] Matthew Fredrickson * channels/chan_zap.c, /, configure, include/asterisk/autoconfig.h.in, configure.ac: Merged revisions 115600 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r115600 | mattf | 2008-05-10 22:23:05 -0500 (Sat, 10 May 2008) | 1 line Add Zap MTP2 support to chan_zap ........ * channels/chan_zap.c, /: Merged revisions 115598 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r115598 | mattf | 2008-05-10 21:19:21 -0500 (Sat, 10 May 2008) | 1 line Open up audio channel when we get ACM on SS7 event ........ 2008-05-10 14:22 +0000 [r115597] Tilghman Lesher * /, cdr/cdr_pgsql.c: Merged revisions 115596 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r115596 | tilghman | 2008-05-10 09:19:41 -0500 (Sat, 10 May 2008) | 2 lines Ensure that "calldate" is acceptable for a column name. ........ 2008-05-09 16:38 +0000 [r115581] Joshua Colp * /, configure, include/asterisk/autoconfig.h.in, configure.ac: Merged revisions 115580 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r115580 | file | 2008-05-09 13:36:58 -0300 (Fri, 09 May 2008) | 10 lines Merged revisions 115579 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r115579 | file | 2008-05-09 13:34:08 -0300 (Fri, 09 May 2008) | 2 lines Improve res_ninit and res_ndestroy autoconf logic on the Darwin platform. ........ ................ 2008-05-08 19:21 +0000 [r115553-115570] Russell Bryant * /, channels/chan_iax2.c: Merged revisions 115569 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r115569 | russell | 2008-05-08 14:20:35 -0500 (Thu, 08 May 2008) | 10 lines Merged revisions 115568 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r115568 | russell | 2008-05-08 14:19:50 -0500 (Thu, 08 May 2008) | 2 lines Remove debug output. ........ ................ * /, channels/chan_iax2.c: Merged revisions 115566 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r115566 | russell | 2008-05-08 14:17:04 -0500 (Thu, 08 May 2008) | 41 lines Merged revisions 115565 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r115565 | russell | 2008-05-08 14:15:25 -0500 (Thu, 08 May 2008) | 33 lines Merged revisions 115564 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r115564 | russell | 2008-05-08 14:14:04 -0500 (Thu, 08 May 2008) | 25 lines Fix a race condition that bbryant just found while doing some IAX2 testing. He was running Asterisk trunk running IAX2 calls through a few Asterisk boxes, however, the audio was extremely choppy. We looked at a packet trace and saw a storm of INVAL and VNAK frames being sent from one box to another. It turned out that what had happened was that one box tried to send a CONTROL frame before the 3 way handshake had completed. So, that frame did not include the destination call number, because it didn't have it yet. Part of our recent work for security issues included an additional check to ensure that frames that are supposed to include the destination call number have the correct one. This caused the frame to be rejected with an INVAL. The frame would get retransmitted for forever, rejected every time ... This race condition exists in all versions that got the security changes, in theory. However, it is really only likely that this would cause a problem in Asterisk trunk. There was a control frame being sent (SRCUPDATE) at the _very_ beginning of the call, which does not exist in 1.2 or 1.4. However, I am fixing all versions that could potentially be affected by the introduced race condition. These changes are what bbryant and I came up with to fix the issue. Instead of simply dropping control frames that get sent before the handshake is complete, the code attempts to wait a little while, since in most cases, the handshake will complete very quickly. If it doesn't complete after yielding for a little while, then the frame gets dropped. ........ ................ ................ * /, channels/chan_sip.c: Merged revisions 115562 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r115562 | russell | 2008-05-08 11:14:08 -0500 (Thu, 08 May 2008) | 11 lines Merged revisions 115561 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r115561 | russell | 2008-05-08 11:11:33 -0500 (Thu, 08 May 2008) | 3 lines Don't give up on attempting an outbound registration if we receive a 408 Timeout. (closes issue #12323) ........ ................ * /, contrib/scripts/postgres_cdr.sql (removed): Merged revisions 115558 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r115558 | russell | 2008-05-08 10:38:27 -0500 (Thu, 08 May 2008) | 11 lines Merged revisions 115557 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r115557 | russell | 2008-05-08 10:37:49 -0500 (Thu, 08 May 2008) | 3 lines remove postgres_cdr.sql, as the CDR schema is in realtime_pgsql.sql, as well (closes issue #9676) ........ ................ * contrib/init.d/rc.debian.asterisk, /: Merged revisions 115555 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r115555 | russell | 2008-05-08 10:32:48 -0500 (Thu, 08 May 2008) | 11 lines Merged revisions 115554 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r115554 | russell | 2008-05-08 10:32:08 -0500 (Thu, 08 May 2008) | 3 lines Don't exit the script if Asterisk is not running. (closes issue #12611) ........ ................ * main/pbx.c, /: Merged revisions 115552 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r115552 | russell | 2008-05-08 10:26:49 -0500 (Thu, 08 May 2008) | 12 lines Merged revisions 115551 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r115551 | russell | 2008-05-08 10:24:54 -0500 (Thu, 08 May 2008) | 4 lines Don't use a channel before checking for channel allocation failure. (closes issue #12609) Reported by: edantie ........ ................ 2008-05-08 15:08 +0000 [r115549] Matthew Fredrickson * channels/chan_zap.c, /: Merged revisions 115548 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r115548 | mattf | 2008-05-08 10:04:45 -0500 (Thu, 08 May 2008) | 1 line Remove unused code as well as demote an error message to a debug message ........ 2008-05-08 14:41 +0000 [r115538-115547] Russell Bryant * contrib/init.d/rc.debian.asterisk, /: Merged revisions 115546 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r115546 | russell | 2008-05-08 09:41:12 -0500 (Thu, 08 May 2008) | 12 lines Merged revisions 115545 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r115545 | russell | 2008-05-08 09:40:53 -0500 (Thu, 08 May 2008) | 4 lines Use the same method for executing Asterisk as the rest of the script. (closes issue #12611) Reported by: b_plessis ........ ................ 2008-05-07 18:35 +0000 [r115514-115524] Russell Bryant * /, res/res_config_ldap.c: Merged revisions 115523 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r115523 | russell | 2008-05-07 13:33:50 -0500 (Wed, 07 May 2008) | 6 lines Only save a password if a username exists. (closes issue #12600) Reported By: suretec Patch by me ........ * /, res/res_config_ldap.c: Merged revisions 115521 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r115521 | russell | 2008-05-07 13:30:12 -0500 (Wed, 07 May 2008) | 7 lines Use the default that the log output claims will be used for the basedn (closes issue #12599) Reported by: suretec Patches: 12599.patch uploaded by juggie (license 24) ........ * /, channels/chan_h323.c: Merged revisions 115519 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r115519 | russell | 2008-05-07 13:24:51 -0500 (Wed, 07 May 2008) | 2 lines Let chan_h323 build in dev mode ........ * /, include/asterisk/dlinkedlists.h (removed), channels/chan_iax2.c: Merged revisions 115513 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r115513 | russell | 2008-05-07 12:28:19 -0500 (Wed, 07 May 2008) | 19 lines Merged revisions 115512 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r115512 | russell | 2008-05-07 11:24:09 -0500 (Wed, 07 May 2008) | 11 lines Merged revisions 115511 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r115511 | russell | 2008-05-07 11:22:49 -0500 (Wed, 07 May 2008) | 3 lines Remove remnants of dlinkedlists. I didn't actually use them in the final version of my IAX2 improvements. ........ ................ ................ 2008-05-07 13:49 +0000 [r115510] Tilghman Lesher * contrib/scripts/asterisk.ldap-schema, contrib/scripts/asterisk.ldif, /: Merged revisions 115509 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r115509 | tilghman | 2008-05-07 08:49:15 -0500 (Wed, 07 May 2008) | 2 lines Update typos in description fields (closes issue #12598) Reported by: suretec Patches: asterisk_schema_changes.patch uploaded by suretec (license 70) ........ 2008-05-06 19:56 +0000 [r115420-115424] Jason Parker * /, contrib/scripts/get_ilbc_source.sh: Merged revisions 115423 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r115423 | qwell | 2008-05-06 14:55:45 -0500 (Tue, 06 May 2008) | 23 lines Merged revisions 115422 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r115422 | qwell | 2008-05-06 14:55:29 -0500 (Tue, 06 May 2008) | 15 lines Merged revisions 115421 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r115421 | qwell | 2008-05-06 14:54:57 -0500 (Tue, 06 May 2008) | 7 lines read requires an argument on some non-bash shells (closes issue #12593) Reported by: bkruse Patches: getilbc.sh_12593_v1.diff uploaded by bkruse (license 132) ........ ................ ................ * /, res/res_musiconhold.c: Merged revisions 115419 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r115419 | qwell | 2008-05-06 14:38:44 -0500 (Tue, 06 May 2008) | 15 lines Merged revisions 115418 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r115418 | qwell | 2008-05-06 14:34:58 -0500 (Tue, 06 May 2008) | 7 lines Switch to using ast_random() rather than just rand(). This does not fix the bug reported, but I believe it is correct. (from issue #12446) Patches: bug_12446.diff uploaded by snuffy (license 35) ........ ................ 2008-05-06 19:33 +0000 [r115417] Tilghman Lesher * main/asterisk.c, /: Merged revisions 115416 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r115416 | tilghman | 2008-05-06 14:32:29 -0500 (Tue, 06 May 2008) | 10 lines Merged revisions 115415 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r115415 | tilghman | 2008-05-06 14:31:39 -0500 (Tue, 06 May 2008) | 2 lines Don't print the terminating NUL. (Closes issue #12589) ........ ................ 2008-05-06 13:57 +0000 [r115343] Joshua Colp * /, configure, configure.ac: Merged revisions 115342 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r115342 | file | 2008-05-06 10:55:44 -0300 (Tue, 06 May 2008) | 10 lines Merged revisions 115341 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r115341 | file | 2008-05-06 10:54:15 -0300 (Tue, 06 May 2008) | 2 lines Add in missing argument. ........ ................ 2008-05-05 23:01 +0000 [r115335] Tilghman Lesher * main/asterisk.c, /, main/logger.c: Merged revisions 115334 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r115334 | tilghman | 2008-05-05 18:00:31 -0500 (Mon, 05 May 2008) | 15 lines Merged revisions 115333 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r115333 | tilghman | 2008-05-05 17:50:31 -0500 (Mon, 05 May 2008) | 7 lines Separate verbose output from CLI output, by using a preamble. (closes issue #12402) Reported by: Corydon76 Patches: 20080410__no_verbose_in_rx_output.diff.txt uploaded by Corydon76 (license 14) 20080501__no_verbose_in_rx_output__1.4.diff.txt uploaded by Corydon76 (license 14) ........ ................ 2008-05-05 22:17 +0000 [r115331] Joshua Colp * /, build_tools/menuselect-deps.in, configure, include/asterisk/autoconfig.h.in, codecs/codec_speex.c, configure.ac: Merged revisions 115328 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r115328 | file | 2008-05-05 19:13:57 -0300 (Mon, 05 May 2008) | 10 lines Merged revisions 115327 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r115327 | file | 2008-05-05 19:10:05 -0300 (Mon, 05 May 2008) | 2 lines Make sure that either the main speex library contains preprocess functions or that speexdsp does. If both fail then speex stuff can not be built. ........ ................ 2008-05-05 22:14 +0000 [r115330] Mark Michelson * main/config.c, /: Merged revisions 115329 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r115329 | mmichelson | 2008-05-05 17:14:06 -0500 (Mon, 05 May 2008) | 15 lines #execing the same file multiple times led to warning messages saying that the same file was being #included twice. This was due to the fact that #exec created a temporary file which was then #included. The name of the temporary file was the name of the #exec'd file, with the Unix timestamp and thread ID concatenated. The issue was that if multiple #exec statements of the same file were reached in the same second, then the result was that the temporary files would have duplicate names. To resolve this, the temporary file now has microsecond resolution for the timestamp portion. (closes issue #12574) Reported by: jmls Patches: 12574.patch uploaded by putnopvut (license 60) Tested by: jmls, putnopvut ........ 2008-05-05 21:44 +0000 [r115322] Mark Michelson * /, apps/app_queue.c: Merged revisions 115321 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r115321 | mmichelson | 2008-05-05 16:43:21 -0500 (Mon, 05 May 2008) | 21 lines Merged revisions 115320 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r115320 | mmichelson | 2008-05-05 16:41:34 -0500 (Mon, 05 May 2008) | 13 lines Don't consider a caller "handled" until the caller is bridged with a queue member. There was too much of an opportunity for the member to hang up (either during a delay, announcement, or overly long agi) between the time that he answered the phone and the time when he actually was bridged with the caller. The consequence of this was that if the member hung up in that interval, then proper abandonment details would not be noted in the queue log if the caller were to hang up at any point after the member hangup. (closes issue #12561) Reported by: ablackthorn ........ ................ 2008-05-05 20:28 +0000 [r115316] Russell Bryant * /, channels/chan_iax2.c: Merged revisions 115315 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r115315 | russell | 2008-05-05 15:28:17 -0500 (Mon, 05 May 2008) | 2 lines Remove my rant, since I have now replaced the rant with code. ........ 2008-05-05 19:58 +0000 [r115310] Tilghman Lesher * include/asterisk/res_odbc.h, /: Merged revisions 115309 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r115309 | tilghman | 2008-05-05 14:57:28 -0500 (Mon, 05 May 2008) | 10 lines Merged revisions 115308 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r115308 | tilghman | 2008-05-05 14:55:55 -0500 (Mon, 05 May 2008) | 2 lines Err, the documentation on the return value of ast_odbc_backslash_is_escape is exactly backwards. ........ ................ 2008-05-05 19:50 +0000 [r115306] Russell Bryant * /, channels/chan_sip.c: Merged revisions 115305 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r115305 | russell | 2008-05-05 14:50:24 -0500 (Mon, 05 May 2008) | 13 lines Merged revisions 115304 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r115304 | russell | 2008-05-05 14:49:25 -0500 (Mon, 05 May 2008) | 5 lines Avoid putting opaque="" in Digest authentication. This patch came from switchvox. It fixes authentication with Primus in Canada, and has been in use for a very long time without causing problems with any other providers. (closes issue AST-36) ........ ................ 2008-05-05 19:43 +0000 [r115303] Tilghman Lesher * /, UPGRADE.txt: Merged revisions 115302 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r115302 | tilghman | 2008-05-05 14:42:36 -0500 (Mon, 05 May 2008) | 2 lines Note change for ExecIf syntax (caught by jmls on IRC) ........ 2008-05-05 10:55 +0000 [r115289] Kevin P. Fleming * /, UPGRADE.txt: Merged revisions 115288 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r115288 | kpfleming | 2008-05-05 05:55:09 -0500 (Mon, 05 May 2008) | 2 lines clarify wording ........ 2008-05-05 03:26 +0000 [r115287] Tilghman Lesher * contrib/scripts/safe_asterisk, contrib/init.d/rc.suse.asterisk, contrib/init.d/rc.debian.asterisk, contrib/init.d/rc.mandrake.asterisk, /, contrib/init.d/rc.redhat.asterisk, contrib/init.d/rc.gentoo.asterisk, contrib/init.d/rc.slackware.asterisk: Merged revisions 115286 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r115286 | tilghman | 2008-05-04 22:25:35 -0500 (Sun, 04 May 2008) | 15 lines Merged revisions 115285 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r115285 | tilghman | 2008-05-04 22:22:25 -0500 (Sun, 04 May 2008) | 7 lines When starting Asterisk, bug out if Asterisk is already running. (closes issue #12525) Reported by: explidous Patches: 20080428__bug12525.diff.txt uploaded by Corydon76 (license 14) Tested by: mvanbaak ........ ................ 2008-05-04 02:12 +0000 [r115278-115284] Joshua Colp * /, configure, acinclude.m4: Merged revisions 115283 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r115283 | file | 2008-05-03 23:11:01 -0300 (Sat, 03 May 2008) | 10 lines Merged revisions 115282 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r115282 | file | 2008-05-03 23:09:44 -0300 (Sat, 03 May 2008) | 2 lines Expand the test function for GCC attributes so that more complex attributes are properly recognized. ........ ................ * /, include/asterisk/compiler.h: Merged revisions 115280 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r115280 | file | 2008-05-03 22:52:00 -0300 (Sat, 03 May 2008) | 10 lines Merged revisions 115279 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r115279 | file | 2008-05-03 22:50:59 -0300 (Sat, 03 May 2008) | 2 lines For my next trick I will make these work with what our autoconf header file gives us. ........ ................ * /, configure, acinclude.m4: Merged revisions 115277 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r115277 | file | 2008-05-03 22:45:21 -0300 (Sat, 03 May 2008) | 10 lines Merged revisions 115276 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r115276 | file | 2008-05-03 22:43:26 -0300 (Sat, 03 May 2008) | 2 lines Treat warnings as errors when checking if a GCC attribute exists. We have to do this as GCC will just ignore the attribute and pop up a warning, it won't actually fail to compile. ........ ................ 2008-05-03 04:25 +0000 [r115269-115275] Dwayne M. Hubbard * /: block voicemail mwi notification subscriptions taskprocessor * /: block pbx taskprocessor * /: block app_queue taskprocessor * /: blocked taskprocessors 2008-05-02 14:55 +0000 [r115198-115200] Mark Michelson * /, include/asterisk/sched.h: Merged revisions 115197 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r115197 | mmichelson | 2008-05-02 09:28:55 -0500 (Fri, 02 May 2008) | 14 lines Merged revisions 115196 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r115196 | mmichelson | 2008-05-02 09:28:19 -0500 (Fri, 02 May 2008) | 6 lines Clarify a comment that was, well, just wrong. It turns out that ignoring the way that macros expand. Instead, I have clarified in the comment why the macro will work even if the scheduler id for the task to be deleted changes during the execution of the macro. ........ ................ 2008-05-02 02:57 +0000 [r115107-115160] Tilghman Lesher * include/asterisk/res_odbc.h, /: Merged revisions 115104 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r115104 | tilghman | 2008-05-01 18:21:13 -0500 (Thu, 01 May 2008) | 10 lines Merged revisions 115102 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r115102 | tilghman | 2008-05-01 18:20:25 -0500 (Thu, 01 May 2008) | 2 lines Change the comment of deprecated to an actual compiler deprecation ........ ................ 2008-05-01 19:01 +0000 [r115020] Tilghman Lesher * /, main/utils.c: Merged revisions 115018 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r115018 | tilghman | 2008-05-01 14:00:18 -0500 (Thu, 01 May 2008) | 14 lines Merged revisions 115017 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r115017 | tilghman | 2008-05-01 13:59:08 -0500 (Thu, 01 May 2008) | 6 lines '#' is another reserved character for URIs that also needs to be escaped. (closes issue #10543) Reported by: blitzrage Patches: 20080418__bug10543.diff.txt uploaded by Corydon76 (license 14) ........ ................ 2008-05-01 17:28 +0000 [r114932] Russell Bryant * /, UPGRADE.txt: Merged revisions 114931 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r114931 | russell | 2008-05-01 12:28:25 -0500 (Thu, 01 May 2008) | 4 lines Clarify the deprecation notice about Macro() to note that it will not be removed for the sake of backwards compatibility, since it is a non-trivial task to convert existing large dialplans that depend on Macro() to use GoSub(), instead. ........ 2008-05-01 16:52 +0000 [r114923] Jason Parker * channels/chan_zap.c, /: Merged revisions 114922 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r114922 | qwell | 2008-05-01 11:49:24 -0500 (Thu, 01 May 2008) | 10 lines Allow dringXrange to properly default to 10, as was done in 1.4. dringXrange is a new feature that was added, and it attempted to default, but only when the option was specified. (closes issue #12536) Reported by: bjm Patches: 12536-dringXrange.diff uploaded by qwell (license 4) Tested by: bjm ........ 2008-04-30 20:20 +0000 [r114909] Russell Bryant * include/asterisk/dlinkedlists.h (added): Add the dlinkedlists implementation from trunk 2008-04-30 20:17 +0000 [r114907-114908] Mark Michelson * channels/chan_sip.c: Make 1.6.0 compile 2008-04-30 17:06 +0000 [r114900] Olle Johansson * /, channels/chan_sip.c: Merged revisions 114899 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r114899 | oej | 2008-04-30 18:55:49 +0200 (Ons, 30 Apr 2008) | 15 lines Merged revisions 114890 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r114890 | oej | 2008-04-30 18:23:17 +0200 (Ons, 30 Apr 2008) | 7 lines Don't crash on bad SIP replys. Fix created in Huntsville together with Mark M (putnopvut) (closes issue #12363) Reported by: jvandal Tested by: putnopvut, oej ........ ................ 2008-04-30 16:41 +0000 [r114893] Russell Bryant * /, channels/chan_console.c, channels/chan_iax2.c: Merged revisions 114892 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r114892 | russell | 2008-04-30 11:34:24 -0500 (Wed, 30 Apr 2008) | 36 lines Merged revisions 114891 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r114891 | russell | 2008-04-30 11:30:01 -0500 (Wed, 30 Apr 2008) | 28 lines Merge changes from team/russell/iax2_find_callno and iax2_find_callno_1.4 These changes address a critical performance issue introduced in the latest release. The fix for the latest security issue included a change that made Asterisk randomly choose call numbers to make them more difficult to guess by attackers. However, due to some inefficient (this is by far, an understatement) code, when Asterisk chose high call numbers, chan_iax2 became unusable after just a small number of calls. On a small embedded platform, it would not be able to handle a single call. On my Intel Core 2 Duo @ 2.33 GHz, I couldn't run more than about 16 IAX2 channels. Ouch. These changes address some performance issues of the find_callno() function that have bothered me for a very long time. On every incoming media frame, it iterated through every possible call number trying to find a matching active call. This involved a mutex lock and unlock for each call number checked. So, if the random call number chosen was 20000, then every media frame would cause 20000 locks and unlocks. Previously, this problem was not as obvious since Asterisk always chose the lowest call number it could. A second container for IAX2 pvt structs has been added. It is an astobj2 hash table. When we know the remote side's call number, the pvt goes into the hash table with a hash value of the remote side's call number. Then, lookups for incoming media frames are a very fast hash lookup instead of an absolutely insane array traversal. In a quick test, I was able to get more than 3600% more IAX2 channels on my machine with these changes. ........ ................ 2008-04-30 16:15 +0000 [r114889] Jeff Peeler * /, channels/chan_console.c: Merged revisions 114888 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r114888 | jpeeler | 2008-04-30 11:14:43 -0500 (Wed, 30 Apr 2008) | 3 lines Fixes a bug where if a stream monitor thread was not created (caused from failure of opening or starting the stream) pthread_cancel was called with an invalid thread ID. ........ 2008-04-30 14:55 +0000 [r114877-114886] Kevin P. Fleming * /, channels/iax2.h, channels/chan_iax2.c: Merged revisions 114884 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r114884 | kpfleming | 2008-04-30 09:49:51 -0500 (Wed, 30 Apr 2008) | 10 lines Merged revisions 114880 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r114880 | kpfleming | 2008-04-30 09:46:57 -0500 (Wed, 30 Apr 2008) | 2 lines use the ARRAY_LEN macro for indexing through the iaxs/iaxsl arrays so that the size of the arrays can be adjusted in one place, and change the size of the arrays from 32768 calls to 2048 calls when LOW_MEMORY is defined ........ ................ * /, Makefile.rules: Merged revisions 114876 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r114876 | kpfleming | 2008-04-30 07:15:43 -0500 (Wed, 30 Apr 2008) | 10 lines Merged revisions 114875 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r114875 | kpfleming | 2008-04-30 07:14:07 -0500 (Wed, 30 Apr 2008) | 2 lines pay attention to *all* header files for dependency tracking, not just the local ones (inspired by r578 of asterisk-addons by tilghman) ........ ................ 2008-04-29 22:55 +0000 [r114867] Jeff Peeler * /, channels/iax2-provision.c: Merged revisions 114866 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r114866 | jpeeler | 2008-04-29 17:54:14 -0500 (Tue, 29 Apr 2008) | 2 lines Fixes a problem where all the templates were marked as dead no matter what. The templates should only be marked as dead if a configuration file has been successfully loaded and has changes. Bug found while making API documentation for 1.6.0. ........ 2008-04-29 21:09 +0000 [r114850-114858] Mark Michelson * /, apps/app_queue.c: Merged revisions 114849 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r114849 | mmichelson | 2008-04-29 14:42:04 -0500 (Tue, 29 Apr 2008) | 22 lines Merged revisions 114848 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r114848 | mmichelson | 2008-04-29 14:40:06 -0500 (Tue, 29 Apr 2008) | 14 lines Use the MACRO_CONTEXT and MACRO_EXTEN channel variables instead of the channel's macrocontext and macroexten fields. This is needed because if macros are daisy-chained, the incorrect context and extension are placed on the new channel. I also added locking to the channel prior to accessing these variables as noted in trunk's janitor project file. (closes issue #12549) Reported by: darren1713 Patches: app_queue.c.macroextenpatch uploaded by darren1713 (license 116) (with modifications from me) Tested by: putnopvut ........ ................ 2008-04-29 19:04 +0000 [r114846] Kevin P. Fleming * /: bug is not present in this branch 2008-04-29 17:11 +0000 [r114831] Jason Parker * res/res_config_pgsql.c, /: Merged revisions 114830 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r114830 | qwell | 2008-04-29 12:10:55 -0500 (Tue, 29 Apr 2008) | 9 lines Merged revisions 114829 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r114829 | qwell | 2008-04-29 12:08:55 -0500 (Tue, 29 Apr 2008) | 1 line Change warning message to debug, since there are cases where 0 results is perfectly fine. ........ ................ 2008-04-29 12:55 +0000 [r114825] Kevin P. Fleming * /, contrib/scripts/get_ilbc_source.sh: Merged revisions 114824 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r114824 | kpfleming | 2008-04-29 07:54:31 -0500 (Tue, 29 Apr 2008) | 18 lines Merged revisions 114823 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r114823 | kpfleming | 2008-04-29 07:53:12 -0500 (Tue, 29 Apr 2008) | 10 lines Merged revisions 114822 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r114822 | kpfleming | 2008-04-29 07:52:32 -0500 (Tue, 29 Apr 2008) | 2 lines stop script from appending source code if run multiple times ........ ................ ................ 2008-04-28 17:04 +0000 [r114777] Matthew Fredrickson * channels/chan_zap.c, /: Merged revisions 114776 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r114776 | mattf | 2008-04-28 12:00:38 -0500 (Mon, 28 Apr 2008) | 1 line Fix deadlock issue in chan_zap with libss7 due to channel variables being set with the channel pvt lock being held. #12512 ........ 2008-04-28 13:44 +0000 [r114714] Joshua Colp * /, configure, configure.ac: Merged revisions 114713 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r114713 | file | 2008-04-28 10:42:13 -0300 (Mon, 28 Apr 2008) | 2 lines Update autoconf logic with latest API change for libss7. ........ 2008-04-28 04:54 +0000 [r114707-114710] Tilghman Lesher * channels/chan_jingle.c, /, channels/chan_gtalk.c: Merged revisions 114709 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r114709 | tilghman | 2008-04-27 23:53:20 -0500 (Sun, 27 Apr 2008) | 13 lines Merged revisions 114708 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r114708 | tilghman | 2008-04-27 23:47:39 -0500 (Sun, 27 Apr 2008) | 5 lines When modules are embedded, they take on a different name, without the ".so" extension. Specifically check for this name, when we're checking if a module is loaded. (Closes issue #12534) ........ ................ 2008-04-27 15:20 +0000 [r114701] Michiel van Baak * /, channels/chan_skinny.c: Merged revisions 114700 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk Merged to 1.6 because it fixes a crash. ........ r114700 | mvanbaak | 2008-04-27 17:17:18 +0200 (Sun, 27 Apr 2008) | 8 lines Make MWI in chan_skinny event based modeled after chan_zap and chan_mgcp. (closes issue #12214) Reported by: DEA Patches: chan_skinny-vm-events-v3.txt uploaded by DEA (license 3) Tested by: DEA and me ........ 2008-04-27 01:30 +0000 [r114697] Sean Bright * /, configure, configure.ac: Merged revisions 114696 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r114696 | seanbright | 2008-04-26 21:28:32 -0400 (Sat, 26 Apr 2008) | 13 lines Merged revisions 114695 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r114695 | seanbright | 2008-04-26 21:26:15 -0400 (Sat, 26 Apr 2008) | 5 lines When we don't explicitly pass a path to the --with-tds configure option, we may end up finding tds.h in /usr/local/include instead of /usr/include. If this happens, the grep that looks for the version (from tdsver.h) will fail and we'll have some problems during the build. ........ ................ 2008-04-26 15:09 +0000 [r114684-114693] Tilghman Lesher * /, contrib/scripts/vmail.cgi: Merged revisions 114690 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r114690 | tilghman | 2008-04-26 08:17:19 -0500 (Sat, 26 Apr 2008) | 14 lines Merged revisions 114689 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r114689 | tilghman | 2008-04-26 08:15:21 -0500 (Sat, 26 Apr 2008) | 6 lines Clicking forward without selecting a message leaves an errant .lock file. (closes issue #12528) Reported by: pukepail Patches: patch.diff uploaded by pukepail (license 431) ........ ................ 2008-04-25 22:05 +0000 [r114671-114677] Russell Bryant * /, pbx/pbx_lua.c: Merged revisions 114676 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r114676 | russell | 2008-04-25 17:04:46 -0500 (Fri, 25 Apr 2008) | 7 lines Lock the channel around datastore access (closes issue #12527) Reported by: mnicholson Patches: pbx_lua4.diff uploaded by mnicholson (license 96) ........ * /, channels/chan_iax2.c: Merged revisions 114674 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r114674 | russell | 2008-04-25 17:00:35 -0500 (Fri, 25 Apr 2008) | 11 lines Merged revisions 114673 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r114673 | russell | 2008-04-25 16:54:40 -0500 (Fri, 25 Apr 2008) | 3 lines Use consistent logic for checking to see if a call number has been chosen yet. Also, remove some redundant logic I recently added in a fix. ........ ................ 2008-04-25 19:34 +0000 [r114664] Mark Michelson * apps/app_chanspy.c, /: Merged revisions 114663 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r114663 | mmichelson | 2008-04-25 14:33:27 -0500 (Fri, 25 Apr 2008) | 12 lines Merged revisions 114662 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r114662 | mmichelson | 2008-04-25 14:32:02 -0500 (Fri, 25 Apr 2008) | 4 lines Move the unlock of the spyee channel to outside the start_spying() function so that the channel is not unlocked twice when using whisper mode. ........ ................ 2008-04-25 16:26 +0000 [r114652] Mark Michelson * apps/app_voicemail.c, /: Merged revisions 114651 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r114651 | mmichelson | 2008-04-25 11:25:17 -0500 (Fri, 25 Apr 2008) | 4 lines Fix a memory leak and protect against potential dereferences of a NULL pointer. ........ 2008-04-24 22:14 +0000 [r114636] Joshua Colp * /, channels/chan_sip.c: Merged revisions 114635 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r114635 | file | 2008-04-24 19:11:46 -0300 (Thu, 24 Apr 2008) | 4 lines Hey look, it builds. (closes issue #12519) Reported by: falves11 ........ 2008-04-24 21:36 +0000 [r114626-114634] Mark Michelson * /, channels/chan_sip.c: Merged revisions 114633 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r114633 | mmichelson | 2008-04-24 16:35:39 -0500 (Thu, 24 Apr 2008) | 19 lines Merged revisions 114632 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r114632 | mmichelson | 2008-04-24 16:35:08 -0500 (Thu, 24 Apr 2008) | 11 lines Re-invite RTP during a masquerade so that, for instance, an AMI redirect of two channels which are natively bridged will preserve audio on both channels. This prevents a problem with Asterisk not re-inviting due to one of the channels having being a zombie. (closes issue #12513) Reported by: mneuhauser Patches: asterisk-1.4-114602_restore-RTP-on-fixup.patch uploaded by mneuhauser (license 425) ........ ................ * /, apps/app_queue.c: Merged revisions 114629 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r114629 | mmichelson | 2008-04-24 15:43:52 -0500 (Thu, 24 Apr 2008) | 16 lines Merged revisions 114628 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r114628 | mmichelson | 2008-04-24 15:43:03 -0500 (Thu, 24 Apr 2008) | 8 lines Output of channel variables when eventwhencalled=vars was set was being truncated two characters. This patch corrects the problem. (closes issue #12493) Reported by: davidw ........ ................ * channels/chan_local.c, /: Merged revisions 114625 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r114625 | mmichelson | 2008-04-24 15:06:06 -0500 (Thu, 24 Apr 2008) | 18 lines Merged revisions 114624 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r114624 | mmichelson | 2008-04-24 15:04:24 -0500 (Thu, 24 Apr 2008) | 10 lines Resolve a deadlock in chan_local by releasing the channel lock temporarily. (closes issue #11712) Reported by: callguy Patches: 11712.patch uploaded by putnopvut (license 60) Tested by: acunningham ........ ................ 2008-04-24 19:55 +0000 [r114619-114623] Tilghman Lesher * channels/chan_local.c, /: Merged revisions 114622 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r114622 | tilghman | 2008-04-24 14:54:57 -0500 (Thu, 24 Apr 2008) | 12 lines Merged revisions 114621 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r114621 | tilghman | 2008-04-24 14:53:36 -0500 (Thu, 24 Apr 2008) | 4 lines Ensure that when we set the accountcode, it actually shows up in the CDR. (Fix for AMI Originate) (Closes issue #12007) ........ ................ * /, apps/app_meetme.c: Merged revisions 114617 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r114617 | tilghman | 2008-04-24 14:24:31 -0500 (Thu, 24 Apr 2008) | 6 lines Fix DST calculation, and fix bug in calculation of whether conf has started yet or not (Closes issue #12292) Reported by: DEA Patches: app_meetme-rt-dst-sched-fix.txt uploaded by DEA (license 3) ........ 2008-04-24 16:48 +0000 [r114613] Jason Parker * channels/chan_misdn.c, /: Merged revisions 114612 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r114612 | qwell | 2008-04-24 11:47:01 -0500 (Thu, 24 Apr 2008) | 17 lines Merged revisions 51989 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 (closes issue #12496) Reported by: daniele Patches: misdn-moh-1.6.0-beta7.1.patch uploaded by daniele (license 471) Tested by: daniele Technically, I didn't use the patch above except to find out what revision to merge - but it's the same thing as this revision. ........ r51989 | crichter | 2007-01-24 06:57:22 -0600 (Wed, 24 Jan 2007) | 1 line added fix from #8899 ........ ................ 2008-04-24 15:57 +0000 [r114610] Russell Bryant * /, channels/chan_iax2.c: Merged revisions 114609 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r114609 | russell | 2008-04-24 10:56:55 -0500 (Thu, 24 Apr 2008) | 12 lines Merged revisions 114608 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r114608 | russell | 2008-04-24 10:55:21 -0500 (Thu, 24 Apr 2008) | 4 lines Fix a silly mistake in a change I made yesterday that caused chan_iax2 to blow up very quickly. (issue #12515) ........ ................ 2008-04-24 15:00 +0000 [r114607] Olle Johansson * channels/chan_sip.c: Merged revisions 114606 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r114606 | oej | 2008-04-24 16:59:05 +0200 (Tor, 24 Apr 2008) | 11 lines Merged revisions 114603 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r114603 | oej | 2008-04-24 16:55:18 +0200 (Tor, 24 Apr 2008) | 3 lines Only have one max-forwards header in outbound REFERs. Discovered in the Asterisk SIP Masterclass in Orlando. Thanks Joe! ........ ................ 2008-04-24 14:56 +0000 [r114599-114605] Russell Bryant * /, channels/chan_sip.c: Merged revisions 114604 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r114604 | russell | 2008-04-24 09:55:21 -0500 (Thu, 24 Apr 2008) | 3 lines Change a verbose message to debug. (closes issue #12514) ........ * /, main/http.c: Merged revisions 114601 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r114601 | russell | 2008-04-23 17:53:20 -0500 (Wed, 23 Apr 2008) | 14 lines Merged revisions 114600 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r114600 | russell | 2008-04-23 17:18:12 -0500 (Wed, 23 Apr 2008) | 6 lines Improve some broken cookie parsing code. Previously, manager login over HTTP would only work if the mansession_id cookie was first. Now, the code builds a list of all of the cookies in the Cookie header. This fixes a problem observed by users of the Asterisk GUI. (closes AST-20) ........ ................ * apps/app_chanspy.c, /: Merged revisions 114598 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r114598 | russell | 2008-04-23 15:53:05 -0500 (Wed, 23 Apr 2008) | 18 lines Merged revisions 114597 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r114597 | russell | 2008-04-23 15:49:18 -0500 (Wed, 23 Apr 2008) | 10 lines Fix an issue that caused getting the correct next channel to not always work. Also, remove setting the amount of time to wait for a digit from 5 seconds back down to 1/10 of a second. I believe this was so the beep didn't get played over and over really fast, but a while back I put in another fix for that issue. (closes issue #12498) Reported by: jsmith Patches: app_chanspy_channel_walk.trunk.patch uploaded by jsmith (license 15) ........ ................ 2008-04-23 18:34 +0000 [r114596] Jason Parker * /, res/res_musiconhold.c: Merged revisions 114595 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r114595 | qwell | 2008-04-23 13:33:28 -0500 (Wed, 23 Apr 2008) | 16 lines Merged revisions 114594 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r114594 | qwell | 2008-04-23 13:28:44 -0500 (Wed, 23 Apr 2008) | 8 lines Fix reload/unload for res_musiconhold module. (closes issue #11575) Reported by: sunder Patches: M11575_14_rev3.diff uploaded by junky (license 177) bug11575_trunk.diff.txt uploaded by jamesgolovich (license 176) ........ ................ 2008-04-23 18:01 +0000 [r114589-114593] Russell Bryant * main/manager.c, /, include/asterisk/manager.h: Merged revisions 114592 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r114592 | russell | 2008-04-23 13:01:00 -0500 (Wed, 23 Apr 2008) | 13 lines Merged revisions 114591 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r114591 | russell | 2008-04-23 12:55:31 -0500 (Wed, 23 Apr 2008) | 5 lines Store the manager session ID explicitly as 4 byte ID instead of a ulong. The mansession_id cookie is coded to be limited to 8 characters of hex, and this could break logins from 64-bit machines in some cases. (inspired by AST-20) ........ ................ * /, channels/chan_iax2.c: Merged revisions 114588 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r114588 | russell | 2008-04-23 12:18:29 -0500 (Wed, 23 Apr 2008) | 10 lines Merged revisions 114587 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r114587 | russell | 2008-04-23 12:16:32 -0500 (Wed, 23 Apr 2008) | 2 lines Fix find_callno_locked() to actually return the callno locked in some more cases. ........ ................ 2008-04-23 16:57 +0000 [r114586] Olle Johansson * channels/chan_sip.c: Merged revisions 114585 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r114585 | oej | 2008-04-23 18:53:34 +0200 (Ons, 23 Apr 2008) | 10 lines Merged revisions 114584 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r114584 | oej | 2008-04-23 18:51:41 +0200 (Ons, 23 Apr 2008) | 2 lines Add 502 support for both directions, not only one... (see r114571) ........ ................ 2008-04-23 14:56 +0000 [r114581] Joshua Colp * main/pbx.c, /: Merged revisions 114580 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r114580 | file | 2008-04-23 11:55:03 -0300 (Wed, 23 Apr 2008) | 12 lines Merged revisions 114579 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r114579 | file | 2008-04-23 11:54:11 -0300 (Wed, 23 Apr 2008) | 4 lines Instead of stopping dialplan execution when SayNumber attempts to say a large number that it can not print out a message informing the user and continue on. (closes issue #12502) Reported by: bcnit ........ ................ 2008-04-23 01:00 +0000 [r114576-114578] Mark Michelson * apps/app_voicemail.c, /: Merged revisions 114575 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r114575 | mmichelson | 2008-04-22 19:40:30 -0500 (Tue, 22 Apr 2008) | 10 lines Round 1 of IMAP_STORAGE-related app_voicemail changes This makes IMAP_STORAGE include the proper headers if you have specified the "system" option for --with-imap when running the configure script and your IMAP-related headers exist in /usr/include/c-client. This change is due to a hasty merge of a 1.4 change I made. ........ 2008-04-22 23:59 +0000 [r114573] Tilghman Lesher * /, channels/chan_sip.c: Merged revisions 114572 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r114572 | tilghman | 2008-04-22 18:58:19 -0500 (Tue, 22 Apr 2008) | 10 lines Merged revisions 114571 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r114571 | tilghman | 2008-04-22 18:51:44 -0500 (Tue, 22 Apr 2008) | 2 lines Treat a 502 just like a 503, when it comes to processing a response code ........ ................ 2008-04-22 Russell Bryant * Asterisk 1.6.0-beta8 released. 2008-04-22 22:18 +0000 [r114560] Russell Bryant * /, channels/chan_iax2.c: Merged revisions 114559 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r114559 | russell | 2008-04-22 17:17:31 -0500 (Tue, 22 Apr 2008) | 13 lines Merged revisions 114558 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r114558 | russell | 2008-04-22 17:15:36 -0500 (Tue, 22 Apr 2008) | 5 lines When we receive a full frame that is supposed to contain our call number, ensure that it has the correct one. (closes issue #10078) (AST-2008-006) ........ ................ 2008-04-22 22:04 +0000 [r114556] Steve Murphy * main/pbx.c, /: Merged revisions 114553 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r114553 | murf | 2008-04-22 15:57:57 -0600 (Tue, 22 Apr 2008) | 14 lines (closes issue #12469) Reported by: triccyx I had a bit a problem reproducing this in my setup (trying not to disturb my other stuff) but finally, I got it. The problem appears to be that the extension is being added in replace mode, which kinda assumes that the pattern trie has been formed, when in fact, in this case, it was not. The checks being done are not nec. when the tree is not yet formed, as changes like this will be summarized when the trie is formed in the future. I tested the fix, and the crash no longer happens. Feel free to open the bug again if this fix doesn't cure the problem. ........ 2008-04-22 21:16 +0000 [r114544-114552] Russell Bryant * main/channel.c, /: Merged revisions 114548 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r114548 | russell | 2008-04-22 15:25:56 -0500 (Tue, 22 Apr 2008) | 2 lines re-add a fix that got lost with a recent change ........ 2008-04-22 18:14 +0000 [r114541] Jason Parker * main/pbx.c, /, include/asterisk/pbx.h, apps/app_queue.c: Merged revisions 114540 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r114540 | qwell | 2008-04-22 13:14:09 -0500 (Tue, 22 Apr 2008) | 8 lines Allow setqueuevar=yes (et al) to work, after changes to pbx_builtin_setvar() (closes issue #12490) Reported by: bcnit Patches: 12490-queuevars-3.diff uploaded by qwell (license 4) Tested by: qwell ........ 2008-04-22 18:06 +0000 [r114534-114539] Russell Bryant * /, channels/chan_iax2.c: Merged revisions 114538 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r114538 | russell | 2008-04-22 13:04:39 -0500 (Tue, 22 Apr 2008) | 17 lines Merged revisions 114537 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r114537 | russell | 2008-04-22 13:03:33 -0500 (Tue, 22 Apr 2008) | 9 lines If the dial string passed to the call channel callback does not indicate an extension, then consider the extension on the channel before falling back to the default. (closes issue #12479) Reported by: darren1713 Patches: exten_dial_fix_chan_iax2.c.patch uploaded by darren1713 (license 116) ........ ................ 2008-04-22 15:46 +0000 [r114524-114528] Russell Bryant * main/manager.c, /: Merged revisions 114527 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r114527 | russell | 2008-04-22 10:46:01 -0500 (Tue, 22 Apr 2008) | 8 lines Correct action_ping() and action_events() with regards to Manager 1.1 documentation. Also, fix a bug in xml_translate(). (closes issue #11649) Reported by: ys Patches: trunk_manager.c.diff uploaded by ys (license 281) ........ 2008-04-21 20:23 +0000 [r114422] Matthew Fredrickson * channels/chan_zap.c, /: Merged revisions 114389 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r114389 | mattf | 2008-04-21 13:44:35 -0500 (Mon, 21 Apr 2008) | 1 line Add support for generic name transmission (#12484) on SS7 in chan_zap ........ 2008-04-21 15:38 +0000 [r114328] Jeff Peeler * /, apps/app_authenticate.c: Merged revisions 114327 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r114327 | jpeeler | 2008-04-21 10:34:37 -0500 (Mon, 21 Apr 2008) | 2 lines This removes an invalid warning message for an incorrectly entered pin, but more importantly removes an inapplicable check. If the first argument passed to app_authenticate does not contain a '/', the argument should be treated as the sole fixed "password" to match against and that is all. (Previous behavior was attempting to open a file based on the pin.) ........ 2008-04-21 14:42 +0000 [r114321-114324] Joshua Colp * /, channels/chan_sip.c: Merged revisions 114323 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r114323 | file | 2008-04-21 11:40:33 -0300 (Mon, 21 Apr 2008) | 12 lines Merged revisions 114322 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r114322 | file | 2008-04-21 11:39:32 -0300 (Mon, 21 Apr 2008) | 4 lines Only drop audio if we receive it without a progress indication. We allow other frames through such as DTMF because they may be needed to complete the call. (closes issue #12440) Reported by: aragon ........ ................ * /, res/res_config_ldap.c: Merged revisions 114320 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r114320 | file | 2008-04-21 11:34:06 -0300 (Mon, 21 Apr 2008) | 6 lines Only print out the error message if ldap_modify_ext_s actually returns an error, and not success. (closes issue #12438) Reported by: gservat Patches: res_config_ldap.c-patch-code uploaded by gservat (license 466) ........ 2008-04-19 17:00 +0000 [r114304] Matthew Fredrickson * channels/chan_zap.c: SS7:Added - Generic Name / Access Transport / Redirecting Number handling. #12425 2008-04-18 21:51 +0000 [r114277-114286] Russell Bryant * main/manager.c, /: Merged revisions 114285 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r114285 | russell | 2008-04-18 16:51:05 -0500 (Fri, 18 Apr 2008) | 10 lines Merged revisions 114284 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r114284 | russell | 2008-04-18 16:48:06 -0500 (Fri, 18 Apr 2008) | 2 lines Don't destroy a manager session if poll() returns an error of EAGAIN. ........ ................ * Makefile, /: Merged revisions 114279 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r114279 | russell | 2008-04-18 15:01:47 -0500 (Fri, 18 Apr 2008) | 10 lines Merged revisions 114278 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r114278 | russell | 2008-04-18 15:01:09 -0500 (Fri, 18 Apr 2008) | 2 lines ensure directories are created before we try to install stuff into them ........ ................ * Makefile, /: Merged revisions 114276 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r114276 | russell | 2008-04-18 14:59:17 -0500 (Fri, 18 Apr 2008) | 10 lines Merged revisions 114275 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r114275 | russell | 2008-04-18 14:58:55 -0500 (Fri, 18 Apr 2008) | 2 lines SUBDIRS_INSTALL is already listed as a subtarget for bininstall ........ ................ 2008-04-18 19:36 +0000 [r114262-114272] Joshua Colp * channels/chan_unistim.c, /: Merged revisions 114271 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r114271 | file | 2008-04-18 16:35:33 -0300 (Fri, 18 Apr 2008) | 4 lines Make sure ADSI is marked as unavailable on Unistim channels so voicemail does not try to do some ADSI jazz. (closes issue #12460) Reported by: PerryB ........ 2008-04-18 18:04 +0000 [r114260] Mark Michelson * channels/chan_zap.c, /, main/callerid.c: Merged revisions 114259 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r114259 | mmichelson | 2008-04-18 13:03:06 -0500 (Fri, 18 Apr 2008) | 14 lines Merged revisions 114257 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r114257 | mmichelson | 2008-04-18 12:44:29 -0500 (Fri, 18 Apr 2008) | 6 lines Clearing up error messages so they make a bit more sense. Also removing a redundant error message. Issue AST-15 ........ ................ 2008-04-18 16:12 +0000 [r114255] Joshua Colp * /, res/res_config_ldap.c: Merged revisions 114254 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r114254 | file | 2008-04-18 13:11:27 -0300 (Fri, 18 Apr 2008) | 4 lines If the parsing of the config file fails make sure we unlock ldap_lock. (closes issue #12477) Reported by: IgorG ........ 2008-04-18 13:40 +0000 [r114247] Sean Bright * channels/chan_sip.c: Merged revisions 114246 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r114246 | seanbright | 2008-04-18 09:38:07 -0400 (Fri, 18 Apr 2008) | 9 lines Merged revisions 114245 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r114245 | seanbright | 2008-04-18 09:33:32 -0400 (Fri, 18 Apr 2008) | 1 line Only complete the SIP channel name once for 'sip show channel ' ........ ................ 2008-04-18 06:54 +0000 [r114244] Tilghman Lesher * apps/app_setcallerid.c, /: Merged revisions 114243 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r114243 | tilghman | 2008-04-18 01:53:47 -0500 (Fri, 18 Apr 2008) | 11 lines Merged revisions 114242 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r114242 | tilghman | 2008-04-18 01:49:16 -0500 (Fri, 18 Apr 2008) | 3 lines For consistency sake, ensure that the values that ${CALLINGPRES} returns are valid as an input to SetCallingPres. (Closes issue #12472) ........ ................ 2008-04-17 23:09 +0000 [r114232-114241] Russell Bryant * /, channels/chan_sip.c: Merged revisions 114151 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r114151 | oej | 2008-04-15 15:39:29 -0500 (Tue, 15 Apr 2008) | 10 lines Merged revisions 114148 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r114148 | oej | 2008-04-15 22:26:05 +0200 (Tis, 15 Apr 2008) | 2 lines Handle subscribe queues in all situations... Thanks to festr_ on irc for telling me about this bug. ........ ................ * /, channels/chan_sip.c: Merged revisions 114150 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r114150 | oej | 2008-04-15 15:31:08 -0500 (Tue, 15 Apr 2008) | 2 lines Adding chanvar to SIPPEER from 1.4 branch ........ * main/autoservice.c, /: Merged revisions 114233 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r114233 | russell | 2008-04-17 17:24:00 -0500 (Thu, 17 Apr 2008) | 14 lines Merged revisions 114230 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r114230 | russell | 2008-04-17 17:15:43 -0500 (Thu, 17 Apr 2008) | 6 lines Remove redundant safety net. The check for the autoservice channel list state accomplishes the same goal in a better way. (issue #12470) Reported By: atis ........ ................ 2008-04-17 21:05 +0000 [r114228] Mark Michelson * apps/app_chanspy.c, /: Merged revisions 114227 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r114227 | mmichelson | 2008-04-17 16:04:40 -0500 (Thu, 17 Apr 2008) | 17 lines Merged revisions 114226 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r114226 | mmichelson | 2008-04-17 16:03:29 -0500 (Thu, 17 Apr 2008) | 9 lines Declaration of the peer channel in this scope was making it so the peer variable defined in the outer scope was never set properly, therefore making iterating through the channel list always restart from the beginning. This bug would have affected anyone who called chanspy without specifying a first argument. (closes issue #12461) Reported by: stever28 ........ ................ 2008-04-17 16:51 +0000 [r114210-114213] Mark Michelson * main/dsp.c, main/frame.c, /, include/asterisk/dsp.h, include/asterisk/frame.h: Merged revisions 114208 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r114208 | mmichelson | 2008-04-17 11:40:12 -0500 (Thu, 17 Apr 2008) | 20 lines Merged revisions 114207 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r114207 | mmichelson | 2008-04-17 11:28:03 -0500 (Thu, 17 Apr 2008) | 12 lines It was possible for a reference to a frame which was part of a freed DSP to still be referenced, leading to memory corruption and eventual crashes. This code change ensures that the dsp is freed when we are finished with the frame. This change is very similar to a change Russell made with translators back a month or so ago. (closes issue #11999) Reported by: destiny6628 Patches: 11999.patch uploaded by putnopvut (license 60) Tested by: destiny6628, victoryure ........ ................ 2008-04-17 16:26 +0000 [r114206] Russell Bryant * Makefile, /: Merged revisions 114205 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r114205 | russell | 2008-04-17 11:25:29 -0500 (Thu, 17 Apr 2008) | 11 lines Merged revisions 114204 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r114204 | russell | 2008-04-17 11:23:45 -0500 (Thu, 17 Apr 2008) | 3 lines Fix the bininstall target to install from subdirs, as well. (closes issue AST-8, patch from bmd at switchvox) ........ ................ 2008-04-17 15:17 +0000 [r114203] Tilghman Lesher * doc/CODING-GUIDELINES, /: Merged revisions 114202 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r114202 | tilghman | 2008-04-17 10:12:52 -0500 (Thu, 17 Apr 2008) | 2 lines fileio.h does not exist; io.h does, though. ........ 2008-04-17 13:55 +0000 [r114200] Philippe Sultan * /, res/res_jabber.c: Merged revisions 114199 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r114199 | phsultan | 2008-04-17 15:46:17 +0200 (Thu, 17 Apr 2008) | 10 lines Merged revisions 114198 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r114198 | phsultan | 2008-04-17 15:42:23 +0200 (Thu, 17 Apr 2008) | 2 lines Use keepalives effectively in order diagnose bug #12432. ........ ................ 2008-04-17 12:59 +0000 [r114197] Tilghman Lesher * /, res/res_agi.c: Merged revisions 114196 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r114196 | tilghman | 2008-04-17 07:59:04 -0500 (Thu, 17 Apr 2008) | 16 lines Merged revisions 114195 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r114195 | tilghman | 2008-04-17 07:56:38 -0500 (Thu, 17 Apr 2008) | 8 lines Add special case for when the agi cannot be executed, to comply with the documentation that we return failure in that case. (closes issue #12462) Reported by: fmueller Patches: 20080416__bug12462.diff.txt uploaded by Corydon76 (license 14) Tested by: fmueller ........ ................ 2008-04-17 10:56 +0000 [r114193] Sean Bright * apps/app_chanspy.c, /: Merged revisions 114192 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r114192 | seanbright | 2008-04-17 06:55:05 -0400 (Thu, 17 Apr 2008) | 9 lines Merged revisions 114191 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r114191 | seanbright | 2008-04-17 06:51:20 -0400 (Thu, 17 Apr 2008) | 1 line Make sure we have enough room for the recording's filename. ........ ................ 2008-04-16 20:48 +0000 [r114186] Kevin P. Fleming * channels/chan_zap.c, /: Merged revisions 114185 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r114185 | kpfleming | 2008-04-16 15:47:30 -0500 (Wed, 16 Apr 2008) | 14 lines Merged revisions 114184 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r114184 | kpfleming | 2008-04-16 15:46:38 -0500 (Wed, 16 Apr 2008) | 6 lines use the ZT_SET_DIALPARAMS ioctl properly by initializing the structure to all zeroes in case it contains fields that we don't write values into (which it does as of Zaptel 1.4.10) (closes issue #12456) Reported by: fnordian ........ ................ 2008-04-15 20:53 +0000 [r114153] Tilghman Lesher * /, cdr/cdr_pgsql.c: Merged revisions 114152 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r114152 | tilghman | 2008-04-15 15:51:08 -0500 (Tue, 15 Apr 2008) | 2 lines Oops, buffer wasn't long enough for query ........ 2008-04-15 20:09 +0000 [r114147] Steve Murphy * main/pbx.c, /: Merged revisions 114146 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r114146 | murf | 2008-04-15 13:59:50 -0600 (Tue, 15 Apr 2008) | 8 lines These changes: a. fix a self-found problem with SPAWN-ing an extension, where matches were not being found b. correct some wording in a comment c. Add some debug for future debugging. ........ 2008-04-15 17:22 +0000 [r114132-114142] Jason Parker * channels/chan_unistim.c, /: Merged revisions 114141 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r114141 | qwell | 2008-04-15 12:21:58 -0500 (Tue, 15 Apr 2008) | 8 lines Shorten the mac address pattern, since some phones use different identifiers (such as the i2050 softphone). (closes issue #12398) Reported by: c_hans Patches: chan_unistim_svn.diff uploaded by c (license 460) Tested by: c_hans ........ * contrib/scripts/autosupport, /: Merged revisions 114139 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r114139 | qwell | 2008-04-15 12:17:37 -0500 (Tue, 15 Apr 2008) | 15 lines Merged revisions 114138 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r114138 | qwell | 2008-04-15 12:17:18 -0500 (Tue, 15 Apr 2008) | 7 lines Update Digium autosupport script, for more useful information. (closes issue #12452) Reported by: angler Patches: autosupport.diff uploaded by angler (license 106) ........ ................ * /, apps/app_queue.c: Merged revisions 114134 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r114134 | qwell | 2008-04-15 11:18:38 -0500 (Tue, 15 Apr 2008) | 16 lines Merged revisions 114133 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r114133 | qwell | 2008-04-15 11:18:08 -0500 (Tue, 15 Apr 2008) | 8 lines Allow autofill to work in the general section of queues.conf. Additionally, don't try to (re)set options when they have empty values in realtime (all unset columns would have an empty value). (closes issue #12445) Reported by: atis Patches: 12445-autofill.diff uploaded by qwell (license 4) ........ ................ 2008-04-14 18:34 +0000 [r114122] Jason Parker * /, channels/chan_h323.c: Merged revisions 114121 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r114121 | qwell | 2008-04-14 13:34:17 -0500 (Mon, 14 Apr 2008) | 15 lines Merged revisions 114120 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r114120 | qwell | 2008-04-14 13:31:57 -0500 (Mon, 14 Apr 2008) | 7 lines The call_token on the pvt can occasionally be NULL, causing a crash. If it is NULL, we can skip this channel, since it can't the one we're looking for. (closes issue #9299) Reported by: vazir ........ ................ 2008-04-14 17:42 +0000 [r114119] Mark Michelson * main/channel.c, /: Merged revisions 114118 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r114118 | mmichelson | 2008-04-14 12:42:20 -0500 (Mon, 14 Apr 2008) | 19 lines Merged revisions 114117 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r114117 | mmichelson | 2008-04-14 12:41:03 -0500 (Mon, 14 Apr 2008) | 11 lines Increase the retry count when attempting to show channels. This apparently cleared an issue someone was seeing when attempting to show channels when the load was high. (closes issue #11667) Reported by: falves11 Patches: 11677.txt uploaded by russell (license 2) Tested by: falves11 ........ ................ 2008-04-14 16:33 +0000 [r114116] Tilghman Lesher * /, contrib/scripts/astcli: Merged revisions 114115 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r114115 | tilghman | 2008-04-14 11:32:59 -0500 (Mon, 14 Apr 2008) | 2 lines Make tab-completion work for all cases ........ 2008-04-14 16:25 +0000 [r114114] Mark Michelson * apps/app_dial.c, /, apps/app_queue.c: Merged revisions 114113 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r114113 | mmichelson | 2008-04-14 11:25:09 -0500 (Mon, 14 Apr 2008) | 17 lines Merged revisions 114112 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r114112 | mmichelson | 2008-04-14 11:24:22 -0500 (Mon, 14 Apr 2008) | 9 lines If the datastore has been moved to another channel due to a masquerade, then freeing the datastore here causes an eventual double free when the new channel hangs up. We should only free the datastore if we were able to successfully remove it from the channel we are referencing (i.e. the datastore was not moved). (closes issue #12359) Reported by: pguido ........ ................ 2008-04-14 15:02 +0000 [r114108] Mark Michelson * main/channel.c, /: Merged revisions 114107 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r114107 | mmichelson | 2008-04-14 10:01:36 -0500 (Mon, 14 Apr 2008) | 13 lines Merged revisions 114106 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r114106 | mmichelson | 2008-04-14 09:58:02 -0500 (Mon, 14 Apr 2008) | 5 lines Save a local copy of the generate callback prior to unlocking the channel in case the generate callback goes NULL on us after the channel is unlocked. Thanks to Russell for pointing this need out to me. ........ ................ 2008-04-14 14:54 +0000 [r114102-114105] Joshua Colp * /, channels/chan_sip.c: Merged revisions 114104 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r114104 | file | 2008-04-14 11:53:33 -0300 (Mon, 14 Apr 2008) | 12 lines Merged revisions 114103 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r114103 | file | 2008-04-14 11:52:46 -0300 (Mon, 14 Apr 2008) | 4 lines It is possible for the remote side to say they want T38 but not give any capabilities. (closes issue #12414) Reported by: MVF ........ ................ * main/rtp.c, /: Merged revisions 114101 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r114101 | file | 2008-04-14 10:53:33 -0300 (Mon, 14 Apr 2008) | 12 lines Merged revisions 114100 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r114100 | file | 2008-04-14 10:52:49 -0300 (Mon, 14 Apr 2008) | 4 lines Don't change the SSRC when a new source comes into play, this might happen quite often and depending on the remote side... they might not like this. (closes issue #12353) Reported by: dimas ........ ................ 2008-04-14 02:59 +0000 [r114097-114099] Tilghman Lesher * /, contrib/scripts/astcli: Merged revisions 114098 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r114098 | tilghman | 2008-04-13 21:55:41 -0500 (Sun, 13 Apr 2008) | 3 lines Add tab command-line completion (Closes issue #12428) ........ * /, apps/app_meetme.c: Merged revisions 114096 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r114096 | tilghman | 2008-04-13 09:35:43 -0500 (Sun, 13 Apr 2008) | 3 lines Use ast_mkdir instead of mkdir (Closes issue #12430) ........ 2008-04-12 16:22 +0000 [r114094-114095] Matthew Fredrickson * channels/chan_zap.c: Make sure linkset is locked exiting ss7_start_call * channels/chan_zap.c: Make sure we start incoming calls on SS7 with echo cancellation enabled. Also make sure when completing a COT we call ss7_start_call with the proper locks held. Lastly, make sure if we fail to get a channel from zt_new that we don't assume it's there. 2008-04-11 23:27 +0000 [r114089-114091] Tilghman Lesher * /, cdr/cdr_pgsql.c: Merged revisions 114090 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r114090 | tilghman | 2008-04-11 18:26:56 -0500 (Fri, 11 Apr 2008) | 3 lines If any field is not null, but has no default, then it must be set or the insert will fail. (Closes issue #12285) ........ * /, configs/res_ldap.conf.sample: Merged revisions 114088 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r114088 | tilghman | 2008-04-11 18:21:54 -0500 (Fri, 11 Apr 2008) | 3 lines Make the sample config match the contributed LDAP schema (Closes issue #12421) ........ 2008-04-11 23:21 +0000 [r114087] Terry Wilson * /, channels/chan_iax2.c: Merged revisions 114084 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r114084 | twilson | 2008-04-11 17:48:52 -0500 (Fri, 11 Apr 2008) | 15 lines Merged revisions 114083 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r114083 | twilson | 2008-04-11 17:32:51 -0500 (Fri, 11 Apr 2008) | 7 lines Several places in the code called find_callno() (which releases the lock on the pvt structure) and then immediately locked the call and did things with it. Unfortunately, the call can disappear between the find_callno and the lock, causing Bad Stuff(tm) to happen. Added find_callno_locked() function to return the callno withtout unlocking for instances that it is needed. (issue #12400) Reported by: ztel ........ ................ 2008-04-11 23:13 +0000 [r114086] Tilghman Lesher * /, res/res_config_ldap.c: Merged revisions 114085 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r114085 | tilghman | 2008-04-11 18:12:16 -0500 (Fri, 11 Apr 2008) | 7 lines Use the correct function for free'ing objects, and maybe we won't crash. (closes issue #12163) Reported by: gservat Patches: 20080411__bug12163.diff.txt uploaded by Corydon76 (license 14) Tested by: gservat ........ 2008-04-11 15:51 +0000 [r114065] Mark Michelson * /, main/features.c: Merged revisions 114064 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r114064 | mmichelson | 2008-04-11 10:49:35 -0500 (Fri, 11 Apr 2008) | 19 lines Merged revisions 114063 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r114063 | mmichelson | 2008-04-11 10:44:28 -0500 (Fri, 11 Apr 2008) | 11 lines Fix a race condition that may happen between a sip hangup and a "core show channel" command. This patch adds locking to prevent the resulting crash. (closes issue #12155) Reported by: tsearle Patches: show_channels_crash2.patch uploaded by tsearle (license 373) Tested by: tsearle ........ ................ 2008-04-11 14:56 +0000 [r114062] Tilghman Lesher * /, res/res_config_ldap.c: Merged revisions 114061 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r114061 | tilghman | 2008-04-11 09:54:22 -0500 (Fri, 11 Apr 2008) | 6 lines Errors are all greater than 0 (closes issue #12422) Reported by: nito Patches: res_config_ldap_result_check_patch.diff uploaded by nito (license 340) ........ 2008-04-10 22:23 +0000 [r114056] Mark Michelson * utils/conf2ael.c, utils/check_expr.c, utils/Makefile, main/manager.c, /, utils/astman.c, utils/hashtest.c, main/utils.c, include/asterisk/lock.h, utils/ael_main.c, utils/hashtest2.c: Merged revisions 114052 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r114052 | mmichelson | 2008-04-10 17:02:32 -0500 (Thu, 10 Apr 2008) | 11 lines Merged revisions 114051 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r114051 | mmichelson | 2008-04-10 15:59:49 -0500 (Thu, 10 Apr 2008) | 3 lines Fix 1.4 build when LOW_MEMORY is enabled. ........ ................ 2008-04-10 19:59 +0000 [r114047] Mark Michelson * /, channels/chan_sip.c: Merged revisions 114046 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r114046 | mmichelson | 2008-04-10 14:58:36 -0500 (Thu, 10 Apr 2008) | 14 lines Merged revisions 114045 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r114045 | mmichelson | 2008-04-10 14:55:33 -0500 (Thu, 10 Apr 2008) | 6 lines Be sure that we're not about to set bridgepvt NULL prior to dereferencing it. (closes issue #11775) Reported by: fujin ........ ................ 2008-04-10 19:09 +0000 [r114043] Tilghman Lesher * /, contrib/scripts/astcli: Merged revisions 114042 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r114042 | tilghman | 2008-04-10 14:04:29 -0500 (Thu, 10 Apr 2008) | 7 lines The hydra grows yet another head... (closes issue #12401) Reported by: davevg Patches: astcli.diff2 uploaded by davevg (license 209) Tested by: davevg, Corydon76 ........ 2008-04-10 17:27 +0000 [r114037] Jason Parker * /, main/file.c: Merged revisions 114036 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r114036 | qwell | 2008-04-10 12:27:16 -0500 (Thu, 10 Apr 2008) | 18 lines Merged revisions 114035 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r114035 | qwell | 2008-04-10 12:26:10 -0500 (Thu, 10 Apr 2008) | 10 lines Only try to prefix language if we are not using an absolute path (suffix it otherwise). en/var/lib/asterisk/sounds/blah.gsm is a very silly path. (closes issue #12379) Reported by: kuj Patches: 12379-absolutepath.diff uploaded by qwell (license 4) Tested by: kuj, qwell ........ ................ 2008-04-10 16:00 +0000 [r114023-114034] Joshua Colp * /, apps/app_meetme.c: Merged revisions 114030 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r114030 | file | 2008-04-10 12:10:47 -0300 (Thu, 10 Apr 2008) | 14 lines Merged revisions 114029 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r114029 | file | 2008-04-10 12:09:04 -0300 (Thu, 10 Apr 2008) | 6 lines Create the directory where name recordings will go if it does not exist. (closes issue #12311) Reported by: rkeene Patches: 12311-mkdir.diff uploaded by qwell (license 4) ........ ................ * apps/app_voicemail.c, /: Merged revisions 114027 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r114027 | file | 2008-04-10 11:53:19 -0300 (Thu, 10 Apr 2008) | 6 lines Don't hardcode ru into the digits filename so that languageprefix can work. (closes issue #12404) Reported by: IgorG Patches: voicemail_ru_hardcoded-v1.patch uploaded by IgorG (license 20) ........ * main/rtp.c, channels/chan_unistim.c, /, channels/chan_skinny.c: Merged revisions 114024 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r114024 | file | 2008-04-10 10:45:45 -0300 (Thu, 10 Apr 2008) | 4 lines Fix spelling of existent in a few places. (closes issue #12409) Reported by: candlerb ........ * /, channels/chan_sip.c: Merged revisions 114022 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r114022 | file | 2008-04-10 10:28:30 -0300 (Thu, 10 Apr 2008) | 14 lines Merged revisions 114021 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r114021 | file | 2008-04-10 10:27:11 -0300 (Thu, 10 Apr 2008) | 6 lines Don't add custom URI options if they don't exist OR they are empty. (closes issue #12407) Reported by: homesick Patches: uri_options-1.4.diff uploaded by homesick (license 91) ........ ................ 2008-04-09 22:34 +0000 [r113929-113982] Mark Michelson * /, apps/app_queue.c: Merged revisions 113980 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r113980 | mmichelson | 2008-04-09 17:32:32 -0500 (Wed, 09 Apr 2008) | 8 lines Fix a crash that happened due to accessing free'd memory (closes issue #12396) Reported by: tcalosi Patches: 12396.patch uploaded by putnopvut (license 60) Tested by: tcalosi ........ * /, channels/chan_sip.c: Merged revisions 113928 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r113928 | mmichelson | 2008-04-09 15:56:14 -0500 (Wed, 09 Apr 2008) | 16 lines Merged revisions 113927 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r113927 | mmichelson | 2008-04-09 15:54:31 -0500 (Wed, 09 Apr 2008) | 8 lines We need to set the persistant_route [sic] parameter for the sip_pvt during the initial INVITE, no matter if we're building the route set from an INVITE request or response. (closes issue #12391) Reported by: benjaminbohlmann Tested by: benjaminbohlmann ........ ................ 2008-04-09 19:02 +0000 [r113876] Tilghman Lesher * cdr/cdr_csv.c, /, configs/cdr.conf.sample: Merged revisions 113875 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r113875 | tilghman | 2008-04-09 14:00:40 -0500 (Wed, 09 Apr 2008) | 12 lines Merged revisions 113874 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r113874 | tilghman | 2008-04-09 13:57:33 -0500 (Wed, 09 Apr 2008) | 4 lines If the [csv] section does not exist in cdr.conf, then an unload/load sequence is needed to correct the problem. Track whether the load succeeded with a variable, so we can fix this with a simple reload event, instead. ........ ................ 2008-04-09 17:56 +0000 [r113839] Jason Parker * /, contrib/scripts/astcli: Merged revisions 113838 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r113838 | qwell | 2008-04-09 12:56:07 -0500 (Wed, 09 Apr 2008) | 2 lines Fix a small file handle "leak" pointed out by jjshoe on #asterisk. ........ 2008-04-09 17:50 +0000 [r113837] Mark Michelson * main/pbx.c, /: Merged revisions 113836 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r113836 | mmichelson | 2008-04-09 12:48:33 -0500 (Wed, 09 Apr 2008) | 14 lines There was a subtle logical difference between 1.4 and trunk with regards to how timeouts were handled. In 1.4, if the absolute timeout were reached on a call, no matter what the return value of ast_spawn_extension was, the pbx would attempt to go to the 'T' extension or hangup otherwise. The rearrangement of this function in trunk made this check only happen in the case that ast_spawn_extension returned 0. If ast_spawn_extension returned 1, then the fact that the timeout expired resulted in a no-op, and would cause an infinite loop to occur in __ast_pbx_run. This change fixes this problem. Now timeouts will behave as they did in 1.4 (closes issue #11550) Reported by: pj Tested by: putnopvut ........ 2008-04-09 16:53 +0000 [r113786] Joshua Colp * /, channels/chan_iax2.c: Merged revisions 113785 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r113785 | file | 2008-04-09 13:52:04 -0300 (Wed, 09 Apr 2008) | 12 lines Merged revisions 113784 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r113784 | file | 2008-04-09 13:50:45 -0300 (Wed, 09 Apr 2008) | 4 lines If we receive an AUTHREQ from the remote server and we are unable to reply (for example they have a secret configured, but we do not) then queue a hangup frame on the Asterisk channel. This will cause the channel to hangup and a HANGUP to be sent via IAX2 to the remote side which is the proper thing to do in this scenario. (closes issue #12385) Reported by: viraptor ........ ................ 2008-04-09 14:42 +0000 [r113683] Mark Michelson * /, channels/chan_sip.c: Merged revisions 113682 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r113682 | mmichelson | 2008-04-09 09:41:58 -0500 (Wed, 09 Apr 2008) | 17 lines Merged revisions 113681 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r113681 | mmichelson | 2008-04-09 09:40:05 -0500 (Wed, 09 Apr 2008) | 9 lines If Asterisk receives a 488 on an INVITE (not a reinvite), then we should not send a BYE. (closes issue #12392) Reported by: fnordian Patches: chan_sip.patch uploaded by fnordian (license 110) with small modification from me ........ ................ 2008-04-09 13:56 +0000 [r113648-113650] Tilghman Lesher * /, contrib/scripts/astcli: Merged revisions 113647 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r113647 | tilghman | 2008-04-09 08:23:44 -0500 (Wed, 09 Apr 2008) | 6 lines Additional enhancements (closes issue #12390) Reported by: tzafrir Patches: astcli_fixes.diff uploaded by tzafrir (license 46) ........ 2008-04-09 01:40 +0000 [r113598] Terry Wilson * /, channels/chan_iax2.c: Merged revisions 113597 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r113597 | twilson | 2008-04-08 20:36:58 -0500 (Tue, 08 Apr 2008) | 10 lines Merged revisions 113596 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r113596 | twilson | 2008-04-08 20:34:25 -0500 (Tue, 08 Apr 2008) | 2 lines Initialize fr->cacheable to make valgrind happy ........ ................ 2008-04-08 21:34 +0000 [r113560] Tilghman Lesher * /, contrib/scripts/astcli (added): Merged revisions 113559 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r113559 | tilghman | 2008-04-08 16:33:11 -0500 (Tue, 08 Apr 2008) | 6 lines Add commandline tool for doing CLI commands through AMI (instead of using asterisk -rx) (closes issue #12389) Reported by: davevg Patches: astcli uploaded by davevg (license 209) ........ 2008-04-08 18:49 +0000 [r113404-113506] Jason Parker * /, channels/chan_skinny.c: Merged revisions 113505 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r113505 | qwell | 2008-04-08 13:49:21 -0500 (Tue, 08 Apr 2008) | 9 lines Merged revisions 113504 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r113504 | qwell | 2008-04-08 13:48:55 -0500 (Tue, 08 Apr 2008) | 1 line Add a little more that is required for previously added devices. ........ ................ * /, channels/chan_skinny.c: Merged revisions 113455 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r113455 | qwell | 2008-04-08 13:08:35 -0500 (Tue, 08 Apr 2008) | 12 lines Merged revisions 113454 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r113454 | qwell | 2008-04-08 13:07:49 -0500 (Tue, 08 Apr 2008) | 4 lines Add support for several new(ish) devices - most notably, 7942/7945, 7962/7965, 7975. Thanks to Greg Oliver for providing me the required information. ........ ................ * main/asterisk.c, /: Merged revisions 113403 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r113403 | qwell | 2008-04-08 12:00:55 -0500 (Tue, 08 Apr 2008) | 9 lines Merged revisions 113402 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r113402 | qwell | 2008-04-08 11:56:52 -0500 (Tue, 08 Apr 2008) | 1 line Work around some silliness caused by sys/capability.h - this should fix compile errors a number of users have been experiencing. ........ ................ 2008-04-08 16:56 +0000 [r113350-113401] Tilghman Lesher * /, contrib/scripts/astgenkey.8: Merged revisions 113400 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r113400 | tilghman | 2008-04-08 11:54:21 -0500 (Tue, 08 Apr 2008) | 14 lines Merged revisions 113399 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r113399 | tilghman | 2008-04-08 11:51:28 -0500 (Tue, 08 Apr 2008) | 6 lines Add security note on astgenkey's manpage. (closes issue #12373) Reported by: lmamane Patches: 20080406__bug12373.diff.txt uploaded by Corydon76 (license 14) ........ ................ * /, channels/chan_sip.c: Merged revisions 113349 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r113349 | tilghman | 2008-04-08 10:48:58 -0500 (Tue, 08 Apr 2008) | 15 lines Merged revisions 113348 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r113348 | tilghman | 2008-04-08 10:39:16 -0500 (Tue, 08 Apr 2008) | 7 lines Move check for still-bridged channels out a little further, to avoid possible deadlocks. (Closes issue #12252) Reported by: callguy Patches: 20080319__bug12252.diff.txt uploaded by Corydon76 (license 14) Tested by: callguy ........ ................ 2008-04-08 15:10 +0000 [r113298-113299] Joshua Colp * /, main/audiohook.c: Merged revisions 113297 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r113297 | file | 2008-04-08 12:05:35 -0300 (Tue, 08 Apr 2008) | 12 lines Merged revisions 113296 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r113296 | file | 2008-04-08 12:03:43 -0300 (Tue, 08 Apr 2008) | 4 lines If audio suddenly gets fed into one side of a channel after a lapse of frames flush the other factory so that old audio does not remain in the factory causing the sync code to not execute. (closes issue #12296) Reported by: jvandal ........ ................ 2008-04-07 22:17 +0000 [r113246] Tilghman Lesher * /, configs/manager.conf.sample: Merged revisions 113245 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r113245 | tilghman | 2008-04-07 17:16:46 -0500 (Mon, 07 Apr 2008) | 2 lines Additional note ........ 2008-04-07 21:49 +0000 [r113244] Jason Parker * /, configs/manager.conf.sample: Merged revisions 113243 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r113243 | qwell | 2008-04-07 16:49:27 -0500 (Mon, 07 Apr 2008) | 1 line Document 'originate' permission in manager sample config. ........ 2008-04-07 21:36 +0000 [r113242] Jeff Peeler * /, channels/chan_sip.c: Merged revisions 113241 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r113241 | jpeeler | 2008-04-07 16:35:48 -0500 (Mon, 07 Apr 2008) | 23 lines Merged revisions 113013 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r113013 | jpeeler | 2008-04-07 10:18:10 -0500 (Mon, 07 Apr 2008) | 15 lines Merged revisions 113012 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r113012 | jpeeler | 2008-04-07 10:16:44 -0500 (Mon, 07 Apr 2008) | 7 lines (closes issue #12362) (closes issue #12372) Reported by: vinsik Tested by: tecnoxarxa This one line change makes an if inside a for loop (in realtime_peer) check all the ast_variables the loop was intending to test rather than just the first one. ........ ................ ................ 2008-04-07 19:10 +0000 [r113174] Jason Parker * /, channels/chan_skinny.c, configs/skinny.conf.sample: Merged revisions 113119 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r113119 | qwell | 2008-04-07 13:02:51 -0500 (Mon, 07 Apr 2008) | 16 lines Merged revisions 113118 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r113118 | qwell | 2008-04-07 13:00:09 -0500 (Mon, 07 Apr 2008) | 8 lines Allow playback with noanswer (and add earlyrtp option). (closes issue #9077) Reported by: pj Patches: earlyrtp.diff uploaded by wedhorn (license 30) Tested by: pj, qwell, DEA, wedhorn ........ ................ 2008-04-07 19:08 +0000 [r113173] Tilghman Lesher * /, funcs/func_strings.c: Merged revisions 113172 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r113172 | tilghman | 2008-04-07 14:06:46 -0500 (Mon, 07 Apr 2008) | 11 lines Merged revisions 113117 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r113117 | tilghman | 2008-04-07 12:51:49 -0500 (Mon, 07 Apr 2008) | 3 lines Force ast_mktime() to check for DST, since strptime(3) does not. (Closes issue #12374) ........ ................ 2008-04-07 16:13 +0000 [r113067] Mark Michelson * main/channel.c, /: Merged revisions 113066 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r113066 | mmichelson | 2008-04-07 11:12:30 -0500 (Mon, 07 Apr 2008) | 21 lines Merged revisions 113065 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r113065 | mmichelson | 2008-04-07 11:08:45 -0500 (Mon, 07 Apr 2008) | 13 lines This fix prevents a deadlock that was experienced in chan_local. There was deadlock prevention in place in chan_local, but it would not work in a specific case because the channel was recursively locked. By unlocking the channel prior to calling the generator's generate callback in ast_read_generator_actions(), we prevent the recursive locking, and therefore the deadlock. (closes issue #12307) Reported by: callguy Patches: 12307.patch uploaded by putnopvut (license 60) Tested by: callguy ........ ................ 2008-04-07 15:28 +0000 [r113042] Jeff Peeler * /, channels/chan_sip.c: Merged revisions 113013 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r113013 | jpeeler | 2008-04-07 10:18:10 -0500 (Mon, 07 Apr 2008) | 15 lines Merged revisions 113012 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r113012 | jpeeler | 2008-04-07 10:16:44 -0500 (Mon, 07 Apr 2008) | 7 lines (closes issue #12362) (closes issue #12372) Reported by: vinsik Tested by: tecnoxarxa This one line change makes an if inside a for loop (in realtime_peer) check all the ast_variables the loop was intending to test rather than just the first one. ........ ................ 2008-04-05 13:30 +0000 [r112973-112975] Tilghman Lesher * /, res/res_agi.c: Merged revisions 112972 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r112972 | tilghman | 2008-04-05 08:24:12 -0500 (Sat, 05 Apr 2008) | 6 lines AsyncAGI should not close the manager session on error. (closes issue #12370) Reported by: srt Patches: asterisk-12370.diff uploaded by srt (license 378) ........ 2008-04-04 19:30 +0000 [r112786-112822] Philippe Sultan * /, channels/chan_gtalk.c: Merged revisions 112821 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r112821 | phsultan | 2008-04-04 21:28:49 +0200 (Fri, 04 Apr 2008) | 9 lines Merged revisions 112820 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r112820 | phsultan | 2008-04-04 21:26:15 +0200 (Fri, 04 Apr 2008) | 1 line Free newly allocated channel before returning ........ ................ * /, channels/chan_gtalk.c: Merged revisions 112785 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r112785 | phsultan | 2008-04-04 19:32:46 +0200 (Fri, 04 Apr 2008) | 15 lines Merged revisions 112766 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r112766 | phsultan | 2008-04-04 19:16:59 +0200 (Fri, 04 Apr 2008) | 7 lines Prevent call connections when codecs don't match. (closes issue #10604) Reported by: keepitcool Patches: branch-1.4-10604-2.diff uploaded by phsultan (license 73) Tested by: phsultan ........ ................ 2008-04-04 01:08 +0000 [r112715] Dwayne M. Hubbard * main/asterisk.c, /: Merged revisions 112653,112656,112714 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r112653 | dhubbard | 2008-04-03 17:13:11 -0500 (Thu, 03 Apr 2008) | 1 line add a Zaptel timer check to verify the timer is responding when Zaptel support is compiled into Asterisk and Zaptel drivers are loaded. This will help people not waste their valuable time debugging side effects. ........ r112656 | dhubbard | 2008-04-03 17:19:43 -0500 (Thu, 03 Apr 2008) | 1 line satisfy buildbot ........ r112714 | dhubbard | 2008-04-03 19:57:33 -0500 (Thu, 03 Apr 2008) | 1 line sleep long enough for the zaptel timer error message to display before exit ........ 2008-04-04 00:54 +0000 [r112713] Joshua Colp * /, main/Makefile: Merged revisions 112712 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r112712 | file | 2008-04-03 21:53:19 -0300 (Thu, 03 Apr 2008) | 10 lines Merged revisions 112711 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r112711 | file | 2008-04-03 21:52:36 -0300 (Thu, 03 Apr 2008) | 2 lines Pass in the path to Zaptel for systems that install Zaptel headers in a separate location. ........ ................ 2008-04-03 14:42 +0000 [r112601] Mark Michelson * channels/chan_zap.c, /: Merged revisions 112600 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r112600 | mmichelson | 2008-04-03 09:35:47 -0500 (Thu, 03 Apr 2008) | 17 lines Merged revisions 112599 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r112599 | mmichelson | 2008-04-03 09:32:20 -0500 (Thu, 03 Apr 2008) | 9 lines Fix the testing of the "res" variable so that it is more logically correct and makes the correct warning and debug messages print. (closes issue #12361) Reported by: one47 Patches: chan_zap_deferred_digit.patch uploaded by one47 (license 23) ........ ................ 2008-04-02 17:37 +0000 [r112470] Mark Michelson * main/manager.c, /: Merged revisions 112469 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r112469 | mmichelson | 2008-04-02 12:36:49 -0500 (Wed, 02 Apr 2008) | 21 lines Merged revisions 112468 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r112468 | mmichelson | 2008-04-02 12:36:04 -0500 (Wed, 02 Apr 2008) | 13 lines Fix a race condition in the manager. It is possible that a new manager event could be appended during a brief time when the manager is not waiting for input. If an event comes during this period, we need to set an indicator that there is an event pending so that the manager doesn't attempt to wait forever for an event that already happened. (closes issue #12354) Reported by: bamby Patches: manager_race_condition.diff uploaded by bamby (license 430) (comments added by me) ........ ................ 2008-04-02 15:27 +0000 [r112436] Joshua Colp * /, channels/chan_sip.c: Merged revisions 112431 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r112431 | file | 2008-04-02 12:26:51 -0300 (Wed, 02 Apr 2008) | 7 lines Since the SIP request structure gets reused multiple times with TCP handling we have to clear the debug state or else we will keep spitting out debug even after it has been turned off. (closes issue #12169) Reported by: pj Patches: 12169-debugoff-2.diff uploaded by qwell (license 4) Tested by: pj ........ 2008-04-02 14:33 +0000 [r112395] Mark Michelson * /, apps/app_queue.c: Merged revisions 112394 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r112394 | mmichelson | 2008-04-02 09:32:43 -0500 (Wed, 02 Apr 2008) | 14 lines Merged revisions 112393 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r112393 | mmichelson | 2008-04-02 09:32:00 -0500 (Wed, 02 Apr 2008) | 6 lines Ensure that there is no timeout if none is specified. (closes issue #12349) Reported by: johnlange ........ ................ 2008-04-01 22:48 +0000 [r112359] Steve Murphy * main/pbx.c, /: Merged revisions 112357 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r112357 | murf | 2008-04-01 16:45:10 -0600 (Tue, 01 Apr 2008) | 1 line Bumped across another test set for the new exten pattern matcher, which revealed a problem with the CANMATCH/MATCHMORE modes. Direct matches were getting in the way. Fixed. ........ 2008-04-01 20:20 +0000 [r112299] Steve Murphy * main/pbx.c, /: Merged revisions 112289 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r112289 | murf | 2008-04-01 14:02:19 -0600 (Tue, 01 Apr 2008) | 21 lines (closes issue #12298) Reported by: falves11 Patches: 12298.patch1 uploaded by murf (license 17) Tested by: murf I have hopes that the changes made over the last few days will finalize and solidify this code. While there are bound to be small tweaks still needed, I feel that the job (at last) is somewhat completed. Finally, I had a chance to comprehend how the scoring of extension patterns was done in the previous version, and I've come very close to using the exact same criteria in the new pattern matching code. The left-right sorting is now replicated in the trie structure itself, such that the first match found will the 'best' match. Compared the results against 1.4 for several extensions. Replicated falves11's setup and it works. Used some devious patterns provided by jsmith, supplemented with a few of my own. Looks good. ........ 2008-04-01 18:09 +0000 [r112211] Joshua Colp * main/rtp.c, /: Merged revisions 112210 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r112210 | file | 2008-04-01 15:06:13 -0300 (Tue, 01 Apr 2008) | 12 lines Merged revisions 112209 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r112209 | file | 2008-04-01 15:02:43 -0300 (Tue, 01 Apr 2008) | 4 lines Disable Packet2Packet bridging when we need to feed DTMF frames into the core. Some implementations do not like how we switch between things. (closes issue #12212) Reported by: bamby ........ ................ 2008-04-01 17:52 +0000 [r112170-112206] Joshua Colp * /, channels/chan_sip.c: Merged revisions 112205 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r112205 | file | 2008-04-01 14:48:52 -0300 (Tue, 01 Apr 2008) | 12 lines Merged revisions 112204 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r112204 | file | 2008-04-01 14:43:46 -0300 (Tue, 01 Apr 2008) | 4 lines Do not pass audio until the remote side has indicated they are providing early media, or if the channel has been answered. (closes issue #11823) Reported by: SDamm ........ ................ 2008-04-01 17:25 +0000 [r112157] Mark Michelson * main/dns.c, /: Merged revisions 112148 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r112148 | mmichelson | 2008-04-01 12:23:19 -0500 (Tue, 01 Apr 2008) | 18 lines Merged revisions 112138 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r112138 | mmichelson | 2008-04-01 12:21:21 -0500 (Tue, 01 Apr 2008) | 10 lines Initialize the __res_state structure used for dns purposes to all 0's prior to using it. This is due to valgrind's complaints on issue #12284 as well as an excerpt found in "Description" portion of the online man page found here: http://www.iti.cs.tu-bs.de/cgi-bin/UNIXhelp/man-cgi?res_nquery+3RESOLV (pertains to issue #12284 but does not necessarily close it) ........ ................ 2008-04-01 16:57 +0000 [r112127] Joshua Colp * include/asterisk/slinfactory.h, /, main/slinfactory.c: Merged revisions 112126 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r112126 | file | 2008-04-01 13:50:37 -0300 (Tue, 01 Apr 2008) | 13 lines Merged revisions 112125 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r112125 | file | 2008-04-01 13:45:14 -0300 (Tue, 01 Apr 2008) | 5 lines Ensure that we do not exceed the hold's maximum size with a single frame. (closes issue #12047) Reported by: fabianoheringer Tested by: fabianoheringer ........ ................ 2008-03-31 22:17 +0000 [r112070-112072] Jason Parker * apps/app_voicemail.c, /: Merged revisions 112069 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r112069 | qwell | 2008-03-31 16:48:30 -0500 (Mon, 31 Mar 2008) | 13 lines Merged revisions 112068 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r112068 | qwell | 2008-03-31 16:48:05 -0500 (Mon, 31 Mar 2008) | 5 lines Fix a silly infinite loop when choosing an invalid option. (closes issue #12315) Reported by: jmls ........ ................ 2008-03-31 21:03 +0000 [r112034-112036] Terry Wilson * /, main/http.c: Merged revisions 112033 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r112033 | twilson | 2008-03-31 15:45:05 -0500 (Mon, 31 Mar 2008) | 2 lines Handle blank prefix= in http.conf ........ 2008-03-31 17:15 +0000 [r111997-111999] Russell Bryant * Makefile, /: Merged revisions 111998 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r111998 | russell | 2008-03-31 12:14:58 -0500 (Mon, 31 Mar 2008) | 7 lines Ensure configure gets run on a clean checkout. (closes issue #12197) Reported by: juggie Patches: 12197.diff uploaded by juggie (license 24) ........ 2008-03-31 14:22 +0000 [r111962] Joshua Colp * res/res_config_sqlite.c, /: Merged revisions 111961 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r111961 | file | 2008-03-31 11:20:39 -0300 (Mon, 31 Mar 2008) | 4 lines Initialize all these here tmp pointers at declaration. They confused some compilers a wee bit. (closes issue #12333) Reported by: ovi ........ 2008-03-29 Russell Bryant * Asterisk 1.6.0-beta7.1 released. Asterisk 1.6.0-beta7 was tagged against trunk, instead of the 1.6.0 branch. 2008-03-28 21:46 +0000 [r111858] Jason Parker * codecs/gsm/inc/private.h, /: Merged revisions 111857 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r111857 | qwell | 2008-03-28 16:46:02 -0500 (Fri, 28 Mar 2008) | 20 lines Merged revisions 111856 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r111856 | qwell | 2008-03-28 16:45:35 -0500 (Fri, 28 Mar 2008) | 12 lines Allow gsm to compile correctly on x86 with gcc4 optimizations. (closes issue #11243) Reported by: whiskerp Patches: 11243-maybe-asm.diff uploaded by qwell (license 4) Tested by: Seggy (IRC) Note: While I did write this patch, I would not have found this if fossil had not reported and fixed issue #12253. A huge thanks to him for helping to (indirectly) find the problem here. ........ ................ 2008-03-28 19:11 +0000 [r111722-111776] Jason Parker * /, channels/chan_skinny.c: Merged revisions 111721 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r111721 | qwell | 2008-03-28 12:57:12 -0500 (Fri, 28 Mar 2008) | 9 lines Merged revisions 111720 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r111720 | qwell | 2008-03-28 12:55:05 -0500 (Fri, 28 Mar 2008) | 1 line Remove unimplemented softkeys. Prompted by issue #12325. ........ ................ 2008-03-28 16:21 +0000 [r111660] Jason Parker * /, formats/format_wav_gsm.c: Merged revisions 111659 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r111659 | qwell | 2008-03-28 11:20:59 -0500 (Fri, 28 Mar 2008) | 16 lines Merged revisions 111658 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r111658 | qwell | 2008-03-28 11:19:56 -0500 (Fri, 28 Mar 2008) | 8 lines The file size of WAV49 does not need to be an even number. (closes issue #12128) Reported by: mdu113 Patches: 12128-noevenlength.diff uploaded by qwell (license 4) Tested by: qwell, mdu113 ........ ................ 2008-03-28 14:43 +0000 [r111607-111608] Tilghman Lesher * doc/valgrind.txt, /: Merged revisions 111606 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r111606 | tilghman | 2008-03-28 09:37:28 -0500 (Fri, 28 Mar 2008) | 11 lines Merged revisions 111605 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r111605 | tilghman | 2008-03-28 09:35:45 -0500 (Fri, 28 Mar 2008) | 3 lines Update debugging text, since Valgrind eliminated the --log-file-exactly option. (Closes issue #12320) ........ ................ 2008-03-28 00:56 +0000 [r111566] Joshua Colp * /, apps/app_queue.c: Merged revisions 111565 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r111565 | file | 2008-03-27 21:55:47 -0300 (Thu, 27 Mar 2008) | 2 lines Forgetting to unregister a manager action is bad, mmmk? ........ 2008-03-28 00:17 +0000 [r111534] Mark Michelson * /, apps/app_queue.c: Merged revisions 111533 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r111533 | mmichelson | 2008-03-27 19:12:52 -0500 (Thu, 27 Mar 2008) | 10 lines Fix a crash that would happen when attempting to unload the app_queue module. The problem was that when the refcount on the queue hit 0, the destructor was called, and inside the destructor, another function was called which would increase the refcount back to 1 again and then decrease it again back to 0 for every member in the queue. This meant that the destructor was being recursively called, leading to a double free of the queue. This is now fixed by making sure to unlink the queue from the queues container prior to the final unref of the queue. ........ 2008-03-27 21:28 +0000 [r111498] Steve Murphy * main/pbx.c, /: Merged revisions 111497 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r111497 | murf | 2008-03-27 15:25:55 -0600 (Thu, 27 Mar 2008) | 1 line comment cleanup and iron out a really dumb mistake in handling the '.'-wildcard in the new exten pattern matcher. ........ 2008-03-27 19:30 +0000 [r111444] Tilghman Lesher * /, main/acl.c: Merged revisions 111443 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r111443 | tilghman | 2008-03-27 14:26:45 -0500 (Thu, 27 Mar 2008) | 14 lines Merged revisions 111442 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r111442 | tilghman | 2008-03-27 14:23:12 -0500 (Thu, 27 Mar 2008) | 6 lines For FreeBSD, at least, the ifa_addr element could be NULL. (closes issue #12300) Reported by: festr Patches: acl.c.patch uploaded by festr (license 443) ........ ................ 2008-03-27 13:42 +0000 [r111361-111411] Steve Murphy * apps/app_playback.c, main/pbx.c, /: Merged revisions 111410 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r111410 | murf | 2008-03-27 07:29:41 -0600 (Thu, 27 Mar 2008) | 17 lines Merged revisions 111391 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r111391 | murf | 2008-03-27 07:03:28 -0600 (Thu, 27 Mar 2008) | 9 lines These small documentation updates made in response to a query in asterisk-users, where a user was using Playback, but needed the features of Background, and had no idea that Background existed, or that it might provide the features he needed. I thought the best way to avert these kinds of queries was to provide "See Also" references in all three of "Background", "Playback", "WaitExten". Perhaps a project to do this with all related apps is in order. ........ ................ * res/ael/pval.c, /, include/asterisk/pval.h, pbx/pbx_ael.c, include/asterisk/ael_structs.h: Merged revisions 111360 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r111360 | murf | 2008-03-26 22:47:12 -0600 (Wed, 26 Mar 2008) | 23 lines Merged revisions 111341 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r111341 | murf | 2008-03-26 21:21:05 -0600 (Wed, 26 Mar 2008) | 15 lines (closes issue #12302) Reported by: pj Tested by: murf These changes will set a channel variable ~~EXTEN~~ just before generating code for a switch, with the value of ${EXTEN}. The exten is marked as having a switch, and ever after that, till the end of the exten, we substitute any ${EXTEN} with ${~~EXTEN~~} instead in application arguments; (and the ${EXTEN: also). The reason for this, is that because switches are coded using separate extensions to provide pattern matching, and jumping to/from these switch extensions messes up the ${EXTEN} value, which blows the minds of users. ........ ................ 2008-03-27 00:36 +0000 [r111247-111339] Jason Parker * main/frame.c, /: Merged revisions 111285 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r111285 | qwell | 2008-03-26 19:25:56 -0500 (Wed, 26 Mar 2008) | 9 lines Merged revisions 111280 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r111280 | qwell | 2008-03-26 19:25:13 -0500 (Wed, 26 Mar 2008) | 1 line Put this flag back so we don't change the API. ........ ................ * main/frame.c, /: Merged revisions 111246 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r111246 | qwell | 2008-03-26 18:27:33 -0500 (Wed, 26 Mar 2008) | 17 lines Merged revisions 111245 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r111245 | qwell | 2008-03-26 18:26:33 -0500 (Wed, 26 Mar 2008) | 9 lines Remove excessive smoother optimization that was causing audio glitches (small "pops") after (about 200ms later) an "incorrectly" sized frame was received. While it would be very nice to keep this as optimized as possible, it makes no sense for the smoother to be dropping random bits of audio like this. Isn't that the whole point of a smoother? Closes issue #12093. ........ ................ 2008-03-26 19:57 +0000 [r111131] Joshua Colp * contrib/scripts/autosupport, /: Merged revisions 111130 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r111130 | file | 2008-03-26 16:56:40 -0300 (Wed, 26 Mar 2008) | 14 lines Merged revisions 111129 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r111129 | file | 2008-03-26 16:55:08 -0300 (Wed, 26 Mar 2008) | 6 lines Update autosupport script. (closes issue #12310) Reported by: angler Patches: autosupport.diff uploaded by angler (license 106) ........ ................ 2008-03-26 19:53 +0000 [r111128] Kevin P. Fleming * /, UPGRADE.txt: Merged revisions 111127 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r111127 | kpfleming | 2008-03-26 14:52:27 -0500 (Wed, 26 Mar 2008) | 18 lines Merged revisions 111126 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r111126 | kpfleming | 2008-03-26 14:51:24 -0500 (Wed, 26 Mar 2008) | 10 lines Merged revisions 111125 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r111125 | kpfleming | 2008-03-26 14:49:30 -0500 (Wed, 26 Mar 2008) | 2 lines update UPGRADE notes to document usage of the script ........ ................ ................ 2008-03-26 19:41 +0000 [r111124] Mark Michelson * apps/app_voicemail.c, /: Merged revisions 111123 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r111123 | mmichelson | 2008-03-26 14:39:23 -0500 (Wed, 26 Mar 2008) | 12 lines Merged revisions 111121 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r111121 | mmichelson | 2008-03-26 14:37:36 -0500 (Wed, 26 Mar 2008) | 4 lines This code change is made just for clarification. It does exactly the same thing as before. It just doesn't look as wrong. ........ ................ 2008-03-26 19:27 +0000 [r111072] Mark Michelson * apps/app_voicemail.c, /: Merged revisions 111067 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r111067 | mmichelson | 2008-03-26 14:26:23 -0500 (Wed, 26 Mar 2008) | 17 lines Merged revisions 111049 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r111049 | mmichelson | 2008-03-26 14:22:16 -0500 (Wed, 26 Mar 2008) | 9 lines Add a lock to the vm_state structure and use the lock around mail_open calls to prevent concurrent access of the same mailstream. This, along with trunk's ability to configure TCP timeouts for IMAP storage will help to prevent crashes and hangs when using voicemail with IMAP storage. (closes issue #10487) Reported by: ewilhelmsen ........ ................ 2008-03-26 19:08 +0000 [r111026] Kevin P. Fleming * codecs/ilbc, /, contrib/scripts/get_ilbc_source.sh (added): Merged revisions 111025 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r111025 | kpfleming | 2008-03-26 14:08:00 -0500 (Wed, 26 Mar 2008) | 18 lines Merged revisions 111024 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r111024 | kpfleming | 2008-03-26 14:06:56 -0500 (Wed, 26 Mar 2008) | 10 lines Merged revisions 111019 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r111019 | kpfleming | 2008-03-26 13:58:37 -0500 (Wed, 26 Mar 2008) | 2 lines add a script to make getting the iLBC source code simple for end users ........ ................ ................ 2008-03-26 19:06 +0000 [r111018-111023] Joshua Colp * /, channels/chan_sip.c: Merged revisions 111021 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r111021 | file | 2008-03-26 16:05:42 -0300 (Wed, 26 Mar 2008) | 12 lines Merged revisions 111020 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r111020 | file | 2008-03-26 16:04:35 -0300 (Wed, 26 Mar 2008) | 4 lines If we are requested to authenticate a reinvite make sure that it contains T38 SDP if need be. (closes issue #11995) Reported by: fall ........ ................ * /, channels/chan_iax2.c: Merged revisions 111017 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r111017 | file | 2008-03-26 15:42:52 -0300 (Wed, 26 Mar 2008) | 12 lines Merged revisions 110628 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r110628 | file | 2008-03-25 11:37:35 -0300 (Tue, 25 Mar 2008) | 4 lines Add an option (transmit_silence) which transmits silence during both Record() and DTMF generation. The reason this is an option is that in order to transmit silence we have to setup a translation path. This may not be needed/wanted in all cases. (closes issue #10058) Reported by: tracinet ........ ................ 2008-03-26 17:44 +0000 [r110964] Kevin P. Fleming * /, UPGRADE.txt: Merged revisions 110963 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r110963 | kpfleming | 2008-03-26 12:44:09 -0500 (Wed, 26 Mar 2008) | 10 lines Merged revisions 110962 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r110962 | kpfleming | 2008-03-26 12:43:02 -0500 (Wed, 26 Mar 2008) | 2 lines add note that the user will need to enable codec_ilbc to get it to build ........ ................ 2008-03-26 17:35 +0000 [r110959] Donny Kavanagh * /, doc/snmp.txt: Merged revisions 110911 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r110911 | juggie | 2008-03-26 13:24:54 -0400 (Wed, 26 Mar 2008) | 8 lines update documentation to reflect the changes in the way configure detects net-snmp. (closes issue #12067) Reported by: juggie Patches: 12067_snmp_doc.patch uploaded by juggie (license 24) Tested by: juggie ........ 2008-03-26 17:15 +0000 [r110882] Kevin P. Fleming * codecs/ilbc/constants.h (removed), codecs/ilbc/iLBC_decode.h (removed), codecs/ilbc/iCBSearch.c (removed), codecs/Makefile, codecs/ilbc/filter.c (removed), codecs/ilbc/hpInput.c (removed), codecs/ilbc/gainquant.c (removed), codecs/ilbc/hpOutput.c (removed), codecs/ilbc/iCBSearch.h (removed), codecs/ilbc/filter.h (removed), codecs/ilbc/hpInput.h (removed), codecs/ilbc/gainquant.h (removed), codecs/ilbc/LPCencode.c (removed), codecs/ilbc/hpOutput.h (removed), codecs/ilbc/StateSearchW.c (removed), codecs/codec_ilbc.c, codecs/ilbc/LPCencode.h (removed), codecs/ilbc/iCBConstruct.c (removed), codecs/ilbc/StateSearchW.h (removed), codecs/ilbc/syntFilter.c (removed), /, codecs/ilbc/iCBConstruct.h (removed), codecs/ilbc/syntFilter.h (removed), codecs/ilbc/packing.c (removed), codecs/ilbc/StateConstructW.c (removed), codecs/ilbc/packing.h (removed), codecs/ilbc/libilbc.vcproj (removed), codecs/ilbc/StateConstructW.h (removed), codecs/ilbc/LPCdecode.c (removed), codecs/ilbc/getCBvec.c (removed), codecs/ilbc/enhancer.c (removed), codecs/ilbc/lsf.c (removed), codecs/ilbc/iLBC_encode.c (removed), codecs/ilbc/getCBvec.h (removed), codecs/ilbc/LPCdecode.h (removed), codecs/ilbc/enhancer.h (removed), codecs/ilbc/FrameClassify.c (removed), codecs/ilbc/iLBC_define.h (removed), codecs/ilbc/lsf.h (removed), codecs/ilbc/iLBC_encode.h (removed), codecs/ilbc/FrameClassify.h (removed), codecs/ilbc/helpfun.c (removed), codecs/ilbc/doCPLC.c (removed), codecs/ilbc/anaFilter.c (removed), codecs/ilbc/helpfun.h (removed), codecs/ilbc/createCB.c (removed), codecs/ilbc/doCPLC.h (removed), codecs/ilbc/anaFilter.h (removed), UPGRADE.txt, codecs/ilbc/constants.c (removed), codecs/ilbc/iLBC_decode.c (removed), codecs/ilbc/createCB.h (removed), CHANGES: Merged revisions 110881 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r110881 | kpfleming | 2008-03-26 10:10:28 -0700 (Wed, 26 Mar 2008) | 18 lines Merged revisions 110880 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r110880 | kpfleming | 2008-03-26 09:42:35 -0700 (Wed, 26 Mar 2008) | 10 lines Merged revisions 110869 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r110869 | kpfleming | 2008-03-26 08:53:46 -0700 (Wed, 26 Mar 2008) | 2 lines due to licensing restrictions, we cannot distribute the source code for iLBC encoding and decoding... so remove it, and add instructions on how the user can obtain it themselves ........ ................ ................ 2008-03-26 15:33 +0000 [r110866-110868] Joshua Colp * /: Merged revisions 110726 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r110726 | jpeeler | 2008-03-25 17:02:57 -0300 (Tue, 25 Mar 2008) | 2 lines This one line change makes an if inside a for loop (in realtime_peer) check all the ast_variables the loop was intending to test rather than just the first one. ........ 2008-03-26 00:03 +0000 [r110832] Mark Michelson * main/manager.c, /: Merged revisions 110831 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r110831 | mmichelson | 2008-03-25 19:02:31 -0500 (Tue, 25 Mar 2008) | 6 lines This ensures that the manager interface is not enabled by default. Prior to this change, it was possible to start Asterisk with the manager interface enabled, then either comment out the enabled option or make manager.conf unopenable and the manager interface would still be enabled. ........ 2008-03-25 22:52 +0000 [r110781] Jason Parker * cdr/cdr_custom.c, /: Merged revisions 110780 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r110780 | qwell | 2008-03-25 17:51:55 -0500 (Tue, 25 Mar 2008) | 14 lines Merged revisions 110779 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r110779 | qwell | 2008-03-25 17:51:17 -0500 (Tue, 25 Mar 2008) | 6 lines Make file access in cdr_custom similar to cdr_csv. Fixes issue #12268. Patch borrowed from r82344 ........ ................ 2008-03-25 22:11 +0000 [r110778] Jeff Peeler * channels/chan_sip.c: This one line change makes an if inside a for loop (in realtime_peer) check all the ast_variables the loop was intending to test rather than just the first one. 2008-03-25 17:47 +0000 [r110690-110692] Tilghman Lesher * configs/extensions.conf.sample, /, configs/voicemail.conf.sample: Merged revisions 110691 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r110691 | tilghman | 2008-03-25 12:46:34 -0500 (Tue, 25 Mar 2008) | 6 lines Update sample configurations to make virtual hosting more obvious. (closes issue #11969) Reported by: pprindeville Patches: acme-virtualpbx.1.6.patch uploaded by pprindeville (license 347) ........ * configs/extensions.conf.sample, /: Merged revisions 110689 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r110689 | tilghman | 2008-03-25 12:40:28 -0500 (Tue, 25 Mar 2008) | 6 lines Update the sample configuration, to use Macro less (since it's now deprecated). (closes issue #12293) Reported by: pprindeville Patches: bugid-0012293.1.6.patch uploaded by pprindeville (license 347) ........ 2008-03-25 15:43 +0000 [r110637-110638] Mark Michelson * channels/chan_sip.c: Oops. * /, channels/chan_sip.c: Merged revisions 110636 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r110636 | mmichelson | 2008-03-25 10:41:33 -0500 (Tue, 25 Mar 2008) | 15 lines Merged revisions 110635 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r110635 | mmichelson | 2008-03-25 10:40:33 -0500 (Tue, 25 Mar 2008) | 7 lines When reverting a commit, I accidentally left in this bit which was an experiment to see what would happen. It passed the compile test, and I didn't notice I had left this change in too. So this is a revert of a revert...sort of. ........ ................ 2008-03-25 15:39 +0000 [r110630-110634] Joshua Colp * include/asterisk/options.h, main/asterisk.c, Makefile, /, main/app.c: Merged revisions 110629 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r110629 | file | 2008-03-25 11:39:45 -0300 (Tue, 25 Mar 2008) | 12 lines Merged revisions 110628 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r110628 | file | 2008-03-25 11:37:35 -0300 (Tue, 25 Mar 2008) | 4 lines Add an option (transmit_silence) which transmits silence during both Record() and DTMF generation. The reason this is an option is that in order to transmit silence we have to setup a translation path. This may not be needed/wanted in all cases. (closes issue #10058) Reported by: tracinet ........ ................ 2008-03-24 20:14 +0000 [r110620-110622] Mark Michelson * /, channels/chan_sip.c: Merged revisions 110619 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r110619 | mmichelson | 2008-03-24 14:19:37 -0500 (Mon, 24 Mar 2008) | 23 lines Merged revisions 110618 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r110618 | mmichelson | 2008-03-24 14:17:41 -0500 (Mon, 24 Mar 2008) | 15 lines This is a revert for revision 108288. The reason is that that revision was not for an actual bug fix per se, and so it really should not have been in 1.4 in the first place. Plus, people who compile with DO_CRASH are more likely to encounter a crash due to this change. While I think the usage of DO_CRASH in ast_sched_del is a bit absurd, this sort of change is beyond the scope of 1.4 and should be done instead in a developer branch based on trunk so that all scheduler functions are fixed at once. I also am reverting the change to trunk and 1.6 since they also suffer from the DO_CRASH potential. (closes issue #12272) Reported by: qq12345 ........ ................ 2008-03-24 17:36 +0000 [r110616] Russell Bryant * /, channels/chan_iax2.c: Merged revisions 110615 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r110615 | russell | 2008-03-24 12:36:04 -0500 (Mon, 24 Mar 2008) | 10 lines Merged revisions 110614 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r110614 | russell | 2008-03-24 12:34:56 -0500 (Mon, 24 Mar 2008) | 2 lines Turn a NOTICE into a DEBUG message. ........ ................ 2008-03-24 15:29 +0000 [r110611] Joshua Colp * /, channels/chan_sip.c: Merged revisions 110610 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r110610 | file | 2008-03-24 12:28:25 -0300 (Mon, 24 Mar 2008) | 6 lines Only print out the set_address_from_contact host verbose message if debugging is enabled on the dialog. (closes issue #12280) Reported by: rjain Patches: chan_sip.c.diff uploaded by rjain (license 226) ........ 2008-03-21 21:52 +0000 [r110579] Jason Parker * /, sounds/Makefile: Merged revisions 110578 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r110578 | qwell | 2008-03-21 16:52:06 -0500 (Fri, 21 Mar 2008) | 1 line Update to 1.4.11 core sounds. ........ 2008-03-21 15:25 +0000 [r110501] Russell Bryant * /, configs/sip.conf.sample, CHANGES: Merged revisions 110499 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r110499 | russell | 2008-03-21 10:24:43 -0500 (Fri, 21 Mar 2008) | 3 lines Note that the TCP and TLS support is currently considered experimental and is subject to change while we work out the remaining issues. ........ 2008-03-21 14:36 +0000 [r110476] Jason Parker * /, codecs/gsm/Makefile: Merged revisions 110475 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r110475 | qwell | 2008-03-21 09:36:17 -0500 (Fri, 21 Mar 2008) | 15 lines Merged revisions 110474 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r110474 | qwell | 2008-03-21 09:32:52 -0500 (Fri, 21 Mar 2008) | 7 lines Don't attempt to do optimizations of gsm on mips platforms either. (closes issue #12270) Reported by: zandbelt Patches: 026-gsm-mips.patch uploaded by zandbelt (license 33) ........ ................ 2008-03-20 23:14 +0000 [r110304-110397] Russell Bryant * main/autoservice.c, /: Merged revisions 110396 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r110396 | russell | 2008-03-20 18:14:13 -0500 (Thu, 20 Mar 2008) | 17 lines Merged revisions 110395 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r110395 | russell | 2008-03-20 18:13:56 -0500 (Thu, 20 Mar 2008) | 9 lines Shorten the ast_waitfor() timeout from 500 ms to 50 ms in the autoservice thread. This really should not make a difference except in very rare cases. That case would be that all of the channels in autoservice are not generating any frames. In that case, this change reduces the potential amount of time that a thread waits in ast_autoservice_stop() for the autoservice thread to wrap back around to the beginning of its loop. (closes issue #12266, reported by dimas) ........ ................ * codecs/codec_g722.c, /: Merged revisions 110339 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r110339 | russell | 2008-03-20 17:02:20 -0500 (Thu, 20 Mar 2008) | 3 lines Use the correct buffer for g722tolin16_sample. This shouldn't have caused any problems, but Qwell noticed the typo here. ........ * /, channels/chan_sip.c, channels/chan_iax2.c: Merged revisions 110337 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r110337 | russell | 2008-03-20 16:55:50 -0500 (Thu, 20 Mar 2008) | 22 lines Merged revisions 110336 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r110336 | russell | 2008-03-20 16:54:58 -0500 (Thu, 20 Mar 2008) | 14 lines Merged revisions 110335 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r110335 | russell | 2008-03-20 16:53:27 -0500 (Thu, 20 Mar 2008) | 6 lines Fix some very broken code that was introduced in 1.2.26 as a part of the security fix. The dnsmgr is not appropriate here. The dnsmgr takes a pointer to an address structure that a background thread continuously updates. However, in these cases, a stack variable was passed. That means that the dnsmgr thread would be continuously writing to bogus memory. ........ ................ ................ * /, main/file.c: Merged revisions 110303 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r110303 | russell | 2008-03-20 15:08:26 -0500 (Thu, 20 Mar 2008) | 8 lines Fix a bug when using zaptel timing for playing back files that have a sample rate other than 8 kHz. The issue here is that format modules give a "whennext" sample value, which is used to calculate when to set a timer for to retrieve the next frame. However, the zaptel timer operates on 8 kHz samples, so this must be taken into account. (another part of issue #12164, reported by milazzo and jsmith, patch by me) ........ 2008-03-20 18:02 +0000 [r110273] Mark Michelson * main/dial.c, /: Merged revisions 110272 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r110272 | mmichelson | 2008-03-20 13:01:36 -0500 (Thu, 20 Mar 2008) | 3 lines Add missing unlock ........ 2008-03-20 17:45 +0000 [r110269-110271] Russell Bryant * main/channel.c, /, res/res_musiconhold.c: Merged revisions 110268 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r110268 | russell | 2008-03-20 12:41:22 -0500 (Thu, 20 Mar 2008) | 27 lines Add some fixes that I made in regards to wideband codec handling to get G.722 music on hold working for me. (issue #12164, reported by milazzo and jsmith, patches by me) res/res_musiconhold.c: - I moved a single line so that the sample queue update happened before ast_write(). The reason that this was a bug is that the G.722 frame originally says it has 320 samples in it (which is correct). However, when the frame is written to a channel that uses RTP, main/rtp.c modifies the frame to cut the number of samples in half before it sends it on the wire. This is to account for the stupid incorrect G.722 spec that makes it so we have to lie about the number of samples with RTP. I should probably go and re-work the RTP code so it doesn't modify the frame so that a bug like this won't happen in the future. However, this change to MOH is harmless. main/channel.c: - I made two fixes in regards to generator timing. Generators use samples for timing. However, this code assumed 8 kHz samples. In one case, it was a hard coded 160 samples, that is now written as the sample rate / 50. The other place was dealing with timing a generator based on frames coming from the other direction. However, that would have only worked if the sample rates for the formats in both directions were the same. The code now takes into account that the sample rates may differ, and scales the generator samples accordingly. ........ 2008-03-19 23:00 +0000 [r110165] Russell Bryant * /, apps/app_meetme.c: Merged revisions 110164 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r110164 | russell | 2008-03-19 17:58:33 -0500 (Wed, 19 Mar 2008) | 13 lines Merged revisions 110163 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r110163 | russell | 2008-03-19 17:57:59 -0500 (Wed, 19 Mar 2008) | 5 lines Fix a bug where when calls on the trunk side hang up while on hold, the state is not properly reflected. (closes issue #11990, reported by anakaoka, patched by me) ........ ................ 2008-03-19 21:06 +0000 [r110088] Jeff Peeler * /: marking rev 110087 from trunk as not applying 2008-03-19 20:37 +0000 [r110085] Mark Michelson * apps/app_chanspy.c, /: Merged revisions 110084 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r110084 | mmichelson | 2008-03-19 15:34:13 -0500 (Wed, 19 Mar 2008) | 12 lines Merged revisions 110083 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r110083 | mmichelson | 2008-03-19 15:33:03 -0500 (Wed, 19 Mar 2008) | 4 lines Add a missing unlock in the case that memory allocation fails in app_chanspy. Thanks to Russell for confirming that this was an issue. ........ ................ 2008-03-19 19:14 +0000 [r110037] Joshua Colp * /, res/res_musiconhold.c: Merged revisions 110036 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r110036 | file | 2008-03-19 16:13:39 -0300 (Wed, 19 Mar 2008) | 12 lines Merged revisions 110035 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r110035 | file | 2008-03-19 16:11:33 -0300 (Wed, 19 Mar 2008) | 4 lines Add sanity checking for position resuming. We *have* to make sure that the position does not exceed the total number of files present, and we have to make sure that the position's filename is the same as previous. These values can change if a music class is reloaded and give unpredictable behavior. (closes issue #11663) Reported by: junky ........ ................ 2008-03-19 19:00 +0000 [r110024-110032] Russell Bryant * Makefile, build_tools/cflags.xml, build_tools/cflags-devmode.xml (added), /: Merged revisions 109974 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r109974 | qwell | 2008-03-19 12:15:14 -0500 (Wed, 19 Mar 2008) | 13 lines Merged revisions 109973 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r109973 | qwell | 2008-03-19 12:12:52 -0500 (Wed, 19 Mar 2008) | 5 lines People report bugs about Asterisk crashing with DO_CRASH enabled was getting a little silly... Now we only show certain cflags when you run configure with --enable-dev-mode (corresponding menuselect change to follow) ........ ................ 2008-03-19 18:26 +0000 [r109971-110021] Joshua Colp * main/rtp.c, /: Merged revisions 110020 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r110020 | file | 2008-03-19 15:25:33 -0300 (Wed, 19 Mar 2008) | 14 lines Merged revisions 110019 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r110019 | file | 2008-03-19 15:20:28 -0300 (Wed, 19 Mar 2008) | 6 lines Make sure that the mark bit does not incorrectly cause video frame timestamps to be calculated as if they are audio frames. (closes issue #11429) Reported by: sperreault Patches: 11429-frametype.diff uploaded by qwell (license 4) ........ ................ 2008-03-19 16:46 +0000 [r109969] Steve Murphy * main/config.c, /: Merged revisions 109942 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r109942 | murf | 2008-03-19 10:24:51 -0600 (Wed, 19 Mar 2008) | 80 lines Merged revisions 109908 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r109908 | murf | 2008-03-19 09:41:13 -0600 (Wed, 19 Mar 2008) | 72 lines (closes issue #11442) Reported by: tzafrir Patches: 11442.patch uploaded by murf (license 17) Tested by: murf I didn't give tzafrir very much time to test this, but if he does still have remaining issues, he is welcome to re-open this bug, and we'll do what is called for. I reproduced the problem, and tested the fix, so I hope I am not jumping by just going ahead and committing the fix. The problem was with what file_save does with templates; firstly, it tended to print out multiple options: [my_category](!)(templateref) instead of [my_category](!,templateref) which is fixed by this patch. Nextly, the code to suppress output of duplicate declarations that would occur because the reader copies inherited declarations down the hierarchy, was not working. Thus: [master-template](!) mastervar = bar [template](!,master-template) tvar = value [cat](template) catvar = val would be rewritten as: ;! ;! Automatically generated configuration file ;! Filename: experiment.conf (/etc/asterisk/experiment.conf) ;! Generator: Manager ;! Creation Date: Tue Mar 18 23:17:46 2008 ;! [master-template](!) mastervar = bar [template](!,master-template) mastervar = bar tvar = value [cat](template) mastervar = bar tvar = value catvar = val This has been fixed. Since the config reader 'explodes' inherited vars into the category, users may, in certain circumstances, see output different from what they originally entered, but it should be both correct and equivalent. ........ ................ 2008-03-19 04:06 +0000 [r109834-109840] Russell Bryant * /, main/utils.c: Merged revisions 109839 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r109839 | russell | 2008-03-18 23:06:31 -0500 (Tue, 18 Mar 2008) | 10 lines Merged revisions 109838 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r109838 | russell | 2008-03-18 23:06:05 -0500 (Tue, 18 Mar 2008) | 2 lines Tweak spacing in a recent change because I'm very picky. ........ ................ * apps/app_chanspy.c, /: Merged revisions 109764 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r109764 | russell | 2008-03-18 17:36:02 -0500 (Tue, 18 Mar 2008) | 11 lines Merged revisions 109763 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r109763 | russell | 2008-03-18 17:34:42 -0500 (Tue, 18 Mar 2008) | 3 lines Fix one place where the chanspy datastore isn't removed from a channel. (issue #12243, reported by atis, patch by me) ........ ................ 2008-03-18 23:23 +0000 [r109779] Tilghman Lesher * /, configs/res_ldap.conf.sample, res/res_config_ldap.c: Merged revisions 109775 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r109775 | tilghman | 2008-03-18 18:22:25 -0500 (Tue, 18 Mar 2008) | 3 lines Change back to using ldap_initialize() and let the user specify a URL directly, instead of trying to piece it together, badly. ........ 2008-03-18 21:03 +0000 [r109716] Mark Michelson * /, apps/app_queue.c: Merged revisions 109714 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r109714 | mmichelson | 2008-03-18 15:59:02 -0500 (Tue, 18 Mar 2008) | 20 lines Merged revisions 109713 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r109713 | mmichelson | 2008-03-18 15:52:15 -0500 (Tue, 18 Mar 2008) | 12 lines This patch makes it so that all queue member status changes are handled through device state code. This removes several problems people were seeing where their queue members would get into an "unknown" state. Huge props go to atis on this one since he was the one who found the code section that was causing the problem and proposed the solution. I just wrote what he suggested :) (closes issue #12127) Reported by: atis Patches: 12127v3.patch uploaded by putnopvut (license 60) Tested by: atis, jvandal ........ ................ 2008-03-18 20:14 +0000 [r109684] Tilghman Lesher * /, res/res_config_ldap.c: Merged revisions 109683 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r109683 | tilghman | 2008-03-18 15:13:40 -0500 (Tue, 18 Mar 2008) | 4 lines Set protocol version, port number correctly. (closes issue #12211, closes issue #12209) Reported by: sylvain ........ 2008-03-18 19:24 +0000 [r109654] Jason Parker * /, codecs/log2comp.h: Merged revisions 109651 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r109651 | qwell | 2008-03-18 14:24:15 -0500 (Tue, 18 Mar 2008) | 15 lines Merged revisions 109648 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r109648 | qwell | 2008-03-18 14:23:44 -0500 (Tue, 18 Mar 2008) | 7 lines Allow codecs that use log2comp (g726) to compile correctly on x86 with gcc4 optimizations. (closes issue #12253) Reported by: fossil Patches: log2comp.patch uploaded by fossil (license 140) ........ ................ 2008-03-18 19:00 +0000 [r109546-109622] Mark Michelson * /, channels/chan_agent.c: Merged revisions 109576 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r109576 | mmichelson | 2008-03-18 12:59:18 -0500 (Tue, 18 Mar 2008) | 14 lines Merged revisions 109575 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r109575 | mmichelson | 2008-03-18 12:58:11 -0500 (Tue, 18 Mar 2008) | 6 lines Make sure an agent doesn't try to send dtmf to a NULL channel closes issue #12242 Reported by Yourname ........ ................ * include/asterisk/astmm.h, /: Merged revisions 109545 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r109545 | mmichelson | 2008-03-18 12:00:53 -0500 (Tue, 18 Mar 2008) | 3 lines Add format attribute to printf-style functions in astmm.h ........ 2008-03-18 Russell Bryant * Asterisk 1.6.0-beta6 released. 2008-03-18 17:01 +0000 [r109546] Mark Michelson * include/asterisk/astmm.h, /: Merged revisions 109545 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r109545 | mmichelson | 2008-03-18 12:00:53 -0500 (Tue, 18 Mar 2008) | 3 lines Add format attribute to printf-style functions in astmm.h ........ 2008-03-18 16:26 +0000 [r109487] Kevin P. Fleming * channels/misdn/isdn_lib.c, channels/misdn/isdn_lib.h, channels/chan_misdn.c, /: Merged revisions 109475 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r109475 | kpfleming | 2008-03-18 11:23:05 -0500 (Tue, 18 Mar 2008) | 2 lines fix up various warnings found via the addition of format string checking... some of these were really, really bad code ........ 2008-03-18 15:58 +0000 [r109454-109459] Russell Bryant * Makefile, channels/chan_misdn.c, include/asterisk/strings.h, res/res_indications.c, utils/extconf.c, main/asterisk.c, apps/app_voicemail.c, utils/check_expr.c, cdr/cdr_sqlite3_custom.c, apps/app_meetme.c, /, res/res_phoneprov.c, main/utils.c, channels/chan_iax2.c, utils/frame.c, main/cli.c, funcs/func_enum.c, main/manager.c, include/asterisk/astobj.h, res/res_agi.c, main/features.c, apps/app_minivm.c, res/res_realtime.c, res/res_config_ldap.c, include/asterisk/utils.h, channels/chan_sip.c, apps/app_festival.c, main/translate.c, main/jitterbuf.c, utils/astman.c, include/jitterbuf.h, apps/app_queue.c: Merged revisions 109447 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r109447 | twilson | 2008-03-18 10:43:34 -0500 (Tue, 18 Mar 2008) | 3 lines Go through and fix a bunch of places where character strings were being interpreted as format strings. Most of these changes are solely to make compiling with -Wsecurity and -Wformat=2 happy, and were not actual problems, per se. I also added format attributes to any printf wrapper functions I found that didn't have them. -Wsecurity and -Wmissing-format-attribute added to --enable-dev-mode. ........ * configs/sip_notify.conf.sample, /: Merged revisions 109111 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r109111 | qwell | 2008-03-17 11:37:31 -0500 (Mon, 17 Mar 2008) | 10 lines Add sample events for aastra phones. aastra-check-cfg is the same as the other check-cfg entries, and aastra-xml is to load a pre-configured xml script. (closes issue #12229) Reported by: gowen72 Patches: aastra.patch uploaded by gowen72 (license 432) ........ 2008-03-18 15:50 +0000 [r109453] Kevin P. Fleming * /, configure, include/asterisk/autoconfig.h.in, acinclude.m4: Merged revisions 109451 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r109451 | kpfleming | 2008-03-18 10:50:29 -0500 (Tue, 18 Mar 2008) | 2 lines ensure that dependencies on AST_C_DEFINE_CHECK symbols work properly ........ 2008-03-18 15:50 +0000 [r109448-109452] Russell Bryant * main/dial.c, /: Merged revisions 108962 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r108962 | mvanbaak | 2008-03-16 16:50:58 -0500 (Sun, 16 Mar 2008) | 15 lines Merged revisions 108961 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r108961 | mvanbaak | 2008-03-16 22:47:10 +0100 (Sun, 16 Mar 2008) | 7 lines add missing break to case AST_CONTROL_SRCUPDATE (closes issue #12228) Reported by: andrew Patches: SRC.patch uploaded by andrew (license 240) ........ ................ 2008-03-18 15:16 +0000 [r109398] Joshua Colp * main/manager.c, /, main/logger.c: Merged revisions 109396 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r109396 | file | 2008-03-18 12:13:07 -0300 (Tue, 18 Mar 2008) | 3 lines Make sure values are interpreted as character strings and not format strings. (AST-2008-004) ........ 2008-03-18 15:14 +0000 [r109397] Steve Murphy * pbx/ael/ael-test/ael-ntest23 (added), pbx/ael/ael-test/ael-ntest23/t1/a.ael, pbx/ael/ael-test/ael-ntest23/t1/b.ael, pbx/ael/ael-test/ael-ntest23/t1/c.ael, pbx/ael/ael-test/ael-ntest23/t2/d.ael, pbx/ael/ael-test/ael-ntest23/t2/e.ael, pbx/ael/ael-test/ael-ntest23/t2/f.ael, res/ael/ael_lex.c, pbx/ael/ael-test/ref.ael-ntest23 (added), pbx/ael/ael-test/ael-ntest23/t3/g.ael, pbx/ael/ael-test/ael-ntest23/t3/h.ael, pbx/ael/ael-test/ael-ntest23/t3/i.ael, res/ael/ael.flex, pbx/ael/ael-test/ael-ntest23/t3/j.ael, pbx/ael/ael-test/ael-ntest23/qq.ael, pbx/ael/ael-test/ael-ntest23/t1, pbx/ael/ael-test/ael-ntest23/t2, pbx/ael/ael-test/ael-ntest23/t3, /, pbx/ael/ael-test/ael-ntest23/extensions.ael: Merged revisions 109357 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r109357 | murf | 2008-03-18 08:09:50 -0600 (Tue, 18 Mar 2008) | 25 lines Merged revisions 109309 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r109309 | murf | 2008-03-18 00:37:15 -0600 (Tue, 18 Mar 2008) | 17 lines (closes issue #11903) Reported by: atis Many thanks to atis for spotting this problem and reporting it. The fix was to straighten out how items are placed on and removed from the file stack. Regressions as well as the provided test case helped to straighten out all code paths. valgrind was used to make sure all memory allocated was freed. Sorry for not solving this earlier. I got distracted. Added the ntest23 regression test, which is mainly a copy of ntest22, but with a few juicy errors thrown in, to replicate the kind of error that atis spotted. ........ ................ 2008-03-18 15:11 +0000 [r109395] Jason Parker * /, channels/chan_sip.c: Merged revisions 109389 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r109389 | qwell | 2008-03-18 10:07:04 -0500 (Tue, 18 Mar 2008) | 3 lines Do not return with a successful authentication if the From header ends up empty. (AST-2008-003) ........ 2008-03-18 15:09 +0000 [r109392] Joshua Colp * main/rtp.c, /, channels/chan_sip.c: Merged revisions 109390 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r109390 | file | 2008-03-18 12:08:09 -0300 (Tue, 18 Mar 2008) | 11 lines Merged revisions 109386 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r109386 | file | 2008-03-18 11:58:39 -0300 (Tue, 18 Mar 2008) | 3 lines Put a maximum limit on the number of payloads accepted, and also make sure a given payload does not exceed our maximum value. (AST-2008-002) ........ ................ 2008-03-18 00:40 +0000 [r109283] Sean Bright * /, configure, configure.ac: Merged revisions 109282 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r109282 | seanbright | 2008-03-17 20:28:39 -0400 (Mon, 17 Mar 2008) | 1 line Fix a typo ........ 2008-03-17 22:24 +0000 [r109254] Terry Wilson * build_tools/cflags.xml, /, build_tools/menuselect-deps.in, configure, include/asterisk/autoconfig.h.in, main/Makefile, configure.ac, main/http.c, main/minimime (removed), build_tools/make_buildopts_h, makeopts.in: Merged revisions 109229 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r109229 | twilson | 2008-03-17 17:10:06 -0500 (Mon, 17 Mar 2008) | 5 lines Replace minimime with superior GMime library so that the entire contents of an http post are not read into memory. This does introduce a dependency on the GMime library for handling HTTP POSTs, but it is available in most distros. If the library is present, then the compile flag for ENABLE_UPLOADS is enabled by default in menuselect. ........ 2008-03-17 22:07 +0000 [r109228] Mark Michelson * /, main/utils.c: Merged revisions 109227 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r109227 | mmichelson | 2008-03-17 17:06:44 -0500 (Mon, 17 Mar 2008) | 20 lines Merged revisions 109226 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r109226 | mmichelson | 2008-03-17 17:05:49 -0500 (Mon, 17 Mar 2008) | 12 lines Fix a logic flaw in the code that stores lock info which is displayed via the "core show locks" command. The idea behind this section of code was to remove the previous lock from the list if it was a trylock that had failed. Unfortunately, instead of checking the status of the previous lock, we were referencing the index immediately following the previous lock in the lock_info->locks array. The result of this problem, under the right circumstances, was that the lock which we currently in the process of attempting to acquire could "overwrite" the previous lock which was acquired. While this does not in any way affect typical operation, it *could* lead to misleading "core show locks" output. ........ ................ 2008-03-17 18:11 +0000 [r109175] Michiel van Baak * /, channels/chan_skinny.c: Merged revisions 109168 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r109168 | mvanbaak | 2008-03-17 18:43:46 +0100 (Mon, 17 Mar 2008) | 11 lines Update the directory of placed calls on skinny phones when dialing a channel that does not provide progress (analog ZAP lines) The phone does handle the double update on calls to channels that do provide progress and wont insert duplicate items (closes issue #12239) Reported by: DEA Patches: chan_skinny-call-log.txt uploaded by DEA (license 3) ........ 2008-03-17 17:42 +0000 [r109167] Kevin P. Fleming * Makefile, /, configure, configure.ac, acinclude.m4: Merged revisions 109166 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r109166 | kpfleming | 2008-03-17 12:31:46 -0500 (Mon, 17 Mar 2008) | 3 lines don't define Zaptel features as libraries, they aren't, and we don't want '--with-zaptel-' configure options for them also some minor cleanups ........ 2008-03-17 16:47 +0000 [r109109-109114] Joshua Colp * /, channels/chan_sip.c: Merged revisions 109108 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r109108 | file | 2008-03-17 13:26:36 -0300 (Mon, 17 Mar 2008) | 12 lines Merged revisions 109107 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r109107 | file | 2008-03-17 13:24:29 -0300 (Mon, 17 Mar 2008) | 4 lines 200 OKs in response to a reinvite need to be sent reliably. If the remote side does not receive one the dialog will be torn down. (closes issue #12208) Reported by: atrash ........ ................ 2008-03-17 14:21 +0000 [r109027] Mark Michelson * apps/app_chanspy.c, /: Merged revisions 109024 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r109024 | mmichelson | 2008-03-17 09:21:14 -0500 (Mon, 17 Mar 2008) | 14 lines Merged revisions 109012 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r109012 | mmichelson | 2008-03-17 09:18:26 -0500 (Mon, 17 Mar 2008) | 6 lines Make sure that we release the lock on the spyee channel if the spyee or spy has hung up (closes issue #12232) Reported by: atis ........ ................ 2008-03-16 17:56 +0000 [r108928-108930] Russell Bryant * apps/app_voicemail.c, /: Merged revisions 108927 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r108927 | russell | 2008-03-16 12:53:46 -0500 (Sun, 16 Mar 2008) | 7 lines Fix polling for mailbox changes in mailboxes that are not in the default vm context. (closes issue #12223) Reported by: DEA Patches: vm-polled-imap.txt uploaded by DEA (license 3) ........ 2008-03-15 16:21 +0000 [r108741-108895] Russell Bryant * Makefile, /: Merged revisions 108799 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r108799 | russell | 2008-03-14 15:14:06 -0500 (Fri, 14 Mar 2008) | 8 lines Make sure configure is run before menuselect on a clean checkout (closes issue #12197) Reported by: juggie Patches: 12197.diff uploaded by juggie (license 24) ........ * channels/chan_oss.c, /: Merged revisions 108797 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r108797 | russell | 2008-03-14 15:09:37 -0500 (Fri, 14 Mar 2008) | 13 lines Merged revisions 108796 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r108796 | russell | 2008-03-14 15:09:22 -0500 (Fri, 14 Mar 2008) | 5 lines Fix a channel name issue. chan_oss registers the "Console" channel type, but it created channels with an "OSS" prefix. (closes issue #12194, reported by davidw, patched by me) ........ ................ * contrib/init.d/rc.suse.asterisk, /: Merged revisions 108793 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r108793 | russell | 2008-03-14 15:04:56 -0500 (Fri, 14 Mar 2008) | 12 lines Merged revisions 108792 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r108792 | russell | 2008-03-14 15:04:35 -0500 (Fri, 14 Mar 2008) | 4 lines Update the SuSE init script to start networking before asterisk, as well. (closes issue #12200, reported by and change suggested by reinerotto) ........ ................ * /, configure, acinclude.m4: Merged revisions 108740 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r108740 | russell | 2008-03-14 12:05:11 -0500 (Fri, 14 Mar 2008) | 5 lines Do a link test in AST_EXT_TOOL_CHECK() to ensure we have all the required libs reported by the tool. (closes issue #12067, reported by Juggie, patched by me) ........ 2008-03-14 16:54 +0000 [r108739] Mark Michelson * /, channels/chan_sip.c: Merged revisions 108738 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r108738 | mmichelson | 2008-03-14 11:52:51 -0500 (Fri, 14 Mar 2008) | 41 lines Merged revisions 108737 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r108737 | mmichelson | 2008-03-14 11:44:08 -0500 (Fri, 14 Mar 2008) | 33 lines Fix a race condition in the SIP packet scheduler which could cause a crash. chan_sip uses the scheduler API in order to schedule retransmission of reliable packets (such as INVITES). If a retransmission of a packet is occurring, then the packet is removed from the scheduler and retrans_pkt is called. Meanwhile, if a response is received from the packet as previously transmitted, then when we ACK the response, we will remove the packet from the scheduler and free the packet. The problem is that both the ACK function and retrans_pkt attempt to acquire the same lock at the beginning of the function call. This means that if the ACK function acquires the lock first, then it will free the packet which retrans_pkt is about to read from and write to. The result is a crash. The solution: 1. If the ACK function fails to remove the packet from the scheduler and the retransmit id of the packet is not -1 (meaning that we have not reached the maximum number of retransmissions) then release the lock and yield so that retrans_pkt may acquire the lock and operate. 2. Make absolutely certain that the ACK function does not recursively lock the lock in question. If it does, then releasing the lock will do no good, since retrans_pkt will still be unable to acquire the lock. (closes issue #12098) Reported by: wegbert (closes issue #12089) Reported by: PTorres Patches: 12098-putnopvutv3.patch uploaded by putnopvut (license 60) Tested by: jvandal ........ ................ 2008-03-14 14:33 +0000 [r108684] Jason Parker * /, res/res_musiconhold.c: Merged revisions 108683 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r108683 | qwell | 2008-03-14 09:32:55 -0500 (Fri, 14 Mar 2008) | 12 lines Merged revisions 108682 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r108682 | qwell | 2008-03-14 09:29:05 -0500 (Fri, 14 Mar 2008) | 4 lines Fix a potential segfault if chan (or chan->music_state) is NULL. Closes issue #12210, credit to edantie for pointing this out. ........ ................ 2008-03-13 21:48 +0000 [r108587] Mark Michelson * main/manager.c, /: Merged revisions 108586 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r108586 | mmichelson | 2008-03-13 16:47:55 -0500 (Thu, 13 Mar 2008) | 3 lines Make this compile ........ 2008-03-13 21:41 +0000 [r108585] Russell Bryant * apps/app_chanspy.c, main/channel.c, /, include/asterisk/channel.h: Merged revisions 108584 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r108584 | russell | 2008-03-13 16:40:43 -0500 (Thu, 13 Mar 2008) | 19 lines Merged revisions 108583 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r108583 | russell | 2008-03-13 16:38:16 -0500 (Thu, 13 Mar 2008) | 11 lines Fix another issue that was causing crashes in chanspy. This introduces a new datastore callback, called chan_fixup(). The concept is exactly like the fixup callback that is used in the channel technology interface. This callback gets called when the owning channel changes due to a masquerade. Before this was introduced, if a masquerade happened on a channel being spyed on, the channel pointer in the datastore became invalid. (closes issue #12187) (reported by, and lots of testing from atis) (props to file for the help with ideas) ........ ................ 2008-03-13 21:31 +0000 [r108582] Mark Michelson * main/manager.c, /: Merged revisions 108529 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r108529 | mmichelson | 2008-03-13 15:59:00 -0500 (Thu, 13 Mar 2008) | 11 lines Fixing a potential buffer overflow in the manager command ModuleCheck. Though this overflow is exploitable remotely, we are NOT issuing a security advisory for this since in order to exploit the overflow, the attacker would have to establish an authenticated manager session AND have the system privilege. By gaining this privilege, the attacker already has more powerful weapons at his disposal than overflowing a buffer with a malformed manager header, so the vulnerability in this case really lies with the authentication method that allowed the attacker to gain the system privilege in the first place. ........ 2008-03-13 21:07 +0000 [r108347-108532] Russell Bryant * /, channels/chan_sip.c: Merged revisions 108531 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r108531 | russell | 2008-03-13 16:06:52 -0500 (Thu, 13 Mar 2008) | 18 lines Merged revisions 108530 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r108530 | russell | 2008-03-13 16:06:33 -0500 (Thu, 13 Mar 2008) | 10 lines Make a tweak that gets the LEDs on polycom phones to blink when an extension that has been subscribed to goes on hold. Otherwise, they just stay on like it does when an extension is in use. (closes issue #11263) Reported by: russell Patches: notify_hold.rev1.txt uploaded by russell (license 2) Tested by: russell ........ ................ * apps/app_voicemail.c, /: Merged revisions 108508 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r108508 | russell | 2008-03-13 15:35:28 -0500 (Thu, 13 Mar 2008) | 2 lines Fix a place where configuration values could cause an overflow of a buffer. ........ * /, apps/app_followme.c: Merged revisions 108472 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r108472 | russell | 2008-03-13 15:26:59 -0500 (Thu, 13 Mar 2008) | 12 lines Merged revisions 108469 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r108469 | russell | 2008-03-13 15:26:28 -0500 (Thu, 13 Mar 2008) | 4 lines Fix a couple uses of sprintf. The second one could actually cause an overflow of a stack buffer. It's not a security issue though, it only depends on your configuration. ........ ................ * /, main/features.c: Merged revisions 107465 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r107465 | file | 2008-03-11 10:05:17 -0500 (Tue, 11 Mar 2008) | 4 lines Clarify comment about masquerading and playback of the parking slot. (closes issue #12180) Reported by: davidw ........ * /, channels/chan_sip.c: Merged revisions 107157 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r107157 | file | 2008-03-10 15:00:21 -0500 (Mon, 10 Mar 2008) | 4 lines If we receive a 488 on a T38 request reinvite back to audio. As well reinvite across a bridge back to audio if one side doesn't negotiate to T38. (closes issue #8677) Reported by: alex-911 ........ * /: Merged revisions 106892 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r106892 | mattf | 2008-03-07 16:36:49 -0600 (Fri, 07 Mar 2008) | 1 line Make sure we don't start a call when we have already done so in response to a COT message ........ * /, main/editline/Makefile.in: Merged revisions 106843 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r106843 | qwell | 2008-03-07 16:15:20 -0600 (Fri, 07 Mar 2008) | 13 lines Merged revisions 106842 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r106842 | qwell | 2008-03-07 16:14:45 -0600 (Fri, 07 Mar 2008) | 5 lines Fix hardcoded grep in editline, were GNU grep is required. (closes issue #12124) Reported by: dmartin ........ ................ * include/asterisk/http.h, main/tcptls.c, main/manager.c, /, channels/chan_sip.c, res/res_phoneprov.c, main/http.c, include/asterisk/tcptls.h: Merged revisions 108295 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r108295 | russell | 2008-03-12 17:13:18 -0500 (Wed, 12 Mar 2008) | 3 lines Rename ast_tcptls_server_instance to session_instance, since this pertains to server and client usage. ........ * /, main/http.c: Merged revisions 108346 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r108346 | russell | 2008-03-12 17:49:26 -0500 (Wed, 12 Mar 2008) | 4 lines Make the default prefix empty, like it was in Asterisk 1.4. (closes issue #12198, reported by bkruse, patched by me) ........ 2008-03-12 22:10 +0000 [r108246-108294] Mark Michelson * /, channels/chan_sip.c: Merged revisions 108293 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r108293 | mmichelson | 2008-03-12 17:09:52 -0500 (Wed, 12 Mar 2008) | 3 lines Let's get this to compile ........ * /, channels/chan_sip.c: Merged revisions 108289 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r108289 | mmichelson | 2008-03-12 16:57:41 -0500 (Wed, 12 Mar 2008) | 22 lines Merged revisions 108288 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r108288 | mmichelson | 2008-03-12 16:53:46 -0500 (Wed, 12 Mar 2008) | 14 lines Change AST_SCHED_DEL use to ast_sched_del for autocongestion in chan_sip. The scheduler callback will always return 0. This means that this id is never rescheduled, so it makes no sense to loop trying to delete the id from the scheduler queue. If we fail to remove the item from the queue once, it will fail every single time. (Yes I realize that in this case, the macro would exit early because the id is set to -1 in the callback, but it still makes no sense to use that macro in favor of calling ast_sched_del once and being done with it) This is the first of potentially several such fixes. ........ ................ * /, include/asterisk/sched.h: Merged revisions 108238 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r108238 | mmichelson | 2008-03-12 16:19:30 -0500 (Wed, 12 Mar 2008) | 20 lines Merged revisions 108227 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r108227 | mmichelson | 2008-03-12 16:16:28 -0500 (Wed, 12 Mar 2008) | 12 lines Added a large comment before the AST_SCHED_DEL macro to explain its purpose as well as when it is appropriate and when it is not appropriate to use it. I also removed the part of the debug message that mentions that this is probably a bug because there are some perfectly legitimate places where ast_sched_del may fail to delete an entry (e.g. when the scheduler callback manually reschedules with a new id instead of returning non-zero to tell the scheduler to reschedule with the same idea). I also raised the debug level of the debug message in AST_SCHED_DEL since it seems like it could come up quite frequently since the macro is probably being used in several places where it shouldn't be. Also removed the redundant line, file, and function information since that is provided by ast_log. ........ ................ 2008-03-12 20:29 +0000 [r108205] Kevin P. Fleming * /, channels/chan_sip.c: Merged revisions 108191 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r108191 | kpfleming | 2008-03-12 15:27:01 -0500 (Wed, 12 Mar 2008) | 14 lines Merged revisions 108086 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r108086 | kpfleming | 2008-03-12 14:16:07 -0500 (Wed, 12 Mar 2008) | 6 lines if we receive an INVITE with a Content-Length that is not a valid number, or is zero, then don't process the rest of the message body looking for an SDP closes issue #11475 Reported by: andrebarbosa ........ ................ 2008-03-12 19:59 +0000 [r108138] Russell Bryant * apps/app_chanspy.c, main/channel.c, /: Merged revisions 108137 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r108137 | russell | 2008-03-12 14:59:05 -0500 (Wed, 12 Mar 2008) | 48 lines Merged revisions 108135 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r108135 | russell | 2008-03-12 14:57:42 -0500 (Wed, 12 Mar 2008) | 40 lines (closes issue #12187, reported by atis, fixed by me after some brainstorming on the issue with mmichelson) - Update copyright info on app_chanspy. - Fix a race condition that caused app_chanspy to crash. The issue was that the chanspy datastore magic that was used to ensure that spyee channels did not disappear out from under the code did not completely solve the problem. It was actually possible for chanspy to acquire a channel reference out of its datastore to a channel that was in the middle of being destroyed. That was because datastore destruction in ast_channel_free() was done near the end. So, this left the code in app_chanspy accessing a channel that was partially, or completely invalid because it was in the process of being free'd by another thread. The following sort of shows the code path where the race occurred: ============================================================================= Thread 1 (PBX thread for spyee chan) || Thread 2 (chanspy) --------------------------------------||------------------------------------- ast_channel_free() || - remove channel from channel list || - lock/unlock the channel to ensure || that no references retrieved from || the channel list exist. || --------------------------------------||------------------------------------- || channel_spy() - destroy some channel data || - Lock chanspy datastore || - Retrieve reference to channel || - lock channel || - Unlock chanspy datastore --------------------------------------||------------------------------------- - destroy channel datastores || - call chanspy datastore d'tor || which NULL's out the ds' || - Operate on the channel ... reference to the channel || || - free the channel || || || - unlock the channel --------------------------------------||------------------------------------- ============================================================================= ........ ................ 2008-03-12 18:31 +0000 [r108085] Joshua Colp * apps/app_mixmonitor.c, /, include/asterisk/audiohook.h, main/audiohook.c: Merged revisions 108084 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r108084 | file | 2008-03-12 15:29:33 -0300 (Wed, 12 Mar 2008) | 12 lines Merged revisions 108083 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r108083 | file | 2008-03-12 15:26:37 -0300 (Wed, 12 Mar 2008) | 4 lines Add a trigger mode that triggers on both read and write. The actual function that returns the combined audio frame though will wait until both sides have fed in audio, or until one side stops (such as the case when you call Wait). (closes issue #11945) Reported by: xheliox ........ ................ 2008-03-12 17:03 +0000 [r108033] Russell Bryant * main/channel.c, /: Merged revisions 108032 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r108032 | russell | 2008-03-12 12:02:57 -0500 (Wed, 12 Mar 2008) | 12 lines Merged revisions 108031 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r108031 | russell | 2008-03-12 11:59:07 -0500 (Wed, 12 Mar 2008) | 4 lines Destroy the channel lock after the channel datastores. (inspired by issue #12187) ........ ................ 2008-03-12 07:44 +0000 [r107879-107999] Tilghman Lesher * /, channels/chan_sip.c: Merged revisions 107998 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r107998 | tilghman | 2008-03-12 02:43:03 -0500 (Wed, 12 Mar 2008) | 7 lines Deadlock fixes (closes issue #12143) Reported by: kactus Patches: 20080312__bug12143__2.diff.txt uploaded by Corydon76 (license 14) Tested by: kactus ........ * main/loader.c, /, apps/app_dumpchan.c, apps/app_zapras.c: Merged revisions 107960 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r107960 | tilghman | 2008-03-12 00:46:39 -0500 (Wed, 12 Mar 2008) | 4 lines Revert several changes from revision 102525, as the changes were not compatible, and, in fact, introduced regressions. (Closes issue #12190) ........ * contrib/scripts/iax-friends.sql, /, contrib/scripts/sip-friends.sql: Merged revisions 107878 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r107878 | tilghman | 2008-03-11 20:54:00 -0500 (Tue, 11 Mar 2008) | 10 lines Merged revisions 107877 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r107877 | tilghman | 2008-03-11 20:52:40 -0500 (Tue, 11 Mar 2008) | 2 lines Document all of the possible realtime fields ........ ................ 2008-03-11 23:38 +0000 [r107828] Jason Parker * /, doc/voicemail_odbc_postgresql.txt: Merged revisions 107827 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r107827 | qwell | 2008-03-11 18:38:00 -0500 (Tue, 11 Mar 2008) | 15 lines Merged revisions 107826 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r107826 | qwell | 2008-03-11 18:37:05 -0500 (Tue, 11 Mar 2008) | 7 lines Update documentation for pgsql ODBC voicemail. (closes issue #12186) Reported by: jsmith Patches: vm_pgsql_doc_update.patch uploaded by jsmith (license 15) ........ ................ 2008-03-11 22:59 +0000 [r107723-107793] Tilghman Lesher * res/res_config_sqlite.c, main/config.c, res/res_config_curl.c, res/res_config_pgsql.c, res/res_config_odbc.c, /, include/asterisk/config.h, res/res_config_ldap.c: Merged revisions 107791 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r107791 | tilghman | 2008-03-11 17:55:16 -0500 (Tue, 11 Mar 2008) | 5 lines An offhand comment from Russell made me realize that the configuration file caching would not work properly for users.conf and any other file read from more than one place. I needed to add the filename which requested the config file to get it to work properly. ........ 2008-03-11 20:54 +0000 [r107720] Jason Parker * channels/chan_jingle.c, /, channels/chan_gtalk.c: Merged revisions 107718 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r107718 | qwell | 2008-03-11 15:53:48 -0500 (Tue, 11 Mar 2008) | 13 lines Merged revisions 107714 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r107714 | qwell | 2008-03-11 15:49:56 -0500 (Tue, 11 Mar 2008) | 5 lines Copy voicemail dependency logic for res_adsi to chan_gtalk and chan_jingle (for jabber). (closes issue #12014) Reported by: junky ........ ................ 2008-03-11 20:51 +0000 [r107716] Kevin P. Fleming * /, Makefile.rules, channels/Makefile: Merged revisions 107715 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r107715 | kpfleming | 2008-03-11 15:50:57 -0500 (Tue, 11 Mar 2008) | 10 lines Merged revisions 107713 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r107713 | kpfleming | 2008-03-11 15:48:58 -0500 (Tue, 11 Mar 2008) | 2 lines get chan_vpb to build properly in dev mode ........ ................ 2008-03-11 20:37 +0000 [r107584-107711] Joshua Colp * /, apps/app_page.c: Merged revisions 107710 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r107710 | file | 2008-03-11 17:36:14 -0300 (Tue, 11 Mar 2008) | 6 lines Dial a device even if it's state is unknown. (closes issue #12184) Reported by: bluecrow76 Patches: asterisk-svn-app_page.c.devicestate_unknown.diff uploaded by bluecrow76 (license 270) ........ * /, main/features.c: Merged revisions 107659 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r107659 | file | 2008-03-11 16:23:28 -0300 (Tue, 11 Mar 2008) | 12 lines Merged revisions 107646 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r107646 | file | 2008-03-11 16:20:01 -0300 (Tue, 11 Mar 2008) | 4 lines Make sure the visible indication is on the right channel so when the masquerade happens the proper indication is enacted. (closes issue #11707) Reported by: iam ........ ................ * /, apps/app_meetme.c: Merged revisions 107638 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r107638 | file | 2008-03-11 15:48:59 -0300 (Tue, 11 Mar 2008) | 12 lines Merged revisions 107637 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r107637 | file | 2008-03-11 15:47:33 -0300 (Tue, 11 Mar 2008) | 4 lines Add an additional check for setting conference parameter when using the marked user options. It was possible for it to return to a no listen/no talk state if a masquerade happened. (closes issue #12136) Reported by: aragon ........ ................ 2008-03-11 15:39 +0000 [r107374-107526] Kevin P. Fleming * channels/chan_vpb.cc, /: Merged revisions 107525 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r107525 | kpfleming | 2008-03-11 10:39:37 -0500 (Tue, 11 Mar 2008) | 2 lines fix another potential bug found by gcc 4.3 ........ * apps/app_rpt.c, channels/misdn/isdn_lib.c, codecs/Makefile, /, apps/app_sms.c: Merged revisions 107466 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r107466 | kpfleming | 2008-03-11 10:13:38 -0500 (Tue, 11 Mar 2008) | 10 lines Merged revisions 107464 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r107464 | kpfleming | 2008-03-11 09:53:03 -0500 (Tue, 11 Mar 2008) | 2 lines fix various other problems found by gcc 4.3 ........ ................ * /, configure, include/asterisk/autoconfig.h.in, configure.ac, apps/app_sms.c: Merged revisions 107462 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r107462 | kpfleming | 2008-03-11 09:37:03 -0500 (Tue, 11 Mar 2008) | 10 lines Merged revisions 107461 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r107461 | kpfleming | 2008-03-11 09:33:45 -0500 (Tue, 11 Mar 2008) | 2 lines stop checking for mktime() in the configure script... we don't use it, and the test is buggy under gcc 4.3 ........ ................ * /, configure, main/Makefile, configure.ac, makeopts.in: Merged revisions 107409 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r107409 | kpfleming | 2008-03-11 09:09:49 -0500 (Tue, 11 Mar 2008) | 13 lines Merged revisions 107408 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r107408 | kpfleming | 2008-03-11 09:07:59 -0500 (Tue, 11 Mar 2008) | 5 lines check for compiler support for -fno-strict-overflow before using it (tested with Debian's gcc 4.3, 4.1 and 3.4) (closes issue #12179) Reported by: Netview ........ ................ * /, configure, include/asterisk/autoconfig.h.in, configure.ac: Merged revisions 107406 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r107406 | kpfleming | 2008-03-11 08:58:37 -0500 (Tue, 11 Mar 2008) | 10 lines Merged revisions 107405 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r107405 | kpfleming | 2008-03-11 08:57:08 -0500 (Tue, 11 Mar 2008) | 2 lines fix small bug in IMAP toolkit testing ........ ................ * main/udptl.c, utils/Makefile, /, main/Makefile, main/editline/readline.c, res/Makefile: Merged revisions 107373 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r107373 | kpfleming | 2008-03-11 06:36:51 -0500 (Tue, 11 Mar 2008) | 19 lines Merged revisions 107352 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r107352 | kpfleming | 2008-03-11 06:04:29 -0500 (Tue, 11 Mar 2008) | 11 lines fix up various compiler warnings found with gcc-4.3: - the output of flex includes a static function called 'input' that is not used, so for the moment we'll stop having the compiler tell us about unused variables in the flex source files (a better fix would be to improve our flex post-processing to remove the unused function) - main/stdtime/localtime.c makes assumptions about signed integer overflow, and gcc-4.3's improved optimizer tries to take advantage of handling potential overflow conditions at compile time; for now, suppress these optimizations until we can fiure out if the code needs improvement - main/udptl.c has some references to uninitialized variables; in one case there was no bug, but in the other it was certainly possibly for unexpected behavior to occur - main/editline/readline.c had an unused variable ........ ................ 2008-03-11 01:27 +0000 [r107336] Terry Wilson * /, channels/chan_sip.c: Merged revisions 107292 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r107292 | twilson | 2008-03-10 20:09:46 -0500 (Mon, 10 Mar 2008) | 10 lines Merged revisions 107290 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r107290 | twilson | 2008-03-10 19:59:18 -0500 (Mon, 10 Mar 2008) | 2 lines If we fail to alloc a channel, we should re-lock the pvt structure before returning. ........ ................ 2008-03-10 23:46 +0000 [r107289] Steve Murphy * main/cdr.c, /: Merged revisions 107019 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r107019 | murf | 2008-03-10 08:55:21 -0600 (Mon, 10 Mar 2008) | 1 line way back in July, in r.75706, a fix was made ot the strftime usages, which was good, but in this case, the check for a nil time was accidentally removed, and now it is restored, to keep timevals like '1969-12-31 17:00:00' from showing up in the cdrs. No idea what databases will do with this. No bugs filed as yet, but it felt like a bug. ........ 2008-03-10 20:29 +0000 [r107180] Jason Parker * channels/chan_zap.c, /: Merged revisions 107177 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r107177 | qwell | 2008-03-10 15:28:33 -0500 (Mon, 10 Mar 2008) | 13 lines Merged revisions 107173 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r107173 | qwell | 2008-03-10 15:27:08 -0500 (Mon, 10 Mar 2008) | 5 lines Make sure to reenable echo can after a "failed" (canceled, etc) three-way call. (closes issue #11335) Reported by: rebuild ........ ................ 2008-03-10 20:18 +0000 [r107101-107163] Russell Bryant * main/pbx.c, /: Merged revisions 107162 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r107162 | russell | 2008-03-10 15:17:37 -0500 (Mon, 10 Mar 2008) | 16 lines Merged revisions 107161 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r107161 | russell | 2008-03-10 15:17:11 -0500 (Mon, 10 Mar 2008) | 8 lines Fix another bug specifically related to asynchronous call origination. Once the PBX is started on the channel using ast_pbx_start(), then the ownership of the channel has been passed on to another thread. We can no longer access it in this code. If the channel gets hung up very quickly, it is possible that we could access a channel that has been free'd. (inspired by BE-386) ........ ................ * main/pbx.c, /: Merged revisions 107159 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r107159 | russell | 2008-03-10 15:05:12 -0500 (Mon, 10 Mar 2008) | 17 lines Merged revisions 107158 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r107158 | russell | 2008-03-10 15:04:27 -0500 (Mon, 10 Mar 2008) | 9 lines Fix some bugs related to originating calls. If the code failed to start a PBX on the channel (such as if you set a call limit based on the system's load average), then there were cases where a channel that has already been free'd using ast_hangup() got accessed. This caused weird memory corruption and crashes to occur. (fixes issue BE-386) (much debugging credit goes to twilson, final patch written by me) ........ ................ * main/channel.c, /: Merged revisions 107103 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r107103 | russell | 2008-03-10 12:13:34 -0500 (Mon, 10 Mar 2008) | 10 lines Merged revisions 107102 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r107102 | russell | 2008-03-10 12:13:17 -0500 (Mon, 10 Mar 2008) | 2 lines Resolve a compiler warning. ........ ................ * main/channel.c, /: Merged revisions 107100 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r107100 | russell | 2008-03-10 11:59:13 -0500 (Mon, 10 Mar 2008) | 11 lines Merged revisions 107099 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r107099 | russell | 2008-03-10 11:58:57 -0500 (Mon, 10 Mar 2008) | 3 lines Fix a race condition where the generator can go away (closes issue #12175, reported by edantie, patched by me) ........ ................ 2008-03-10 15:46 +0000 [r107069] Mark Michelson * /, apps/app_queue.c: Merged revisions 107068 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r107068 | mmichelson | 2008-03-10 10:45:13 -0500 (Mon, 10 Mar 2008) | 10 lines app_queue has now been doxygenified thanks to snuffy! The ony thing I changed was the way that locks are referenced, since the old 1.2 names were still used in the comments. (closes issue #11997) Reported by: snuffy Patches: bug_11997_queue_doxy.diff uploaded by snuffy (license 35) ........ 2008-03-10 14:38 +0000 [r107018] Joshua Colp * apps/app_dial.c, main/cdr.c, /, include/asterisk/cdr.h: Merged revisions 107017 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r107017 | file | 2008-03-10 11:36:16 -0300 (Mon, 10 Mar 2008) | 15 lines Merged revisions 107016 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r107016 | file | 2008-03-10 11:33:02 -0300 (Mon, 10 Mar 2008) | 7 lines Move where unanswered CDRs are dropped to the CDR core, not everything uses app_dial. (closes issue #11516) Reported by: ys Patches: branch_1.4_cdr.diff uploaded by ys (license 281) Tested by: anest, jcapp, dartvader ........ ................ 2008-03-08 17:54 +0000 [r106997] Matthew Fredrickson * channels/chan_zap.c: Make sure we don't start a call on a channel that has already started a call 2008-03-08 16:14 +0000 [r106947] Kevin P. Fleming * channels/chan_zap.c, /: Merged revisions 106946 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r106946 | kpfleming | 2008-03-08 10:03:48 -0600 (Sat, 08 Mar 2008) | 10 lines Merged revisions 106945 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r106945 | kpfleming | 2008-03-08 09:59:42 -0600 (Sat, 08 Mar 2008) | 2 lines don't generate D-Channel "up" and "down" messages unless the channel state is actually changing; also, generate the "up" message when an implicit "up" occurs due to reception of a normal event when we thought the channel was "down" ........ ................ 2008-03-07 22:53 +0000 [r106897] Russell Bryant * /, apps/app_meetme.c: Merged revisions 106896 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r106896 | russell | 2008-03-07 16:52:46 -0600 (Fri, 07 Mar 2008) | 10 lines Merged revisions 106895 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r106895 | russell | 2008-03-07 16:51:23 -0600 (Fri, 07 Mar 2008) | 2 lines Only start the SLA thread if SLA has actually been configured. ........ ................ 2008-03-07 19:34 +0000 [r106790] Joshua Colp * main/channel.c, /: Merged revisions 106789 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r106789 | file | 2008-03-07 15:33:09 -0400 (Fri, 07 Mar 2008) | 12 lines Merged revisions 106788 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r106788 | file | 2008-03-07 15:32:00 -0400 (Fri, 07 Mar 2008) | 4 lines Ignore source update control frame. (closes issue #12168) Reported by: plack ........ ................ 2008-03-07 17:18 +0000 [r106686-106713] Russell Bryant * /, include/asterisk/sched.h: Merged revisions 106707 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r106707 | russell | 2008-03-07 11:17:30 -0600 (Fri, 07 Mar 2008) | 16 lines Merged revisions 106704 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r106704 | russell | 2008-03-07 11:16:58 -0600 (Fri, 07 Mar 2008) | 8 lines Change a warning message to a debug message. This is happening quite frequently, and it is not worth spamming users with these messages unless we are pretty confident that it should never happen. As it stands today, it _will_ and _does_ happen and until that gets cleaned up a reasonable amount on the development side, let's not spam the logs of everyone else. (closes issue #12154) ........ ................ * doc/smdi.txt, /: Merged revisions 106684 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r106684 | russell | 2008-03-07 10:31:48 -0600 (Fri, 07 Mar 2008) | 2 lines fix example usage ........ 2008-03-07 16:27 +0000 [r106554-106662] Tilghman Lesher * apps/app_voicemail.c, /: Merged revisions 106654 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r106654 | tilghman | 2008-03-07 10:26:07 -0600 (Fri, 07 Mar 2008) | 11 lines Merged revisions 106635 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r106635 | tilghman | 2008-03-07 10:22:11 -0600 (Fri, 07 Mar 2008) | 3 lines Warn the user when a temporary greeting exists (Closes issue #11409) ........ ................ * main/rtp.c, /: Merged revisions 106607 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r106607 | tilghman | 2008-03-07 09:22:34 -0600 (Fri, 07 Mar 2008) | 11 lines Merged revisions 106606 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r106606 | tilghman | 2008-03-07 09:20:52 -0600 (Fri, 07 Mar 2008) | 3 lines Properly initialize rtp->schedid (Closes issue #12154) ........ ................ * apps/app_chanspy.c, apps/app_rpt.c, main/asterisk.c, apps/app_speech_utils.c, apps/app_voicemail.c, main/channel.c, funcs/func_enum.c, channels/chan_misdn.c, main/frame.c, /, channels/chan_sip.c, funcs/func_odbc.c, funcs/func_strings.c, utils/extconf.c: Merged revisions 106553 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r106553 | tilghman | 2008-03-07 00:54:47 -0600 (Fri, 07 Mar 2008) | 14 lines Merged revisions 106552 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r106552 | tilghman | 2008-03-07 00:36:33 -0600 (Fri, 07 Mar 2008) | 6 lines Safely use the strncat() function. (closes issue #11958) Reported by: norman Patches: 20080209__bug11958.diff.txt uploaded by Corydon76 (license 14) ........ ................ 2008-03-07 01:19 +0000 [r106502-106520] Russell Bryant * doc/smdi.txt, /: Merged revisions 106518 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r106518 | russell | 2008-03-06 19:19:02 -0600 (Thu, 06 Mar 2008) | 1 line minor text changes ........ * doc/smdi.txt, /: Merged revisions 106507 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r106507 | russell | 2008-03-06 19:15:36 -0600 (Thu, 06 Mar 2008) | 2 lines Add updated SMDI documentation that I had only sitting in my email ... oops ........ * main/rtp.c, codecs/codec_g722.c, /, formats/format_pcm.c, main/file.c: Merged revisions 106501 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r106501 | russell | 2008-03-06 18:24:58 -0600 (Thu, 06 Mar 2008) | 28 lines Merge changes from team/russell/g722-sillyness ... Fix a number of other places where the number of samples in a G722 frame was not properly handled because of various reasons. main/rtp.c: - When a G722 frame is read from the smoother, the number of samples in the frame must be divided by 2 before being sent out over the network. Even though G722 is 16 kHz, an error in some previous spec has made it so that we have to list the number of samples such as if it was 8 kHz. main/file.c: - When scheduling the next time to expect a frame, take into account that the format of the file we're reading from may not be 8 kHz. codecs/codec_g722.c: - When converting from G722 to slinear, g722_decode() expects its samples parameter to be in the silly (real samples / 2) format. Make it so. - When converting from slinear to G722, properly set the number of samples in the frame to be the number of bytes of output * 2. formats/format_pcm.c: - This format module handles G722, among a number of other formats. However, the read() and seek() functions did not account for the fact that G722 has 2 samples per byte. (closes issue #12130, reported by rickross, patched by me) ........ 2008-03-06 22:16 +0000 [r106442] Mark Michelson * main/pbx.c, /: Merged revisions 106438 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r106438 | mmichelson | 2008-03-06 16:11:26 -0600 (Thu, 06 Mar 2008) | 16 lines Merged revisions 106437 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r106437 | mmichelson | 2008-03-06 16:10:07 -0600 (Thu, 06 Mar 2008) | 8 lines Quell an annoying message that is likely to print every single time that ast_pbx_outgoing_app is called. The reason is that __ast_request_and_dial allocates the cdr for the channel, so it should be expected that the channel will have a cdr on it. Thanks to joetester on IRC for pointing this out ........ ................ 2008-03-06 22:15 +0000 [r106440] Jason Parker * /, main/file.c: Merged revisions 106439 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r106439 | qwell | 2008-03-06 16:11:30 -0600 (Thu, 06 Mar 2008) | 8 lines Fix file playback in many cases. (closes issue #12115) Reported by: pj Patches: v2-fileexists.patch uploaded by dimas (license 88) (with modifications by me) Tested by: dimas, qwell, russell ........ 2008-03-06 20:39 +0000 [r106433] Donny Kavanagh * /, res/res_agi.c: Merged revisions 106399 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r106399 | juggie | 2008-03-06 14:31:50 -0500 (Thu, 06 Mar 2008) | 9 lines trivial fix for an agi error when attempting to use EAGI on a dead/hungup channel, we now print an error that makes sense given our removal of deadagi as an actual application. (closes issue #12161) Reported by: explidous Patches: res_agi_12161.patch uploaded by juggie (license 24) Tested by: juggie ........ 2008-03-06 05:25 +0000 [r106330-106359] Tilghman Lesher * /, res/res_config_ldap.c: Merged revisions 106346 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r106346 | tilghman | 2008-03-05 23:21:39 -0600 (Wed, 05 Mar 2008) | 7 lines Missing braces, fix parsing (closes issue #12112) Reported by: cyrenity Patches: res_config_ldap.patch-03-03-2008 uploaded by cyrenity (license 416) Tested by: cyrenity, Corydon76 ........ * /, sounds/sounds.xml, sounds/Makefile: Merged revisions 106329 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r106329 | tilghman | 2008-03-05 22:45:16 -0600 (Wed, 05 Mar 2008) | 10 lines Merged revisions 106328 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r106328 | tilghman | 2008-03-05 22:40:06 -0600 (Wed, 05 Mar 2008) | 2 lines Upgrade to the next release of sounds ........ ................ 2008-03-06 00:23 +0000 [r106299-106320] Russell Bryant * channels/chan_oss.c, main/rtp.c, main/channel.c, channels/chan_phone.c, main/dial.c, channels/chan_skinny.c, main/file.c, channels/chan_h323.c, channels/chan_alsa.c, include/asterisk/frame.h, channels/chan_mgcp.c, channels/chan_unistim.c, apps/app_dial.c, channels/chan_zap.c, /, channels/chan_sip.c, channels/chan_console.c, apps/app_followme.c: Merged revisions 106239 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r106239 | file | 2008-03-05 16:43:22 -0600 (Wed, 05 Mar 2008) | 12 lines Merged revisions 106235 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r106235 | file | 2008-03-05 18:32:10 -0400 (Wed, 05 Mar 2008) | 4 lines Add a control frame to indicate the source of media has changed. Depending on the underlying technology it may need to change some things. (closes issue #12148) Reported by: jcomellas ........ ................ * /, channels/chan_iax2.c: Merged revisions 106238 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r106238 | russell | 2008-03-05 16:40:58 -0600 (Wed, 05 Mar 2008) | 11 lines Merged revisions 106237 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r106237 | russell | 2008-03-05 16:37:09 -0600 (Wed, 05 Mar 2008) | 3 lines Fix a potential deadlock and a few different potential crashes. (closes issue #12145, reported by thiagarcia, patched by me) ........ ................ * /, doc/tex/realtime.tex: Merged revisions 106186 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r106186 | mvanbaak | 2008-03-05 15:19:06 -0600 (Wed, 05 Mar 2008) | 7 lines document var_metric usage to prevent bugreports that are actually configuration issues (closes issue #12151) Reported by: caio1982 Patches: DB_metric3.diff uploaded by caio1982 (license 22) ........ * main/rtp.c, /, main/translate.c, include/asterisk/frame.h: Merged revisions 105933 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r105933 | russell | 2008-03-04 19:54:16 -0600 (Tue, 04 Mar 2008) | 13 lines Merged revisions 105932 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r105932 | russell | 2008-03-04 19:52:18 -0600 (Tue, 04 Mar 2008) | 5 lines Fix a bug that I just noticed in the RTP code. The calculation for setting the len field in an ast_frame of audio was wrong when G.722 is in use. The len field represents the number of ms of audio that the frame contains. It would have set the value to be twice what it should be. ........ ................ * funcs/func_global.c, /: Merged revisions 105899 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r105899 | russell | 2008-03-04 18:45:39 -0600 (Tue, 04 Mar 2008) | 3 lines Fix the SHARED() read callback to properly unlock the channel. This function could not have worked, as it left the channel locked in all cases. ........ * main/manager.c, /: Merged revisions 105864 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r105864 | mmichelson | 2008-03-04 17:24:56 -0600 (Tue, 04 Mar 2008) | 5 lines There are several places in manager.c where BUFSIZ is used for a buffer which will contain nowhere near that amount of data. This makes these buffers more reasonably sized. ........ * main/asterisk.c, channels/chan_zap.c, /, channels/console_gui.c, apps/app_queue.c: Merged revisions 105841 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r105841 | tilghman | 2008-03-04 17:10:45 -0600 (Tue, 04 Mar 2008) | 2 lines Fix minor misuses of snprintf ........ * main/rtp.c, main/netsock.c, main/cryptostub.c, main/file.c, main/callerid.c, main/alaw.c, main/dsp.c, main/dlfcn.c, main/frame.c, /, main/say.c, main/utils.c, main/enum.c, main/astobj2.c, main/config.c, main/fskmodem.c, main/poll.c, main/loader.c, main/term.c, main/cli.c, main/channel.c, main/dial.c, main/manager.c, main/tdd.c, main/strcompat.c, main/features.c, main/logger.c, main/app.c, main/image.c, main/dns.c, main/pbx.c, main/translate.c, main/jitterbuf.c: Merged revisions 105840 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r105840 | tilghman | 2008-03-04 17:04:29 -0600 (Tue, 04 Mar 2008) | 2 lines Whitespace changes only ........ * main/tcptls.c, main/manager.c, /, channels/chan_sip.c, main/http.c, include/asterisk/tcptls.h: Merged revisions 105804 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r105804 | russell | 2008-03-04 16:28:03 -0600 (Tue, 04 Mar 2008) | 2 lines add a destroy API call for a server instance ........ * main/tcptls.c, main/manager.c, /, channels/chan_sip.c, main/http.c, include/asterisk/tcptls.h: Merged revisions 105785 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r105785 | russell | 2008-03-04 16:23:21 -0600 (Tue, 04 Mar 2008) | 2 lines More public API name changes to use an appropriate ast_ prefix ........ * include/asterisk/http.h, main/tcptls.c, main/manager.c, /, channels/chan_sip.c, res/res_phoneprov.c, main/http.c, include/asterisk/tcptls.h: Merged revisions 105773 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r105773 | russell | 2008-03-04 16:15:18 -0600 (Tue, 04 Mar 2008) | 2 lines Rename public object server_instance to ast_tcptls_server_instance ........ * /, channels/chan_sip.c: Merged revisions 105734 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r105734 | russell | 2008-03-04 14:36:16 -0600 (Tue, 04 Mar 2008) | 6 lines Fix some bugs in the SIP tcp helper thread. - fix a spot where a lock wouldn't get unlocked in an error condition - call ast_mutex_destroy() on the lock before freeing its memory (related to issue #11972) ........ * /, res/res_phoneprov.c: Merged revisions 105733 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r105733 | twilson | 2008-03-04 14:32:55 -0600 (Tue, 04 Mar 2008) | 2 lines Set username to default to the category name if it isn't overridden by a usernmae= setting in users.conf ........ * main/rtp.c, /: Merged revisions 105677 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r105677 | file | 2008-03-04 12:11:38 -0600 (Tue, 04 Mar 2008) | 10 lines Merged revisions 105676 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r105676 | file | 2008-03-04 14:10:34 -0400 (Tue, 04 Mar 2008) | 2 lines In addition to setting the marker bit let's change our ssrc so they know for sure it is a different source. ........ ................ * main/rtp.c, /, channels/chan_sip.c, include/asterisk/rtp.h: Merged revisions 105675 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r105675 | file | 2008-03-04 12:08:42 -0600 (Tue, 04 Mar 2008) | 16 lines Merged revisions 105674 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r105674 | file | 2008-03-04 14:05:28 -0400 (Tue, 04 Mar 2008) | 8 lines When a new source of audio comes in (such as music on hold) make sure the marker bit gets set. (closes issue #10355) Reported by: wdecarne Patches: 10355.diff uploaded by file (license 11) (closes issue #11491) Reported by: kanderson ........ ................ 2008-03-05 17:42 +0000 [r106140] Tilghman Lesher * /, apps/app_talkdetect.c: Merged revisions 106139 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ........ r106139 | tilghman | 2008-03-05 11:40:42 -0600 (Wed, 05 Mar 2008) | 3 lines Should check these values for non-NULL before scanning. (Closes issue #12147) ........ 2008-03-05 15:43 +0000 [r106041] Kevin P. Fleming * channels/chan_zap.c, /: Merged revisions 106040 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r106040 | kpfleming | 2008-03-05 09:40:40 -0600 (Wed, 05 Mar 2008) | 15 lines Merged revisions 106038 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r106038 | kpfleming | 2008-03-05 09:32:35 -0600 (Wed, 05 Mar 2008) | 7 lines when a PRI call must be moved to a different B channel at the request of the other endpoint, ensure that any DSP active on the original channel is moved to the new one (closes issue #11917) Reported by: mavetju Tested by: mavetju ........ ................ 2008-03-05 15:31 +0000 [r106037] Tilghman Lesher * /, channels/chan_sip.c, include/asterisk/sched.h: Merged revisions 106036 via svnmerge from https://origsvn.digium.com/svn/asterisk/trunk ................ r106036 | tilghman | 2008-03-05 09:23:32 -0600 (Wed, 05 Mar 2008) | 15 lines Merged revisions 106015 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r106015 | tilghman | 2008-03-05 09:17:16 -0600 (Wed, 05 Mar 2008) | 7 lines Correctly initialize retransid in SIP, and ensure that the warning when failing to delete a schedule entry can actually hit the log. (closes issue #12140) Reported by: slavon Patches: sch2.patch uploaded by slavon (license 288) (Patch slightly modified by me) ........ ................ 2008-03-04 Russell Bryant * Asterisk 1.6.0-beta5 released. 2008-03-04 16:55 +0000 [r105574-105597] Russell Bryant * CHANGES: Update CHANGES heading * funcs/func_version.c: Simplify a trivial snprintf() with ast_copy_string() * main/hashtab.c: Make it so you don't have to cast away const in a couple places * main/hashtab.c: remove unnecessary casts * main/pbx.c: - Add curly braces around the while loop - Properly break out of the loop on error when an included context is not found * main/pbx.c: Use ast_copy_string() instead of strncpy(), and use sizeof() instead of a magic number * channels/chan_zap.c: Fix some code that was improperly changed in revision 104866 from issue #12079. (closes issue #12129, reported by elguero, patched by me) 2008-03-03 18:08 +0000 [r105573] Jason Parker * /, res/snmp/agent.c: Merged revisions 105572 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r105572 | qwell | 2008-03-03 12:06:52 -0600 (Mon, 03 Mar 2008) | 7 lines Fix types for astNumChannels and astConfigCallsProcessed. (closes issue #12114) Reported by: jeffg Patches: 12114.patch uploaded by jeffg (license 192) ........ 2008-03-03 17:17 +0000 [r105564-105571] Russell Bryant * channels/chan_local.c, /: Merged revisions 105570 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r105570 | russell | 2008-03-03 11:16:53 -0600 (Mon, 03 Mar 2008) | 3 lines In the case of an ast_channel allocation failure, take the local_pvt out of the pvt list before destroying it. ........ * channels/chan_local.c, /: Merged revisions 105568 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r105568 | russell | 2008-03-03 11:05:16 -0600 (Mon, 03 Mar 2008) | 3 lines Fix a potential memory leak of the local_pvt struct when ast_channel allocation fails. Also, in passing, centralize the code necessary to destroy a local_pvt. ........ * main/autoservice.c, /: Merged revisions 105565 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r105565 | russell | 2008-03-03 10:01:50 -0600 (Mon, 03 Mar 2008) | 3 lines Update the copyright information for autoservice. Most of the code in this file now is stuff that I have written recently ... ........ * main/channel.c, main/autoservice.c, /, include/asterisk/_private.h, main/asterisk.c: 3) In addition to merging the changes below, change trunk back to a regular LIST instead of an RWLIST. The way this list works makes it such that a RWLIST provides no additional benefit. Also, a mutex is needed for use with the thread condition. Merged revisions 105563 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r105563 | russell | 2008-03-03 09:50:43 -0600 (Mon, 03 Mar 2008) | 24 lines Merge in some changes from team/russell/autoservice-nochans-1.4 These changes fix up some dubious code that I came across while auditing what happens in the autoservice thread when there are no channels currently in autoservice. 1) Change it so that autoservice thread doesn't keep looping around calling ast_waitfor_n() on 0 channels twice a second. Instead, use a thread condition so that the thread properly goes to sleep and does not wake up until a channel is put into autoservice. This actually fixes an interesting bug, as well. If the autoservice thread is already running (almost always is the case), then when the thread goes from having 0 channels to have 1 channel to autoservice, that channel would have to wait for up to 1/2 of a second to have the first frame read from it. 2) Fix up the code in ast_waitfor_nandfds() for when it gets called with no channels and no fds to poll() on, such as was the case with the previous code for the autoservice thread. In this case, the code would call alloca(0), and pass the result as the first argument to poll(). In this case, the 2nd argument to poll() specified that there were no fds, so this invalid pointer shouldn't actually get dereferenced, but, this code makes it explicit and ensures the pointers are NULL unless we have valid data to put there. (related to issue #12116) ........ 2008-03-03 15:30 +0000 [r105558-105561] Joshua Colp * main/channel.c, /: Merged revisions 105560 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r105560 | file | 2008-03-03 11:28:59 -0400 (Mon, 03 Mar 2008) | 7 lines It is possible for no audio to pass between the current digit and next digit so expand logic that clears emulation to AST_FRAME_NULL. (closes issue #11911) Reported by: edgreenberg Patches: v1-11911.patch uploaded by dimas (license 88) Tested by: tbsky ........ * /, channels/chan_sip.c: Merged revisions 105557 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r105557 | file | 2008-03-03 11:15:39 -0400 (Mon, 03 Mar 2008) | 6 lines Add a comment to describe some logic. (closes issue #12120) Reported by: flefoll Patches: chan_sip.c.br14.patch-just-a-comment uploaded by flefoll (license 244) ........ 2008-03-01 03:59 +0000 [r105509] Joshua Colp * main/slinfactory.c: Add support for 16KHz signed linear. 2008-03-01 02:03 +0000 [r105479] Tilghman Lesher * /: Drop bad property 2008-03-01 01:30 +0000 [r105477] Terry Wilson * apps/app_dial.c, include/asterisk/app.h, main/global_datastores.c, /, main/features.c, main/app.c, include/asterisk/global_datastores.h: Asterisk, when parking can drop rights a caller when a parking timeout occurs. Also, when doing built-in attended transfers, sometimes incorrectly passes rights from the transferrer to the transferee. This patch tries to fixes the parking issue and lays some groundwork for later fixing the transfer issue. (closes issue #11520) Reported by: pliew Tested by: otherwiseguy 2008-03-01 00:53 +0000 [r105461] Russell Bryant * CHANGES, funcs/func_devstate.c: Add a "devstate change" CLI command to control custom device states. Also, do some additional code cleanup and improvement in passing. (closes issue #12106) Reported by: nizon Patches: devstate-patch.txt uploaded by nizon (license 415) -- Updated to trunk, and tab completion added by me 2008-02-29 23:53 +0000 [r105411] Tilghman Lesher * cdr/cdr_adaptive_odbc.c: Convert to use ast_str 2008-02-29 23:36 +0000 [r105410] Russell Bryant * main/autoservice.c, /: Merged revisions 105409 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r105409 | russell | 2008-02-29 17:34:32 -0600 (Fri, 29 Feb 2008) | 23 lines Fix a major bug in autoservice. There was a race condition in the handling of the list of channels in autoservice. The problem was that it was possible for a channel to get removed from autoservice and destroyed, while the autoservice thread was still messing with the channel. This led to memory corruption, and caused crashes. This explains multiple backtraces I have seen that have references to autoservice, but do to the nature of the issue (memory corruption), could cause crashes in a number of areas. (fixes the crash in BE-386) (closes issue #11694) (closes issue #11940) The following issues could be related. If you are the reporter of one of these, please update to include this fix and try again. (potentially fixes issue #11189) (potentially fixes issue #12107) (potentially fixes issue #11573) (potentially fixes issue #12008) (potentially fixes issue #11189) (potentially fixes issue #11993) (potentially fixes issue #11791) ........ 2008-02-29 18:34 +0000 [r105378] Joshua Colp * configs/sip.conf.sample: Add documentation for setting username/password in SIP dial string. (closes issue #11587) Reported by: sobomax Patches: dialstring_doc.diff uploaded by sobomax (license 359) 2008-02-29 14:50 +0000 [r105263-105327] Philippe Sultan * /, res/res_jabber.c: Merged revisions 105326 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r105326 | phsultan | 2008-02-29 15:47:10 +0100 (Fri, 29 Feb 2008) | 1 line Fix a potential memory leak ........ * channels/chan_jingle.c, channels/chan_gtalk.c, res/res_jabber.c: Remove unnecessary if statements before calling iks_delete (redundant check is done inside iks_delete), thus making the code conform with coding guidelines. 2008-02-29 13:55 +0000 [r105262] Joshua Colp * /, apps/app_voicemail.c: Merged revisions 105261 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r105261 | file | 2008-02-29 09:48:13 -0400 (Fri, 29 Feb 2008) | 4 lines Bump up the size of the uniqueid variable. (closes issue #12107) Reported by: asgaroth ........ 2008-02-29 13:12 +0000 [r105210] Philippe Sultan * res/res_jabber.c: Automatically create new buddy upon reception of a presence stanza of type subscribed. (closes issue #12066) Reported by: ffadaie Patches: branch-1.4-12066-1.diff uploaded by phsultan (license 73) trunk-12066-1.diff uploaded by phsultan (license 73) Tested by: ffadaie, phsultan 2008-02-29 01:15 +0000 [r105176] Tilghman Lesher * contrib/init.d/rc.debian.asterisk, /: Merged revisions 105113 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r105113 | tilghman | 2008-02-28 15:56:54 -0600 (Thu, 28 Feb 2008) | 7 lines Update init script for LSB compat (closes issue #9843) Reported by: ibc Patches: rc.debian.asterisk.patch uploaded by ibc (license 211) Tested by: paravoid ........ 2008-02-28 22:39 +0000 [r105144] Russell Bryant * /, main/utils.c, include/asterisk/lock.h, utils/check_expr.c: Merged revisions 105116 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r105116 | russell | 2008-02-28 16:23:05 -0600 (Thu, 28 Feb 2008) | 8 lines Fix a bug in the lock tracking code that was discovered by mmichelson. The issue is that if the lock history array was full, then the functions to mark a lock as acquired or not would adjust the stats for whatever lock is at the end of the array, which may not be itself. So, do a sanity check to make sure that we're updating lock info for the proper lock. (This explains the bizarre stats on lock #63 in BE-396, thanks Mark!) ........ 2008-02-28 20:14 +0000 [r105060-105061] Mark Michelson * /, apps/app_queue.c: Merged revisions 105059 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r105059 | mmichelson | 2008-02-28 14:11:57 -0600 (Thu, 28 Feb 2008) | 6 lines When using autofill, members who are in use should be counted towards the number of available members to call if ringinuse is set to yes. Thanks to jmls who brought this issue up on IRC ........ * main/dial.c, /: Merged revisions 104841 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r104841 | mmichelson | 2008-02-27 15:49:20 -0600 (Wed, 27 Feb 2008) | 17 lines Two fixes: 1. Make the list of ast_dial_channels a lockable list. This is because in some cases, the ast_dial may exist in multiple threads due to asynchronous execution of its application, and I found some cases where race conditions could exist. 2. Check in ast_dial_join to be sure that the channel still exists before attempting to lock it, since it could have gotten hung up but the is_running_app flag on the ast_dial_channel may not have been cleared yet. (closes issue #12038) Reported by: jvandal Patches: 12038v2.patch uploaded by putnopvut (license 60) Tested by: jvandal ........ 2008-02-28 19:21 +0000 [r105006] Jason Parker * main/cdr.c, main/pbx.c, /: Merged revisions 105005 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r105005 | qwell | 2008-02-28 13:20:10 -0600 (Thu, 28 Feb 2008) | 9 lines Make pbx_exec pass an empty string into applications, if we get NULL. This protects against possible segfaults in applications that may try to use data before checking length (ast_strdupa'ing it, for example) (closes issue #12100) Reported by: foxfire Patches: 12100-nullappargs.diff uploaded by qwell (license 4) ........ 2008-02-28 14:42 +0000 [r104974] Tilghman Lesher * channels/chan_vpb.cc: Fix crash when configuration does not match hardware detection. (closes issue #12096) Reported by: mmickan Patches: chan_vpb.cc.diff uploaded by mmickan (license 400) 2008-02-28 04:37 +0000 [r104921] Jason Parker * /, channels/chan_skinny.c: Merged revisions 104920 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r104920 | qwell | 2008-02-27 22:31:21 -0600 (Wed, 27 Feb 2008) | 2 lines According to a video at www.cisco.com, the 7921G supports 6 line appearances. ........ 2008-02-28 00:11 +0000 [r104869] Tilghman Lesher * /, main/Makefile, build_tools/strip_nonapi: Merged revisions 104868 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r104868 | tilghman | 2008-02-27 18:05:06 -0600 (Wed, 27 Feb 2008) | 7 lines Compatibility fix for PPC64 (closes issue #12081) Reported by: jcollie Patches: asterisk-1.4.18-funcdesc.patch uploaded by jcollie (license 412) Tested by: jcollie, Corydon76 ........ 2008-02-27 23:58 +0000 [r104866] Russell Bryant * channels/chan_zap.c: reduce indentation in alloc_sub (issue #12079) Reported by: tzafrir Patches: alloc_sub uploaded by tzafrir (license 46) 2008-02-27 21:02 +0000 [r104788] Joshua Colp * /, apps/app_chanspy.c: Merged revisions 104787 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r104787 | file | 2008-02-27 16:56:23 -0400 (Wed, 27 Feb 2008) | 2 lines Don't loop around infinitely trying to spy on our own channel, and don't forget to free/detach the datastore upon hangup of the spy. ........ 2008-02-27 20:37 +0000 [r104784] Mark Michelson * /, main/file.c: Merged revisions 104783 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r104783 | mmichelson | 2008-02-27 14:36:26 -0600 (Wed, 27 Feb 2008) | 4 lines Bump a couple of more buffers up by 2 so that annoying warnings aren't generated like crazy on every fileexists_core call. ........ 2008-02-27 19:36 +0000 [r104756] Jason Parker * apps/app_voicemail.c: Remove useless 's' and 'key' variables, in favor of 'val', which serves the exact same purpose. 2008-02-27 18:20 +0000 [r104705] Tilghman Lesher * main/manager.c, /: Merged revisions 104704 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r104704 | tilghman | 2008-02-27 12:15:10 -0600 (Wed, 27 Feb 2008) | 2 lines Ensure the session ID can't be 0. ........ 2008-02-27 17:45 +0000 [r104687] Joshua Colp * /, main/file.c: Merged revisions 104665 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r104665 | file | 2008-02-27 13:41:40 -0400 (Wed, 27 Feb 2008) | 2 lines Bump up the buffer by 2. ........ 2008-02-27 17:36 +0000 [r104643] Russell Bryant * /, apps/app_chanspy.c: Merged revisions 104625 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r104625 | russell | 2008-02-27 11:33:04 -0600 (Wed, 27 Feb 2008) | 4 lines Fix a problem in ChanSpy where it could get stuck in an infinite loop without being able to detect that the calling channel hung up. (closes issue #12076, reported by junky, patched by me) ........ 2008-02-27 17:31 +0000 [r104617] Jason Parker * /, main/features.c: Merged revisions 104598 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r104598 | qwell | 2008-02-27 11:26:55 -0600 (Wed, 27 Feb 2008) | 8 lines Inherit language from the transfering channel on a blind transfer. (closes issue #11682) Reported by: caio1982 Patches: local_atxfer_lang3-1.4.diff uploaded by caio1982 (license 22) Tested by: caio1982, victoryure ........ 2008-02-27 17:12 +0000 [r104595-104597] Joshua Colp * /, main/loader.c: Merged revisions 104596 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r104596 | file | 2008-02-27 13:07:33 -0400 (Wed, 27 Feb 2008) | 4 lines Use the lock (which already existed, it just wasn't used) on the updaters list to protect the contents instead of the overall module list lock. (closes issue #12080) Reported by: ChaseVenters ........ * channels/chan_sip.c: After further discussion revert my previous commit for this. Currently in order to ensure devicestate is the expected value in another module (such as app_queue) then chan_sip must be loaded before hand. 2008-02-27 16:54 +0000 [r104594] Kevin P. Fleming * /, main/file.c: Merged revisions 104593 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r104593 | kpfleming | 2008-02-27 10:53:06 -0600 (Wed, 27 Feb 2008) | 8 lines fallback to standard English prompts properly when using new prompt directory layout (closes issue #11831) Reported by: IgorG Patches: fallbacken.v1.diff uploaded by IgorG (license 20) (modified by me to improve code and conform rest of function to coding guidelines) ........ 2008-02-27 16:26 +0000 [r104537-104539] Joshua Colp * channels/chan_sip.c: When queueing up a device state change when the peer is loaded from the configuration give it a state of not in use. We have to do this because the channel technology may not yet be registered so the state could not be queried and would be considered invalid. (closes issue #12087) Reported by: liorm * res/res_smdi.c, /: Merged revisions 104536 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r104536 | file | 2008-02-27 11:52:02 -0400 (Wed, 27 Feb 2008) | 4 lines Only stop the MWI monitor thread if it was actually started. (closes issue #12086) Reported by: francesco_r ........ 2008-02-27 15:34 +0000 [r104534] Tilghman Lesher * utils/astcanary.c: open(2) needs a mode argument when O_CREAT is specified. (Closes issue #12083) 2008-02-27 15:31 +0000 [r104533] Joshua Colp * channels/chan_sip.c, main/rtp.c: Fix T38 passthrough regression introduced by state changes. (closes issue #12078) Reported by: dimas Patches: v1-12078.patch uploaded by dimas (license 88) (closes issue #12074) Reported by: Ivan 2008-02-27 08:20 +0000 [r104502] Tilghman Lesher * channels/chan_vpb.cc, configs/vpb.conf.sample, include/asterisk/module.h: Bring Voicetronix driver up to date with current drivers (closes issue #12084) Reported by: mmickan Patches: chan_vpb.cc.diff uploaded by mmickan (license 400) module.h.diff uploaded by mmickan (license 400) vpb.conf.sample uploaded by mmickan (license 400) 2008-02-27 04:42 +0000 [r104419-104473] Russell Bryant * doc/janitor-projects.txt: note that the chan_sip conversion is already in progress * doc/janitor-projects.txt: add another janitor project * doc/janitor-projects.txt: Add the stuff from the janitor projects page that is still relevant. I figure that if we keep this in the tree, it will be much easier to keep up to date. The page on asterisk.org just links to this on svn.digium.com/view 2008-02-27 03:52 +0000 [r104418] Jason Parker * doc/janitor-projects.txt (added): Create placeholder file...for now. 2008-02-27 02:05 +0000 [r104388] Tilghman Lesher * apps/app_voicemail.c: Whitespace changes only 2008-02-27 01:16 +0000 [r104333-104335] Russell Bryant * /, apps/app_chanspy.c: Merged revisions 104334 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r104334 | russell | 2008-02-26 19:15:02 -0600 (Tue, 26 Feb 2008) | 3 lines Avoid some recursion in the cleanup code for the chanspy datastore (closes issue #12076, reported by junky, patched by me) ........ 2008-02-26 22:14 +0000 [r104301] Steve Murphy * res/snmp/agent.c: small change to allow this file to compile. No problem if you don't install the libsnmp package. 2008-02-26 20:33 +0000 [r104244-104270] Russell Bryant * main/asterisk.c: I swear I compiled this ... *cough* * res/res_phoneprov.c: fix this module, too * funcs/func_version.c: fix this module * Makefile, include/asterisk, build_tools/make_version_h (added): Re-add the automatically generated version.h, so that modules can include for making build time decisions for cross asterisk version compatibility * main/manager.c, channels/chan_sip.c, include/asterisk/version.h (removed), build_tools/make_version_c, res/res_agi.c, main/http.c, include/asterisk/ast_version.h (added): Rename version.h to ast_version.h. Next, I will be re-adding version.h as an automatically generated file like it used to be. This still needs to be there for modules that have to check it to compile against multiple asterisk versions. 2008-02-26 19:14 +0000 [r104215] Joshua Colp * main/cdr.c, main/pbx.c, include/asterisk/cdr.h, CHANGES: Add an 'e' option to ResetCDR which re-enables a CDR that has been disabled. (closes issue #11170) Reported by: kratzers Patches: ResetCDR.1.diff uploaded by kratzers (license 307) 2008-02-26 18:40 +0000 [r104176] Tilghman Lesher * doc/CODING-GUIDELINES: 1) Make braces mandatory for if/for/while, even around single statements. 2) Revise the argument parsing section, showing use of the standard macros. 3) Fix a typo. 2008-02-26 18:27 +0000 [r104140-104142] Jason Parker * Makefile, /: Merged revisions 104141 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r104141 | qwell | 2008-02-26 12:26:12 -0600 (Tue, 26 Feb 2008) | 1 line Add badshell to .PHONY target (thanks Kevin) ........ * Makefile, /: Merged revisions 104139 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r104139 | qwell | 2008-02-26 12:09:13 -0600 (Tue, 26 Feb 2008) | 2 lines Since all shells aren't as awesome as bash, we have to fail if somebody tries to use a literal "~" in DESTDIR. ........ 2008-02-26 16:51 +0000 [r104137] Olle Johansson * channels/chan_sip.c: Formatting and doxygen while waiting on an airport... 2008-02-26 16:36 +0000 [r104133-104136] Jason Parker * /, sounds/Makefile: Merged revisions 104135 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r104135 | qwell | 2008-02-26 10:35:06 -0600 (Tue, 26 Feb 2008) | 5 lines Revert previous abspath change. ...abspath is new in GNU make 3.81. I feel so...defeated. Must find new fix! ........ * /, sounds/Makefile: Merged revisions 104132 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r104132 | qwell | 2008-02-26 10:08:44 -0600 (Tue, 26 Feb 2008) | 9 lines Fix a very bizarre issue we were seeing with our buildbot when using a DESTDIR that wasn't an absolute path (such as DESTDIR=~/asterisk-1.4). Apparently what was happening, was that some of the targets were being expanded to the full path, so $@ ended up being /root/asterisk-1.4/[...]/ rather than ~/asterisk-1.4/[...]/ It appears that this may be a new "feature" in GNU make. (*cough* http://en.wikipedia.org/wiki/Principle_of_least_surprise *cough*) ........ 2008-02-26 14:51 +0000 [r104127] Mark Michelson * main/features.c: Remove more hardcoded pipe symbols and replace with commas. (closes issue #12072) Reported by: SimonSharman Patches: features.patch uploaded by SimonSharman (license 410) Tested by: SimonSharman 2008-02-26 06:43 +0000 [r104125] Tilghman Lesher * funcs/func_odbc.c: Use the readhandle for reads (closes issue #12069) 2008-02-26 00:38 +0000 [r104120-104124] Russell Bryant * res/res_smdi.c: Add a \todo to convert this module to the event system * CHANGES: Update CHANGES for SMDI stuff * channels/chan_zap.c, res/res_smdi.c, /, configs/smdi.conf.sample, include/asterisk/smdi.h, apps/app_voicemail.c: Merged revisions 104119 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r104119 | russell | 2008-02-25 18:25:29 -0600 (Mon, 25 Feb 2008) | 33 lines Merge changes from team/russell/smdi-1.4 This commit brings in a significant set of changes to the SMDI support in Asterisk. There were a number of bugs in the current implementation, most notably being that it was very likely on busy systems to pop off the wrong message from the SMDI message queue. So, this set of changes fixes the issues discovered as well as introducing some new ways to use the SMDI support which are required to avoid the bugs with grabbing the wrong message off of the queue. This code introduces a new interface to SMDI, with two dialplan functions. First, you get an SMDI message in the dialplan using SMDI_MSG_RETRIEVE() and then you access details in the message using the SMDI_MSG() function. A side benefit of this is that it now supports more than just chan_zap. For example, with this implementation, you can have some FXO lines being terminated on a SIP gateway, but the SMDI link in Asterisk. Another issue with the current implementation is that it is quite common that the station ID that comes in on the SMDI link is not necessarily the same as the Asterisk voicemail box. There are now additional directives in the smdi.conf configuration file which let you map SMDI station IDs to Asterisk voicemail boxes. Yet another issue with the current SMDI support was related to MWI reporting over the SMDI link. The current code could only report a MWI change when the change was made by someone calling into voicemail. If the change was made by some other entity (such as with IMAP storage, or with a web interface of some kind), then the MWI change would never be sent. The SMDI module can now poll for MWI changes if configured to do so. This work was inspired by and primarily done for the University of Pennsylvania. (also related to issue #9260) ........ 2008-02-25 23:56 +0000 [r104103-104110] Russell Bryant * channels/chan_zap.c, UPGRADE.txt: Deprecate the "stripmsd" option in favor of dialplan substring variable syntax. (closes issue #12060) * /, apps/app_chanspy.c: Merged revisions 104106 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r104106 | russell | 2008-02-25 17:42:42 -0600 (Mon, 25 Feb 2008) | 10 lines This patch fixes some pretty significant problems with how app_chanspy handles pointers to channels that are being spied upon. It was very likely that a crash would occur if the channel being spied upon hung up. This was because the current ast_channel handling _requires_ that the object is locked or else it could disappear at any time (except in the owning channel thread). So, this patch uses some channel datastore magic on the spied upon channel to be able to detect if and when the channel goes away. (closes issue #11877) (patch written by me, but thanks to kpfleming for the idea, and to file for review) ........ * /, main/utils.c: Merged revisions 104102 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r104102 | russell | 2008-02-25 17:19:05 -0600 (Mon, 25 Feb 2008) | 7 lines Improve the lock tracking code a bit so that a bunch of old locks that threads failed to lock don't sit around in the history. When a lock is first locked, this checks to see if the last lock in the list was one that was failed to be locked. If it is, then that was a lock that we're no longer sitting in a trylock loop trying to lock, so just remove it. (inspired by issue #11712) ........ 2008-02-25 23:04 +0000 [r104097-104101] Tilghman Lesher * cdr/cdr_pgsql.c, CHANGES: Permit additional CDR columns to be saved in Postgres. Note that these changes are backward-compatible, so no changes to UPGRADE.txt are necessary. (closes issue #9279) Reported by: rottenroddy Patches: 20080125__bug9279.diff.txt uploaded by Corydon76 (license 14) Tested by: Corydon76 * funcs/func_global.c: Shared space for variables (instead of letting other channels muck with your own) (closes issue #11943) Reported by: ramonpeek Patches: 20080208__bug11943__2.diff.txt uploaded by Corydon76 (license 14) Tested by: jmls * /, apps/app_voicemail.c: Merged revisions 104094 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r104094 | tilghman | 2008-02-25 15:31:47 -0600 (Mon, 25 Feb 2008) | 5 lines If the destination folder is full, don't delete a message when exiting. (closes issue #12065) Reported by: selsky Patch by: (myself) ........ 2008-02-25 21:40 +0000 [r104096] Joshua Colp * /, channels/chan_sip.c: Merged revisions 104095 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r104095 | file | 2008-02-25 17:37:20 -0400 (Mon, 25 Feb 2008) | 6 lines Make it so a users.conf user creates both a SIP peer and a SIP user. The user will be used for inbound authentication for the device, and peer will be used for placing calls to the device. (closes issue #9044) Reported by: queuetue Patches: sip-gui-friend.diff uploaded by qwell (license 4) ........ 2008-02-25 20:50 +0000 [r104093] Jason Parker * /, main/config.c: Merged revisions 104092 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r104092 | qwell | 2008-02-25 14:49:42 -0600 (Mon, 25 Feb 2008) | 11 lines Allow the use of #include and #exec in situations where the max include depth was only 1. Specifically, this fixes using #include and #exec in extconfig.conf. This was basically caused because the config file itself raises the include level to 1. I opted not to raise the include limit, because recursion here could cause very bizarre behavior. Pointed out, and tested by jmls (closes issue #12064) ........ 2008-02-25 19:02 +0000 [r104089] Joshua Colp * channels/chan_iax2.c: Instead of outputting a verbose message every so often let's make it a debug message. 2008-02-25 19:00 +0000 [r104088] Brett Bryant * doc/siptls.txt, configs/sip.conf.sample: Adding more tls configuration details to sip.conf sample, with a list of valid ciphers provided in both files. .. First commit since July, woot 2008-02-25 18:38 +0000 [r104087] Russell Bryant * /, channels/chan_agent.c: Merged revisions 104086 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r104086 | russell | 2008-02-25 12:38:10 -0600 (Mon, 25 Feb 2008) | 4 lines Ensure that the channel doesn't disappear in agent_logoff(). If it does, it could cause a crash. (fixes the crash reported in BE-396) ........ 2008-02-25 16:18 +0000 [r104081-104085] Joshua Colp * /, channels/chan_sip.c: Merged revisions 104084 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r104084 | file | 2008-02-25 12:16:13 -0400 (Mon, 25 Feb 2008) | 6 lines If a resubscription comes in for a dialog we no longer know about tell the remote side that the dialog does not exist so they subscribe again using a new dialog. (closes issue #10727) Reported by: s0l4rb03 Patches: 10727-2.diff uploaded by file (license 11) ........ * /, channels/chan_sip.c: Merged revisions 104082 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r104082 | file | 2008-02-25 11:17:18 -0400 (Mon, 25 Feb 2008) | 6 lines Due to recent changes tag will no longer be NULL if not present so we have to use ast_strlen_zero to see if it's actually blank. (closes issue #12061) Reported by: flefoll Patches: chan_sip.c.br14.patch_pedantic_no_totag uploaded by flefoll (license 244) ........ * res/res_config_pgsql.c: Fix building of trunk. dbpass is always going to exist. 2008-02-24 02:37 +0000 [r104073-104074] Steve Murphy * channels/chan_sip.c: Enforce a space between function args as per code review. * res/res_config_pgsql.c: On a 64-bit machine, with dev-mode turned on, and pgsql installed, I get warnings that stops the compile. They are fixed now. 2008-02-22 23:56 +0000 [r104045] Doug Bailey * channels/chan_zap.c, configure, configure.ac: Add protection to chan_zap build when NEONMWI events are not defined 2008-02-22 22:55 +0000 [r104036-104039] Tilghman Lesher * doc/manager_1_1.txt, main/manager.c, UPGRADE.txt, CHANGES, include/asterisk/manager.h: Move Originate to a separate privilege and require the additional System privilege to call out to a subshell. * /, channels/chan_sip.c: Merged revisions 104037 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r104037 | tilghman | 2008-02-22 16:45:14 -0600 (Fri, 22 Feb 2008) | 6 lines Backwards debug message. (closes issue #12052) Reported by: flefoll Patches: chan_sip.c.br14.patch_found-notfound uploaded by flefoll (license 244) ........ * res/res_config_pgsql.c: Allow database password to be NULL and several other cleanups. (closes issue #12048) Reported by: bukaj Patches: 20080222__bug12048.diff.txt uploaded by Corydon76 (license 14) Tested by: bukaj 2008-02-21 21:27 +0000 [r104031] Russell Bryant * channels/chan_sip.c: fix a typo 2008-02-21 21:09 +0000 [r104025-104029] Mark Michelson * res/res_agi.c: Instead of a notice, make the message about a hung-up channel a debug message, and revert the original logic on the if statement. Thanks to Juggie for bringing this to my attention. 2008-02-21 17:38 +0000 [r104024] Doug Bailey * channels/chan_zap.c: Added configuration distinction between neon and fsk mwi detection Add the detection for neon MWI events got rid of extraneous handle_init_event call in monitor loop 2008-02-21 16:46 +0000 [r104020] Mark Michelson * res/res_agi.c: Don't print the fact that we are using dead mode in AGI if called from the 'h' extension since it is well-known that it will be running in dead mode. (closes issue #12046) Reported by: explidous 2008-02-21 16:44 +0000 [r104019] Joshua Colp * configure, include/asterisk/autoconfig.h.in, configure.ac: Disable epoll as it has caused more obscure issues then any of my previous code. I will continue to work on it in a separate branch to make it stable for a release and test it against the following issues. (closes issue #11253) Reported by: falves11 (closes issue #11657) Reported by: davevg (closes issue #11033) Reported by: falves11 2008-02-21 14:44 +0000 [r104016] Kevin P. Fleming * main/manager.c, /: Merged revisions 104015 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r104015 | kpfleming | 2008-02-21 08:33:51 -0600 (Thu, 21 Feb 2008) | 2 lines reduce the likelihood that HTTP Manager session ids will consist of primarily '1' bits ........ 2008-02-21 05:21 +0000 [r104014] Tilghman Lesher * utils/astman.c: Ignore some more unused generated events. (closes issue #12042) Reported by: junky Patches: astman_events.diff uploaded by junky (license 177) 2008-02-20 Russell Bryant * Asterisk 1.6.0-beta4 released. 2008-02-20 22:34 +0000 [r103957] Mark Michelson * /, apps/app_queue.c: Merged revisions 103956 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r103956 | mmichelson | 2008-02-20 16:32:22 -0600 (Wed, 20 Feb 2008) | 8 lines Clear up confusion when viewing the QUEUE_WAITING_COUNT of a "dead" realtime queue. Since from the user's perspective, the queue does exist, we shouldn't tell them we couldn't find the queue. Instead since it is a dead queue, report a 0 waiting count This issue was brought up on IRC by jmls ........ 2008-02-20 22:29 +0000 [r103954-103955] Joshua Colp * channels/chan_h323.c: Try to do Packet2Packet bridging with chan_h323 if reinviting isn't enabled. (closes issue #11901) Reported by: pj * channels/chan_zap.c, /: Merged revisions 103953 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r103953 | file | 2008-02-20 18:06:59 -0400 (Wed, 20 Feb 2008) | 6 lines Don't wait for additional digits when overlap dialing is enabled if the setup message contains the sending_complete information element. (closes issue #11785) Reported by: klaus3000 Patches: sending_complete_overlap_asterisk-1.4.17.patch.txt uploaded by klaus3000 (license 65) ........ 2008-02-20 21:41 +0000 [r103908] Mark Michelson * channels/chan_local.c, /: Merged revisions 103904 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r103904 | mmichelson | 2008-02-20 15:40:08 -0600 (Wed, 20 Feb 2008) | 6 lines Fix a crash if the channel becomes NULL while attempting to lock it. (closes issue #12039) Reported by: danpwi ........ 2008-02-20 21:36 +0000 [r103903] Jason Parker * include/asterisk/dsp.h, main/dsp.c: Largely refactor DSP tone detection routines. Separate fax detection from digit detected. Added CED (called) tone detection for fax (previously, only CNG (calling) was supported). Separate DTMF/MF code paths where appropriate. Allow detection of arbitary tones. (closes issue #11796) Reported by: dimas Patches: v6-dsp-faxtones.patch uploaded by dimas (license 88) Tested by: dimas, IgorG, Cache 2008-02-20 21:08 +0000 [r103902] Mark Michelson * apps/app_voicemail.c: Fix a crash due to the wrong variable being used when building a directory string. (closes issue #12027) Reported by: jaroth Patches: forward.patch uploaded by jaroth (license 50) Tested by: jaroth 2008-02-20 18:29 +0000 [r103846-103847] Tilghman Lesher * include/asterisk/sched.h: Add some documentation fixups * /, main/stdtime/localtime.c: Merged revisions 103845 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r103845 | tilghman | 2008-02-20 11:53:00 -0600 (Wed, 20 Feb 2008) | 7 lines Compat fix for Solaris (closes issue #12022) Reported by: asgaroth Patches: 20080219__bug12022.diff.txt uploaded by Corydon76 (license 14) Tested by: asgaroth ........ 2008-02-20 15:21 +0000 [r103844] Mark Michelson * res/res_monitor.c: Fix another spot where a hard-coded '|' hadn't been converted to ',' (closes issue #12034) Reported by: kowalma 2008-02-20 03:52 +0000 [r103838-103842] Joshua Colp * main/audiohook.c: *mumble* * main/audiohook.c: file not found. * main/audiohook.c: Minor test... 2008-02-20 00:49 +0000 [r103833] Mark Michelson * apps/app_voicemail.c: When using IMAP storage, if the folder you attempt to save to does not exist, create it first. (closes issue #12032) Reported by: jaroth Patches: createfolder.patch uploaded by jaroth (license 50) Tested by: jaroth 2008-02-19 22:35 +0000 [r103831-103832] Jason Parker * main/channel.c: Make sure to mask out non-audio first as well * main/channel.c: Maybe we should set the value before we test it? Fixes an issue people have been seeing (unreported?) with file playback not working. 2008-02-19 21:54 +0000 [r103824-103828] Joshua Colp * main/loader.c: Add a log message that appears when you try to unload a module that isn't loaded. (closes issue #12033) Reported by: jamesgolovich Patches: asterisk-loader.diff.txt uploaded by jamesgolovich (license 176) * main/file.c: Only output a log message saying the format does not exist if it actually does not exist, not if the file itself could not be opened. (closes issue #11828) Reported by: IgorG Patches: readfile.v1.diff uploaded by IgorG (license 20) * /, channels/h323/ast_h323.cxx: Merged revisions 103823 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r103823 | file | 2008-02-19 16:28:08 -0400 (Tue, 19 Feb 2008) | 6 lines Send CallerID Name in setup message. (closes issue #11241) Reported by: tusar Patches: h323id_as_callerid_name.patch uploaded by tusar (license 344) ........ 2008-02-19 20:06 +0000 [r103822] Russell Bryant * channels/chan_local.c, /: Merged revisions 103821 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r103821 | russell | 2008-02-19 14:02:49 -0600 (Tue, 19 Feb 2008) | 8 lines Account for the fact that the "other" channel can disappear while the local pvt is not locked. (fixes a problem introduced in rev 100581) (closes issue #12012) Reported by: stevedavies Patch by me ........ 2008-02-19 19:27 +0000 [r103819-103820] Joshua Colp * apps/app_authenticate.c: len already contains the position we want to examine, if we move one left again we'll actually probably be looking at a digit. (issue #12030) Reported by: alligosh * apps/app_channelredirect.c, UPGRADE.txt, CHANGES: Add CHANNELREDIRECT_STATUS variable to ChannelRedirect() dialplan application. This will either be set to NOCHANNEL if the given channel was not found or SUCCESS if it worked. (closes issue #11553) Reported by: johan Patches: UPGRADE.txt.channelredirect.patch uploaded by johan (license 334) CHANGES.channelredirect.patch uploaded by johan (license 334) app_channelredirect-20080219.patch uploaded by johan (license 334) 2008-02-19 18:14 +0000 [r103818] Jeff Peeler * channels/chan_zap.c: (closes issue #11864) Reported by: julianjm Patches: chan_zap.c-1.4-devicestate-v1.diff uploaded by julianjm (license 99) Patch fixes problem of device state incorrectly reporting idle before PBX answers incoming call on FXO channel. Device status is updated now during new channel creation. 2008-02-19 17:33 +0000 [r103808-103813] Joshua Colp * /, configure, configure.ac: Merged revisions 103812 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r103812 | file | 2008-02-19 13:31:32 -0400 (Tue, 19 Feb 2008) | 4 lines Don't look for launchd when cross compiling. (closes issue #12029) Reported by: ovi ........ 2008-02-19 00:59 +0000 [r103805] Tilghman Lesher * main/say.c: Change verbosity into debug for Hebrew (and various whitespace fixes) (Closes issue #12011) 2008-02-18 23:58 +0000 [r103798-103802] Joshua Colp * main/channel.c, /: Merged revisions 103801 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r103801 | file | 2008-02-18 19:56:48 -0400 (Mon, 18 Feb 2008) | 10 lines Ensure that emulated DTMFs do not get interrupted by another begin frame. (closes issue #11740) Reported by: gserra Patches: v1-11740.patch uploaded by dimas (license 88) (closes issue #11955) Reported by: tsearle (closes issue #10530) Reported by: xmarksthespot ........ * main/channel.c, main/frame.c, channels/chan_sip.c, include/asterisk/channel.h, include/asterisk/frame.h: Add a non-invasive API for application level manipulation of T38 on a channel. This uses control frames (so they can even pass across IAX2) to negotiate T38 and provided a way of getting the current status of T38 using queryoption. This should by no means cause any issues and if it does I will take responsibility for it. (closes issue #11873) Reported by: dimas Patches: v4-t38-api.patch uploaded by dimas (license 88) * main/frame.c: Add some missing control frames. 2008-02-18 22:33 +0000 [r103796] Jason Parker * channels/chan_zap.c, /: Merged revisions 103795 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r103795 | qwell | 2008-02-18 16:28:56 -0600 (Mon, 18 Feb 2008) | 1 line Fix previous commit so that we actually disable echocanbridged if echocancel is off. ........ 2008-02-18 21:57 +0000 [r103794] Matthew Fredrickson * channels/chan_zap.c: Commit chan_zap portion of #11964: add the ability to get ORIG_CALLED_NUM 2008-02-18 21:30 +0000 [r103791] Jason Parker * channels/chan_zap.c, /: Merged revisions 103790 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r103790 | qwell | 2008-02-18 15:23:32 -0600 (Mon, 18 Feb 2008) | 4 lines Correct a message when echocancelwhenbridged is on, but echocancel is not. Closes issue #12019 ........ 2008-02-18 20:58 +0000 [r103788] Matthew Fredrickson * channels/chan_zap.c: Make sure EC is enabled when SS7 call comes in. Also add support for multiple DPCs per linkset. #11779 2008-02-18 20:53 +0000 [r103787] Mark Michelson * /, main/app.c: Merged revisions 103786 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r103786 | mmichelson | 2008-02-18 14:52:09 -0600 (Mon, 18 Feb 2008) | 10 lines There was an invalid assumption when calculating the duration of a file that the filestream in question was created properly. Unfortunately this led to a segfault in the situation where an unknown format was specified in voicemail.conf and a voicemail was recorded. Now, we first check to be sure that the stream was written correctly or else assume a zero duration. (closes issue #12021) Reported by: jakep Tested by: putnopvut ........ 2008-02-18 19:47 +0000 [r103783] Michiel van Baak * main/asterisk.c: make the output of 'core show settings' a bit nicer. (closes issue #12020) Reported by: seanbright Patches: asterisk.c.patch uploaded by seanbright (license 71) 2008-02-18 17:45 +0000 [r103781] Tilghman Lesher * /, channels/chan_sip.c, main/rtp.c: Merged revisions 103780 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r103780 | tilghman | 2008-02-18 11:31:52 -0600 (Mon, 18 Feb 2008) | 9 lines When a SIP channel is being auto-destroyed, it's possible for it to still be in bridge code. When that happens, we crash. Delay the RTP destruction until the bridge is ended. (closes issue #11960) Reported by: norman Patches: 20080215__bug11960__2.diff.txt uploaded by Corydon76 (license 14) Tested by: norman ........ 2008-02-18 Russell Bryant * Asterisk 1.6.0-beta3 released. 2008-02-18 17:12 +0000 [r103772] Olle Johansson * main/channel.c, channels/chan_sip.c: Make sure we can set up calls without audio (text+video). And ... it works! 2008-02-18 16:40 +0000 [r103771] Mark Michelson * channels/chan_zap.c, /: Merged revisions 103770 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r103770 | mmichelson | 2008-02-18 10:37:31 -0600 (Mon, 18 Feb 2008) | 10 lines Fix a linked list corruption that under the right circumstances could lead to a looped list, meaning it will traverse forever. (closes issue #11818) Reported by: michael-fig Patches: 11818.patch uploaded by putnopvut (license 60) Tested by: michael-fig ........ 2008-02-18 16:13 +0000 [r103764-103769] Joshua Colp * apps/app_channelredirect.c, main/pbx.c, include/asterisk/pbx.h: Add an API call (ast_async_parseable_goto) which parses a goto string and does an async goto instead of an explicit goto. (closes issue #11753) Reported by: johan * /, channels/chan_sip.c: Merged revisions 103763 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r103763 | file | 2008-02-18 11:33:14 -0400 (Mon, 18 Feb 2008) | 2 lines Don't care if the extension given doesn't exist for subscription based MWI. ........ 2008-02-18 10:10 +0000 [r103755] Olle Johansson * CHANGES, channels/chan_iax2.c: - No space in manager event names, please - Add new event to CHANGES 2008-02-18 04:43 +0000 [r103754] Tilghman Lesher * build_tools/cflags.xml, main/channel.c, main/pbx.c, funcs/func_channel.c, include/asterisk/channel.h, CHANGES, main/cli.c: Context tracing for channels (closes issue #11268) Reported by: moy Patches: chantrace-datastored-encapsulated-rev94934.patch uploaded by moy (license 222) 2008-02-16 21:22 +0000 [r103750] Michiel van Baak * channels/chan_skinny.c: move two ast_log calls to ast_debug. Now monitoring chan_skinny port with nagios or zabbix wont generate noise on the console. @ok tilghman 2008-02-15 23:32 +0000 [r103742] Russell Bryant * /, channels/chan_iax2.c: Merged revisions 103741 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r103741 | russell | 2008-02-15 17:31:39 -0600 (Fri, 15 Feb 2008) | 8 lines Fix a crash in chan_iax2 due to a race condition (closes issue #11780) Reported by: guillecabeza Patches: bug_iax2_jb_1.4.patch uploaded by guillecabeza (license 380) bug_iax2_jb_trunk.patch uploaded by guillecabeza (license 380) ........ 2008-02-15 23:20 +0000 [r103740] Mark Michelson * CHANGES: Document GotoIfTime change from svn revision 103738 2008-02-15 23:14 +0000 [r103739] Russell Bryant * include/asterisk/aes.h: Fix a regression in Asterisk 1.6 related to the use of AES encryption. 1024 was used instead of 128 when using AES from OpenSSL. Many thanks to d1mas for figuring this one out! (closes issue #11946) Reported by: bbhoss Patches: v1-11946.patch uploaded by dimas (license 88) 2008-02-15 23:07 +0000 [r103737-103738] Mark Michelson * main/pbx.c: Add proper "false" case behavior to GotoIfTime (closes issue #11719) Reported by: kshumard Patches: gotoiftime.twobranches.patch uploaded by kshumard (license 92) Tested by: kshumard * apps/app_voicemail.c: Fix redeclaration of variables when using IMAP storage (closes issue #11988) Reported by: jaroth Patches: variable_cleanup.patch uploaded by jaroth (license 50) 2008-02-15 19:50 +0000 [r103727-103729] Russell Bryant * /, main/loader.c: Merged revisions 103728 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r103728 | russell | 2008-02-15 13:50:11 -0600 (Fri, 15 Feb 2008) | 4 lines In the case that you try to directly reload a module has returned AST_MODULE_LOAD_DECLINE, log a message indicating that the module is not fully initialized and must be initialized using "module load". ........ * /, main/loader.c: Merged revisions 103726 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r103726 | russell | 2008-02-15 12:33:29 -0600 (Fri, 15 Feb 2008) | 6 lines Don't attempt to execute the reload callback for a module that returned AST_MODULE_LOAD_DECLINE. This fixes a crash that was reported against chan_console in trunk. (closes issue #11953, reported by junky, fixed by me) ........ 2008-02-15 17:32 +0000 [r103725] Mark Michelson * doc/tex/imapstorage.tex, /, configure, configure.ac: Merged revisions 103722 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r103722 | mmichelson | 2008-02-15 11:26:37 -0600 (Fri, 15 Feb 2008) | 8 lines Final round of changes for configure script logic for IMAP Now if a directory is specified, then we will search that directory for a source installation of the IMAP toolkit. If none is found, then we will use that directory as the basis for detecting a package installation of the IMAP c-client. If that check fails, then configure will fail. ........ 2008-02-15 17:29 +0000 [r103723] Jason Parker * channels/chan_zap.c, channels/chan_sip.c, res/res_phoneprov.c, include/asterisk/extconf.h, channels/misdn/isdn_msg_parser.c, apps/app_queue.c, channels/misdn/isdn_lib.c, main/config.c, main/channel.c, res/res_config_curl.c, channels/misdn/isdn_lib.h, main/ast_expr2f.c, channels/misdn/ie.c, channels/misdn/chan_misdn_config.h, channels/misdn/portinfo.c, include/asterisk/strings.h, res/res_config_ldap.c, include/asterisk/time.h: Fix up some doxygen issues. (closes issue #11996) Patches: bug_11996_doxygen.diff uploaded by snuffy (license 35) 2008-02-15 15:45 +0000 [r103716] Tilghman Lesher * utils/conf2ael.c: Remove extraneous copy (closes issue #12002) Reported by: junky Patches: conf2ael.diff uploaded by junky (license 177) 2008-02-15 15:11 +0000 [r103699-103715] Mark Michelson * configure, configure.ac: Merging of changes from 1.4 revision 103713. * doc/tex/imapstorage.tex, configure, configure.ac: Same changes as made to 1.4 in revision 103710 * doc/tex/imapstorage.tex: Trunk version of 1.4's imap documentation updates * configure, configure.ac: See commit message for svn revision 103698. This behavior is the same as what is described there. The difference is that trunk already had the --with-imap=system option, but it only checked the include path for headers in the imap directory and not also the c-client directory. 2008-02-14 21:21 +0000 [r103694] Jason Parker * configure, include/asterisk/autoconfig.h.in, configure.ac: Modify ldap autoconf function, so that an incorrect ldap library is not found on Solaris (it is incompatible). Also removes second check for awk, which causes Solaris to find an incompatible version of awk. (closes issue #11829) Reported by: snuffy Patches: bug-11829.diff uploaded by snuffy (license 35) 2008-02-14 21:04 +0000 [r103687-103691] Mark Michelson * /, apps/app_voicemail.c: Merged revisions 103690 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r103690 | mmichelson | 2008-02-14 15:03:02 -0600 (Thu, 14 Feb 2008) | 3 lines Fix build for non-IMAP builds ........ * /, apps/app_voicemail.c: Merged revisions 103688 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r103688 | mmichelson | 2008-02-14 14:55:48 -0600 (Thu, 14 Feb 2008) | 9 lines Fix the new message count if delete=yes when using IMAP storage. (closes issue #11406) Reported by: jaroth Patches: deleteflag_v2.patch uploaded by jaroth (license 50) Tested by: jaroth ........ * configs/queues.conf.sample, UPGRADE.txt, apps/app_queue.c: Change the queue holdtime announcement to happen at any interval (not just greater than two minutes). Remove the saying of less-than for holdtime announcements since it can lead to awkward holdtime announcements. Using '1' as a queue-round-seconds value is no longer valid. (closes issue #9736) Reported by: caio1982 Patches: queue_announce5.diff uploaded by caio1982 (license 22) Tested by: caio1982, putnopvut 2008-02-14 19:52 +0000 [r103685] Jason Parker * /, funcs/func_cdr.c: Merged revisions 103683 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r103683 | qwell | 2008-02-14 13:51:10 -0600 (Thu, 14 Feb 2008) | 5 lines Document the 'l' option to the CDR() function. (Thanks voipgate for pointing out the option, and Leif for providing text for it.) Closes issue #11695. ........ 2008-02-14 19:47 +0000 [r103682] Jeff Peeler * apps/app_externalivr.c: a few syntax changes and safer code 2008-02-14 18:39 +0000 [r103677] Jason Parker * channels/chan_iax2.c: Add periodic jitter stats to CLI and manager. (closes issue #8188) Reported by: stevedavies Patches: jblogging-trunk.patch uploaded by stevedavies jblogging-trunk_wmgrevent.patch uploaded by johann8384 updated_jbloggin-trunk_mgrevent.patch uploaded by johann8384 (license 190) (with additional changes by me) Tested by: stevedavies, johann8384 2008-02-14 10:19 +0000 [r103668] Olle Johansson * res/res_agi.c, apps/app_externalivr.c: Formatting fixes 2008-02-13 21:04 +0000 [r103662] Jeff Peeler * apps/app_externalivr.c: (closes issue #11825) Reported by: ctooley Patches: additional_eivr_commands.patch uploaded by ctooley (license 136) Tested by: ctooley 2008-02-13 15:47 +0000 [r103658] Mark Michelson * UPGRADE.txt, res/res_musiconhold.c: 1. Deprecate SetMusicOnHold and WaitMusicOnHold. 2. Add a duration parameter to MusicOnHold (closes issue #11904) Reported by: dimas Patches: v2-moh.patch uploaded by dimas (license 88) Tested by: dimas 2008-02-13 00:55 +0000 [r103559] Mark Michelson * main/event.c: Fix a small logic error in ast_event_iterator_next. The previous logic allowed for the iterator to indicate there was more data than there really was, causing the iterator read beyond the end of the event structure. This led to invalid memory reads and potential crashes. 2008-02-12 22:26 +0000 [r103447-103506] Jason Parker * main/manager.c: Even more sane permissions. This should be handled via a umask, like in many other places. * main/manager.c: Use slight more sane permissions 2008-02-12 15:39 +0000 [r103387-103388] Russell Bryant * main/asterisk.c: Remove development version notice. * main/manager.c: Fix build on *BSD. These permissions constants are not available there. 2008-02-12 15:13 +0000 [r103386] Joshua Colp * /, channels/chan_sip.c: Merged revisions 103385 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r103385 | file | 2008-02-12 11:09:24 -0400 (Tue, 12 Feb 2008) | 4 lines Even if no CallerID name or number has been provided by the remote party still use the configured sip.conf ones. (closes issue #11977) Reported by: pj ........ 2008-02-12 14:08 +0000 [r103341] Philippe Sultan * include/asterisk/jabber.h, res/res_jabber.c: Use an ast_flags structure in aji_client and aji_buddy rather than an integer. Modify calls to various ast_*_flag macros accordingly. 2008-02-12 00:24 +0000 [r103331] Jeff Peeler * main/manager.c, include/asterisk/config.h, CHANGES, main/config.c: Requested changes from Pari, reviewed by Russell. Added ability to retrieve list of categories in a config file. Added ability to retrieve the content of a particular category. Added ability to empty a context. Created new action to create a new file. Updated delete action to allow deletion by line number with respect to category. Added new action insert to add new variable to category at specified line. Updated action newcat to allow new category to be inserted in file above another existing category. 2008-02-11 22:10 +0000 [r103317-103325] Joshua Colp * /, apps/app_meetme.c: Merged revisions 103324 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r103324 | file | 2008-02-11 18:09:07 -0400 (Mon, 11 Feb 2008) | 4 lines If entering a conference with the 'w' option ensure that we can't listen or speak until the marked user appears. (closes issue #11835) Reported by: alanmcmillan ........ * res/res_agi.c: Remove ast_module_user usage from res_agi. This is taken care of in the core. * main/pbx.c, main/manager.c, main/translate.c, main/logger.c, main/app.c, main/utils.c, main/indications.c, main/asterisk.c, main/rtp.c: Just some minor coding style cleanup... * main/pbx.c: Fix Manager Redirect while in an AGI. (closes issue #10661) Reported by: junky 2008-02-11 17:09 +0000 [r103316] Kevin P. Fleming * /, configs/zapata.conf.sample: Merged revisions 103315 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r103315 | kpfleming | 2008-02-11 11:05:22 -0600 (Mon, 11 Feb 2008) | 2 lines improve 2BCT documentation a bit (thanks Jared) ........ 2008-02-11 16:17 +0000 [r103313-103314] Joshua Colp * main/channel.c, channels/chan_iax2.c: Add support for allowing a native bridge to happen when the L option is enabled. The RTP bridging could already handle this, it just needed to be enabled in the main bridging code. (issue #10647) Reported by: samdell3 * channels/chan_skinny.c: Change chan_skinny to use debug messages as appropriate. (closes issue #11967) Reported by: mvanbaak Patches: 2008021000-skinnydebug.diff.txt uploaded by mvanbaak (license 7) 2008-02-11 06:05 +0000 [r103306] James Golovich * channels/chan_sip.c: Don't wipe out transport and fd in chan_sip on reload (issue #11930) 2008-02-11 03:03 +0000 [r103282-103284] Mark Michelson * apps/app_queue.c: Fix improper indentation. Thanks again to snuffy for pointing it out. * apps/app_queue.c: Add a couple of comments to clarify the unreffing of queues. Thanks to snuffy for the idea. * main/event.c: Fix a problem regarding network vs. host byte order in the event API. ast_event_iterator_get_ie_type should return the ie type in host byte order. Furthermore, ast_event_get_ie_raw should already have its ie type argument in host byte order since it could be called externally (and it in fact is called in this way by ast_event_get_cached). 2008-02-09 11:27 +0000 [r103249] Michiel van Baak * apps/app_dial.c, apps/app_dictate.c, apps/app_echo.c, apps/app_authenticate.c, apps/app_disa.c, apps/app_chanisavail.c, apps/app_exec.c, apps/app_db.c, apps/app_controlplayback.c, apps/app_channelredirect.c, apps/app_directed_pickup.c, apps/app_dumpchan.c, apps/app_amd.c, apps/app_externalivr.c, apps/app_directory.c, apps/app_chanspy.c, apps/app_cdr.c: whitespace fixes only. 2008-02-09 06:33 +0000 [r103198] Tilghman Lesher * /, apps/app_voicemail.c: Merged revisions 103197 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r103197 | tilghman | 2008-02-09 00:23:49 -0600 (Sat, 09 Feb 2008) | 4 lines Commit fix for being unable to send voicemail from VoiceMailMain Reported by: William F Acker (via the -users mailing list) Patch by: Corydon76 (license 14) ........ 2008-02-08 21:26 +0000 [r103171] Russell Bryant * main/udptl.c, main/pbx.c, channels/chan_sip.c, channels/chan_iax2.c, res/res_jabber.c, apps/app_playback.c, main/rtp.c, channels/chan_usbradio.c, main/cdr.c, channels/chan_skinny.c, apps/app_minivm.c, res/res_agi.c, pbx/pbx_ael.c, pbx/pbx_dundi.c, funcs/func_devstate.c, apps/app_rpt.c, main/asterisk.c, channels/chan_mgcp.c, apps/app_voicemail.c: Merge changes from team/mvanbaak/cli-command-audit (closes issue #8925) About a year ago, as Leif Madsen and Jim van Meggelen were going over the CLI commands in Asterisk 1.4 for the next version of their book, they documented a lot of inconsistencies. This set of changes addresses all of these issues and has been reviewed by Leif. While this does introduce even more changes to the CLI command structure, it makes everything consistent, which is the most important thing. Thanks to all that helped with this one! 2008-02-08 18:58 +0000 [r103071-103122] Mark Michelson * apps/app_queue.c: Forgot that AST_LIST_REMOVE_CURRENT takes different arguments in trunk than 1.4. * /, apps/app_queue.c: Merged revisions 103120 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r103120 | mmichelson | 2008-02-08 12:48:17 -0600 (Fri, 08 Feb 2008) | 10 lines Prevent a potential three-thread deadlock. Also added a comment block to explicitly state the locking order necessary inside app_queue. (closes issue #11862) Reported by: flujan Patches: 11862.patch uploaded by putnopvut (license 60) Tested by: flujan ........ * /, channels/chan_iax2.c: Merged revisions 103070 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r103070 | mmichelson | 2008-02-08 12:00:38 -0600 (Fri, 08 Feb 2008) | 6 lines Yield the thread and return -1 if the ioctl fails for Zaptel timing device. (closes issue #11891) Reported by: tzafrir ........ 2008-02-08 16:49 +0000 [r103044] Russell Bryant * UPGRADE-1.2.txt (added), UPGRADE-1.4.txt (added), UPGRADE.txt: At the request of ManxPower, include the UPGRADE.txt from 1.2 and 1.4, as well. This way, if people need to go back and review what was deprecated in previous major releases, it is readily available to them. Thanks for the suggestion! 2008-02-08 15:31 +0000 [r102969-103018] Joshua Colp * channels/chan_sip.c: Fix a network byte order issue and ensure when creating an outgoing dialog that the socket always contains information such as type and port. (closes issue #11916) Reported by: mnnojd * /, channels/chan_iax2.c: Merged revisions 102968 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r102968 | file | 2008-02-08 11:08:20 -0400 (Fri, 08 Feb 2008) | 4 lines Make sure the presence of dbsecret is factored into user scoring. (closes issue #11952) Reported by: bbhoss ........ 2008-02-07 21:37 +0000 [r102933] Mark Michelson * apps/app_chanspy.c: This is a combination new feature/bug fix for app_chanspy. New feature: Add the 'e' option, which takes as an argument a list of interfaces separated by colons. This way, you will only be able to spy on this limited list of interfaces. Bug fix: change some pointer checks to ast_strlen_zero so that spying would work properly even if no channel was specified as the first argument to chanspy. (closes issue #10072) Reported by: xmarksthespot Patches: bugfix+newfeature10072patchtotrunkrev102726.diff uploaded by xmarksthespot (license 16) Tested by: xmarksthespot, mvanbaak 2008-02-07 21:08 +0000 [r102906-102908] Michiel van Baak * apps/app_adsiprog.c: whitespace fixes only * apps/app_alarmreceiver.c: There she goes! First commit from me to trunk \o/ Make app_alarmreceiver honor code guidelines and fix whitespace errors. No functional changes. 2008-02-07 20:02 +0000 [r102859] Jason Parker * /, main/features.c: Merged revisions 102858 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r102858 | qwell | 2008-02-07 13:53:55 -0600 (Thu, 07 Feb 2008) | 7 lines Specify which digit string was matched in debug message. (closes issue #11949) Reported by: dimas Patches: v1-feature-debug.patch uploaded by dimas (license 88) ........ 2008-02-07 16:47 +0000 [r102808] Kevin P. Fleming * /, configs/zapata.conf.sample: Merged revisions 102807 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r102807 | kpfleming | 2008-02-07 10:41:55 -0600 (Thu, 07 Feb 2008) | 2 lines document usage of 'transfer' configuration option for ISDN PRI switch-side transfers ........ 2008-02-06 20:12 +0000 [r102777] Mark Michelson * apps/app_queue.c: Add the channel's unique id to the AgentCalled manager event to make it more consistent with other manager events. 2008-02-06 18:01 +0000 [r102726] Joshua Colp * /, channels/chan_sip.c: Merged revisions 102725 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r102725 | file | 2008-02-06 13:59:23 -0400 (Wed, 06 Feb 2008) | 2 lines Only consider a T.38-only INVITE compatible if we have both a joint capability between us and them and if they provided T.38. ........ 2008-02-06 16:23 +0000 [r102700] Terry Wilson * funcs/func_realtime.c: Add REALTIME_STORE and REALTIME_DESTROY dialplan functions provided by sergee. I just added the ability to set multiple fields at once after discussions with Tilghman and Russell. Currently limited to 30 fields. (closes issue #11887) Reported by: sergee Patches: rt-func-store-destroy-multivalue.diff uploaded by otherwiseguy (license 396) Tested by: sergee, otherwiseguy 2008-02-06 15:20 +0000 [r102652] Russell Bryant * /, configs/features.conf.sample: Merged revisions 102651 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r102651 | russell | 2008-02-06 09:19:41 -0600 (Wed, 06 Feb 2008) | 3 lines Clarify setting DYNAMIC_FEATURES so that it gets inherited by outbound channels. (due to a discussion between me and a user via email) ........ 2008-02-06 03:05 +0000 [r102602] Tilghman Lesher * /, apps/app_voicemail.c: Merged revisions 102576 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r102576 | tilghman | 2008-02-05 18:26:02 -0600 (Tue, 05 Feb 2008) | 4 lines Move around some defines to unbreak ODBC storage. (closes issue #11932) Reported by: snuffy ........ 2008-02-06 00:08 +0000 [r102501-102550] Mark Michelson * apps/app_queue.c: Remove an extra debug message I left in * channels/chan_unistim.c, apps/app_dial.c, main/pbx.c, apps/app_privacy.c, apps/app_alarmreceiver.c, res/res_jabber.c, apps/app_followme.c, main/loader.c, channels/chan_usbradio.c, main/tcptls.c, res/res_agi.c, apps/app_minivm.c, apps/app_dumpchan.c, main/logger.c, apps/app_zapras.c, main/astmm.c: Get rid of any remaining ast_verbose calls in the code in favor of ast_verb (closes issue #11934) Reported by: mvanbaak Patches: 20080205_astverb-2.diff.txt uploaded by mvanbaak (license 7) * apps/app_voicemail.c: Change verbose messages to use the ast_verb macro. (closes issue #11931) Reported by: snuffy Patches: bug-11931.diff uploaded by snuffy (license 35) 2008-02-05 20:51 +0000 [r102500] Jason Parker * main/pbx.c: Change where priority of a goto is adjusted. Partially reverts 102272. Closes issue #11929 (credit to file for fix suggestion - we still <3 you) 2008-02-05 20:03 +0000 [r102454] Mark Michelson * /, channels/chan_mgcp.c: Merged revisions 102453 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r102453 | mmichelson | 2008-02-05 14:02:44 -0600 (Tue, 05 Feb 2008) | 8 lines Clear the DTMF buffer on hangup. (closes issue #11919) Reported by: eferro Patches: mgcp_dtmfclean_on_hangup.diff uploaded by eferro (license 337) Tested by: eferro ........ 2008-02-05 19:58 +0000 [r102379-102452] Joshua Colp * channels/chan_sip.c: Yeah yeah, I broke building on trunk. Shoot me. * /, channels/chan_sip.c: Merged revisions 102450 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r102450 | file | 2008-02-05 15:52:30 -0400 (Tue, 05 Feb 2008) | 3 lines If a REGISTER attempt comes in that is a retransmission of a previous REGISTER do not create a new nonce value. (issue #BE-381) ........ * /, res/res_clioriginate.c: Merged revisions 102378 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r102378 | file | 2008-02-05 11:09:29 -0400 (Tue, 05 Feb 2008) | 4 lines Perform dialing asynchronously when using the originate CLI command so the CLI does not appear to block. (closes issue #11927) Reported by: bbhoss ........ 2008-02-04 21:15 +0000 [r102329] Tilghman Lesher * utils/muted.c, /, configure, include/asterisk/autoconfig.h.in, configure.ac, main/asterisk.c: Merged revisions 102323 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r102323 | tilghman | 2008-02-04 15:06:09 -0600 (Mon, 04 Feb 2008) | 7 lines Cross-platform fix: OS X now deprecates the use of the daemon(3) API. (closes issue #11908) Reported by: oej Patches: 20080204__bug11908.diff.txt uploaded by Corydon76 (license 14) Tested by: Corydon76 ........ 2008-02-04 18:39 +0000 [r102297] Jason Parker * channels/chan_zap.c: Add line numbers to warning/error messages (and pretty up some existing ones). (closes issue #11894) Reported by: jmls Patches: chan_zap.patch uploaded by jmls (license 141) 2008-02-04 15:16 +0000 [r102272] Joshua Colp * main/pbx.c: Update handling of asyncgoto so it properly works on channels that are currently executing a PBX. (closes issue #11914) Reported by: arnd (closes issue #11753) Reported by: johan 2008-02-04 14:37 +0000 [r102262] Jason Parker * configs/extensions.ael.sample, configs/extensions.lua.sample: Change examples to use G here also. Closes issue #11875 2008-02-04 05:32 +0000 [r102190-102238] Tilghman Lesher * /, funcs/func_strings.c: Merged revisions 102214 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r102214 | tilghman | 2008-02-03 23:10:02 -0600 (Sun, 03 Feb 2008) | 6 lines Missing braces. (closes issue #11912) Reported by: dimas Patches: sprintf.patch uploaded by dimas (license 88) ........ * main/manager.c: CoreSettings and CoreStatus are missing the terminating "\r\n". Also, some miscellaneous spacing and initialization issues. (closes issue #11909) Reported by: srt Patches: patch-11909-2.diff uploaded by srt (license 378) Tested by: srt 2008-02-03 16:46 +0000 [r102091-102143] Olle Johansson * /, channels/chan_sip.c: Merged revisions 102142 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r102142 | oej | 2008-02-03 17:38:12 +0100 (Sön, 03 Feb 2008) | 8 lines Use the same CSEQ on CANCEL as on INVITE (according to RFC 3261) (closes issue #9492) Reported by: kryptolus Patches: bug9492.txt uploaded by oej (license 306) Tested by: oej ........ * /, channels/chan_sip.c: Merged revisions 102090 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r102090 | oej | 2008-02-03 11:37:32 +0100 (Sön, 03 Feb 2008) | 8 lines Handle ACK and CANCEL in an invite transaction - even if we get INFO transactions during the actual call setup. (closes issue #10567) Reported by: jacksch Tested by: oej Patch by: oej inspired by suggestions from neutrino88 in the bug tracker ........ 2008-02-03 06:43 +0000 [r102064] Russell Bryant * configure, configure.ac: Change the version number in the configure script from 1.4 to 1.6 2008-02-02 06:10 +0000 [r101990-102037] Russell Bryant * include/asterisk/event.h: The documentation page has to be in its own comment block to work, apparently. Fix it up! * /, channels/chan_sip.c: Merged revisions 101989 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r101989 | russell | 2008-02-01 17:06:32 -0600 (Fri, 01 Feb 2008) | 5 lines Change the SDP_SAMPLE_RATE macro. It turns out that even though G.722 is 16 kHz, it is supposed to specified as 8 kHz in the RTP, and RTP timestamps are supposed to be calculated based on 8 kHz. (Apparently this is due to a bug in a spec, but people follow it anyway, because it's the spec ...) ........ 2008-02-01 22:12 +0000 [r101873-101943] Tilghman Lesher * /, apps/app_voicemail.c: Merged revisions 101942 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r101942 | tilghman | 2008-02-01 15:54:28 -0600 (Fri, 01 Feb 2008) | 8 lines Fix the VM_DUR variable for forwarded voicemail, and fixed several other bugs while I'm in the area. (closes issue #11615) Reported by: jamessan Patches: 20071226__bug11615__2.diff.txt uploaded by Corydon76 (license 14) Tested by: Corydon76, jamessan ........ * /, configure, include/asterisk/autoconfig.h.in, configure.ac: Merged revisions 101894 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r101894 | tilghman | 2008-02-01 13:36:12 -0600 (Fri, 01 Feb 2008) | 2 lines Change detection of getifaddrs to use AST_C_COMPILE_CHECK, backported from trunk (as suggested by kpfleming) ........ * res/res_config_curl.c: Fix multi, when using the LIKE query. (closes issue #11889) Reported by: jmls Patches: res_config_curl.patch uploaded by jmls (license 141) Tested by: jmls 2008-02-01 18:24 +0000 [r101869] Jason Parker * apps/app_authenticate.c: Comparison, not set :) Thanks mvanbaak. 2008-02-01 18:08 +0000 [r101824] Tilghman Lesher * res/res_odbc.c, configs/res_odbc.conf.sample: Clarify the pooling functionality by changing the config file keyword 2008-02-01 17:44 +0000 [r101823] Jason Parker * /, apps/app_authenticate.c: Move an feof() call to before the fgets(). This would have exited the loop early if you had an authentication file with no newline at the end. 2008-02-01 17:28 +0000 [r101819-101821] Russell Bryant * /, apps/app_authenticate.c: Merged revisions 101818 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r101818 | russell | 2008-02-01 11:23:47 -0600 (Fri, 01 Feb 2008) | 4 lines Don't overwrite the last character of a line if it's not a newline. This would happen if the last line in the file doesn't have a newline. (pointed out by Qwell) ........ 2008-02-01 16:01 +0000 [r101773] Tilghman Lesher * /, configure, include/asterisk/autoconfig.h.in, configure.ac, main/acl.c: Merged revisions 101772 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r101772 | tilghman | 2008-02-01 09:55:58 -0600 (Fri, 01 Feb 2008) | 2 lines Compatibility fix for OpenWRT (reported by Brian Capouch via the mailing list) ........ 2008-02-01 06:27 +0000 [r101694-101746] Russell Bryant * apps/app_authenticate.c: simplify some code, tweak formatting, and reduce indentation * apps/app_authenticate.c: reduce a level of indentation * apps/app_channelredirect.c: Get rid of a goto where there was no extra cleanup happening at the exit point * /, channels/chan_iax2.c: Merged revisions 101693 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r101693 | russell | 2008-01-31 18:32:49 -0600 (Thu, 31 Jan 2008) | 8 lines Add some more sanity checking on IAX2 dial strings for the case that no peer or hostname was provided, which is the one part of the dial string that is absolutely required. If it's not there, bail out. (closes issue #11897) Reported by sokhapkin Patch by me ........ 2008-02-01 00:08 +0000 [r101650] Mark Michelson * /, apps/app_amd.c: Merged revisions 101649 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r101649 | mmichelson | 2008-01-31 18:06:37 -0600 (Thu, 31 Jan 2008) | 9 lines From bugtracker: "fix totalAnalysisTime to handle periods of no channel activity" (closes issue #9256) Reported by: cmaj Patches: amd-dont-wait-too-long-for-frames-take3.diff.txt uploaded by cmaj (license 111) Tested by: cmaj, skygreg, ZX81, rjain ........ 2008-01-31 23:14 +0000 [r101611] Russell Bryant * /, main/translate.c, main/file.c: Merged revisions 101601 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r101601 | russell | 2008-01-31 17:10:06 -0600 (Thu, 31 Jan 2008) | 12 lines Fix a couple of places where ast_frfree() was not called on a frame that came from a translator. This showed itself by g729 decoders not getting released. Since the flag inside the translator frame never got unset by freeing the frame to indicate it was no longer in use, the translators never got destroyed, and thus the g729 licenses were not released. (closes issue #11892) Reported by: xrg Patches: 11892.diff uploaded by russell (license 2) Tested by: xrg, russell ........ 2008-01-31 22:12 +0000 [r101578-101580] Mark Michelson * apps/app_queue.c: Forgot an ! * apps/app_queue.c: A change I made to accommodate the "linear" strategy in trunk caused queue strategies to not be loaded from realtime queues. This commit fixes that. Thanks to jmls for pointing this problem out to me on IRC. This also contains some changes to S_OR where it should be used. Thanks to Qwell for pointing these out. 2008-01-31 21:33 +0000 [r101577] Russell Bryant * channels/chan_sip.c: Fix a simple deadlock that was introduced _right_ before this code got merged into trunk. (closes issue #11895, reported by pj, patched by me) 2008-01-31 21:31 +0000 [r101532-101576] Mark Michelson * apps/app_queue.c: Handle the case of a NULL state_interface when checking a realtime member. Thanks to jmls for finding this issue. * /, res/res_monitor.c: Merged revisions 101531 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r101531 | mmichelson | 2008-01-31 15:00:24 -0600 (Thu, 31 Jan 2008) | 10 lines 1. Prevent the addition of an extra '/' to the beginning of an absolute pathname. 2. If ast_monitor_change_fname is called and the new filename is the same as the old, then exit early and don't set the filename_changed field in the monitor structure. Setting it in this case was causing ast_monitor_stop to erroneously delete them. (closes issue #11741) Reported by: garlew Tested by: putnopvut ........ 2008-01-31 19:54 +0000 [r101483] Jason Parker * /, channels/chan_sip.c, channels/chan_iax2.c: Merged revisions 101482 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r101482 | qwell | 2008-01-31 13:52:49 -0600 (Thu, 31 Jan 2008) | 4 lines Solaris compat fixes for struct in_addr funkiness. Issue #11885, patch by snuffy. ........ 2008-01-31 19:43 +0000 [r101481] Steve Murphy * main/pbx.c, /: Merged revisions 101480 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r101480 | murf | 2008-01-31 12:30:37 -0700 (Thu, 31 Jan 2008) | 1 line closes issue #11845; that's the one where there's a 1004 byte cdr leak with every AMI Redirect to a zap channel ........ 2008-01-31 19:20 +0000 [r101416-101449] Russell Bryant * /, channels/chan_agent.c: Merged revisions 101433 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r101433 | russell | 2008-01-31 13:17:05 -0600 (Thu, 31 Jan 2008) | 2 lines Add more missing locking of the agents list ... ........ * /, channels/chan_agent.c: Merged revisions 101413-101414 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r101413 | russell | 2008-01-31 13:04:52 -0600 (Thu, 31 Jan 2008) | 2 lines Add missing locking to the find_agent() function. ........ r101414 | russell | 2008-01-31 13:07:46 -0600 (Thu, 31 Jan 2008) | 3 lines Move the locking from find_agent() into the agent dialplan function handler to ensure that the agent doesn't disappear while we're looking at it. ........ 2008-01-31 15:36 +0000 [r101393] Joshua Colp * funcs/func_realtime.c: Add missing braces. (closes issue #11886) Reported by: sergee Patches: func_realtime_fix-r101392.diff uploaded by sergee (license 138) 2008-01-31 05:28 +0000 [r101373] Russell Bryant * CHANGES: remove entry that is no longer in the tree 2008-01-30 23:10 +0000 [r101344] Mark Michelson * channels/chan_sip.c: The deprecation of "username" in favor of "defaultuser" for SIP peers unfortunately broke realtime configurations which still used the "username" field. This was taken care of properly when reading from realtime but was not handled properly when updating a realtime peer. This change also adds a deprecation NOTICE CLI message that will print if using the deprecated "username" field. (closes issue #11880) Reported by: cabal95 Patches: 11880.patch uploaded by putnopvut (license 60) Tested by: cabal95 2008-01-30 20:08 +0000 [r101322] Olle Johansson * configs/cli.conf.sample: Clarify configuration file that can be misunderstood 2008-01-30 19:03 +0000 [r101296] Jason Parker * apps/app_controlplayback.c: Allow disabling the default ffwd/rewind keys in the ControlPlayback application. This is done in a backward compat way. If the "default" key for ffwd/rew is used for another option (such as stop), the "default" is removed. (closes issue #11754) Reported by: johan Patches: app_controlplayback.c.option3.patch uploaded by johan (license 334) Tested by: johan, qwell 2008-01-30 17:12 +0000 [r101271] Olle Johansson * configs/rtppage.conf.sample (removed), apps/app_rtppage.c (removed): Removing applications that wasn't ready for svn trunk, as trunk now has pre-release status. 2008-01-30 16:54 +0000 [r101269] Tilghman Lesher * apps/app_voicemail.c: Make the VoicemailUsersList AMI command consistent with other manager list functions. (closes issue #11874) Reported by: srt Patches: voicemail_ami-11847.patch uploaded by srt (license 378) 2008-01-30 16:39 +0000 [r101267-101268] Olle Johansson * include/asterisk/rtp.h, main/rtp.c: - doxygen fixes - change function to void because it always returned the same value and no one read it. * main/rtp.c: Formatting fixes 2008-01-30 15:42 +0000 [r101224] Mark Michelson * apps/app_rtppage.c: Get trunk to compile 2008-01-30 15:42 +0000 [r101223] Joshua Colp * /, main/slinfactory.c: Merged revisions 101222 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r101222 | file | 2008-01-30 11:41:04 -0400 (Wed, 30 Jan 2008) | 4 lines Fix an issue where if a frame of higher sample size preceeded a frame of lower sample size and ast_slinfactory_read was called with a sample size of the combined values or higher a crash would happen. (closes issue #11878) Reported by: stuarth ........ 2008-01-30 15:36 +0000 [r101221] Olle Johansson * CHANGES: Update CHANGES with rtppage 2008-01-30 15:35 +0000 [r101220] Jason Parker * /, configs/extensions.conf.sample: Merged revisions 101219 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 (closes issue #11875) ........ r101219 | qwell | 2008-01-30 09:34:37 -0600 (Wed, 30 Jan 2008) | 5 lines Change default config to use descending channel order of groups, rather than ascending. Fixes a potential source of confusion in glare-type situations. Issue 11875, reported by JimVanM. ........ 2008-01-30 15:30 +0000 [r101218] Olle Johansson * configs/rtppage.conf.sample (added), apps/app_rtppage.c (added): Add rtppage() application to do multicast or unicast RTP paging to SIP phones. (closes issue #11797) Reported by: macbrody Patches: app_rtppage-20080130.c uploaded by macbrody (license 352) 2008-01-30 15:27 +0000 [r101217] Mark Michelson * /, apps/app_queue.c: Merged revisions 101216 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r101216 | mmichelson | 2008-01-30 09:23:00 -0600 (Wed, 30 Jan 2008) | 5 lines Fix a logic error with regards to autofill. Prior to this change, it was possible for a caller to go out of turn if autofill were enabled and callers ahead in the queue were attempting to call a member. This change fixes this. ........ 2008-01-30 12:48 +0000 [r101196] Kevin P. Fleming * channels/chan_sip.c: simplify this code and eliminate the return value cast that is no longer necessary 2008-01-30 11:27 +0000 [r101153-101154] Olle Johansson * channels/chan_sip.c, include/asterisk/channel.h: Constifying the interface to get pvt_ids in the bridge, based on suggestion from (const char *) Kevin. Thanks! * /, channels/chan_sip.c: Merged revisions 101152 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r101152 | oej | 2008-01-30 12:20:31 +0100 (Ons, 30 Jan 2008) | 7 lines Stop musiconhold on attended transfer. (closes issue #11872) Reported by: gareth Patches: svn-101018.patch uploaded by gareth (license 208) ........ 2008-01-30 00:58 +0000 [r101126] Jason Parker * CHANGES: Fix a typo 2008-01-30 00:04 +0000 [r101082] Russell Bryant * CHANGES, apps/app_speech_utils.c: Add the 'n' option to SpeechBackground, which has the application not answer the channel if it has not already been answered. (closes SPD-51) 2008-01-29 23:59 +0000 [r101081] Dwayne M. Hubbard * /, build_tools/make_version: Merged revisions 101080 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r101080 | dhubbard | 2008-01-29 17:50:42 -0600 (Tue, 29 Jan 2008) | 1 line updated build_tools to handle the autotag directory structure changes; changes related to BE-353. Patch by The Russell and reviewed by The Me. ........ 2008-01-29 23:02 +0000 [r101036] Mark Michelson * /, apps/app_queue.c: Merged revisions 101035 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r101035 | mmichelson | 2008-01-29 17:02:03 -0600 (Tue, 29 Jan 2008) | 3 lines Remove a memory leak from updating realtime queues ........ 2008-01-29 22:04 +0000 [r101018] Tilghman Lesher * res/res_config_curl.c: Oops, a sizeof error 2008-01-29 19:41 +0000 [r100974] Mark Michelson * /, apps/app_queue.c: Merged revisions 100973 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r100973 | mmichelson | 2008-01-29 13:39:00 -0600 (Tue, 29 Jan 2008) | 6 lines Fixing an erroneous return value returned when attempting to pause or unpause a queue member fails. Fixes BE-366, thanks to John Bigelow for writing the patch. ........ 2008-01-29 17:44 +0000 [r100933] Russell Bryant * /, main/Makefile: Merged revisions 100932 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r100932 | russell | 2008-01-29 11:43:41 -0600 (Tue, 29 Jan 2008) | 4 lines Fix the last couple of issues related to building from a path that contains spaces. (closes issue #11834) ........ 2008-01-29 17:42 +0000 [r100931] Jason Parker * /, channels/misdn_config.c: Merged revisions 100930 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r100930 | qwell | 2008-01-29 11:41:43 -0600 (Tue, 29 Jan 2008) | 6 lines Initialize an array to 0s if config option not specified. (closes issue #11860) Patches: misdn_get_config.v1.diff uploaded by IgorG (license 20) ........ 2008-01-29 17:22 +0000 [r100900-100928] Russell Bryant * Makefile, /: Merged revisions 100922 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r100922 | russell | 2008-01-29 11:21:33 -0600 (Tue, 29 Jan 2008) | 3 lines Use GNU make magic instead of shell magic to escape spaces in the working directory. (related to issue #11834) ........ * Makefile, /: Merged revisions 100882 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r100882 | russell | 2008-01-29 11:06:43 -0600 (Tue, 29 Jan 2008) | 6 lines Fix building Asterisk when the working path has spaces in it. (closes issue #11834) Reported by: spendergrass Patched by: me ........ 2008-01-29 16:14 +0000 [r100843] Jason Parker * channels/chan_zap.c, /: Merged revisions 100835 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r100835 | qwell | 2008-01-29 10:10:00 -0600 (Tue, 29 Jan 2008) | 5 lines Allow zap groups above 30 to work properly. (closes issue #11590) Reported by: tbsky ........ 2008-01-29 15:30 +0000 [r100833] Joshua Colp * channels/chan_sip.c: Make externip work as documented. If no port is specified it will use the value of bindport instead of always being 5060. (closes issue #11858) Reported by: hmodes 2008-01-29 10:50 +0000 [r100794-100795] Christian Richter * channels/chan_misdn.c, /: Merged revisions 100793 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r100793 | crichter | 2008-01-29 11:36:19 +0100 (Di, 29 Jan 2008) | 1 line fixed potential segfault in misdn show channels CLI command ........ * channels/chan_misdn.c, /: Merged revisions 96199 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r96199 | crichter | 2008-01-03 13:12:27 +0100 (Do, 03 Jan 2008) | 1 line make sure frame is completely clean, before we send it to asterisk as DTMF. If we don't make it clean, it happens that one way audio occurs.. ........ 2008-01-29 09:18 +0000 [r100741-100767] Olle Johansson * /, channels/chan_sip.c: Merged revisions 100740 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r100740 | oej | 2008-01-29 09:26:48 +0100 (Tis, 29 Jan 2008) | 8 lines (closes issue #11736) Reported by: MVF Patches: bug11736-2.diff uploaded by oej (license 306) Tested by: oej, MVF, revolution (russellb: This was the showstopper for the release.) ........ * channels/chan_sip.c: Removing code that wasn't supposed to be there at all, only at an experimental stage before I found another solution. Thanks Kevin, for reminding me. 2008-01-28 Russell Bryant * Asterisk 1.6.0-beta2 released. 2008-01-28 21:11 +0000 [r100679] Jason Parker * build_tools/menuselect-deps.in, configs/vpb.conf.sample (added), doc/tex/channelvariables.tex, makeopts.in: Reintroduce more chan_vpb stuff that was removed in r100421 and r100422 2008-01-28 21:07 +0000 [r100678] Mark Michelson * channels/chan_vpb.cc (added), configure, include/asterisk/autoconfig.h.in, configure.ac, channels/Makefile: Re-inserting chan_vpb into trunk. 2008-01-28 21:05 +0000 [r100677] Tilghman Lesher * main/pbx.c, /: Merged revisions 100675 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r100675 | tilghman | 2008-01-28 15:02:02 -0600 (Mon, 28 Jan 2008) | 2 lines WaitExten didn't handle AbsoluteTimeout properly (went to 't' instead of 'T') ........ 2008-01-28 21:02 +0000 [r100676] Jason Parker * /, apps/app_voicemail.c: Merged revisions 100672 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 (closes issue #11795) ........ r100672 | qwell | 2008-01-28 14:42:43 -0600 (Mon, 28 Jan 2008) | 7 lines When using ODBC_STORAGE, make sure we put greeting files into the database like we do with the others. Issue #11795 Reported by: dimas Patches: vmgreet.patch uploaded by dimas (license 88) ........ 2008-01-28 20:40 +0000 [r100632-100671] Joshua Colp * channels/chan_sip.c: Fix up some T38 state change issues. (closes issue #11630) Reported by: dimas Patches: v2-sip-t38state.patch uploaded by dimas (license 88) * channels/chan_sip.c: Fix up two scheduling issues. In one instance a scheduled item was not deleted when it should have been and in the other it was scheduled again when it shouldn't have been. 2008-01-28 18:41 +0000 [r100630-100631] Russell Bryant * main/features.c: Merge rev 100626 from Asterisk 1.4. The svnmerge of this commit was a NoOp, since res_features doesn't exist in trunk. Thanks to qwell for pointing it out! * /, channels/chan_sip.c: Merged revisions 100629 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r100629 | russell | 2008-01-28 12:34:20 -0600 (Mon, 28 Jan 2008) | 5 lines For some reason, the use of this strdupa() is leading to memory corruption on freebsd sparc64. This trivial workaround fixes it. (closes issue #10300, closes issue #11857, reported by mattias04 and Home-of-the-Brave) ........ 2008-01-28 18:27 +0000 [r100628] Tilghman Lesher * configure, include/asterisk/autoconfig.h.in, configure.ac, main/logger.c: Normalize the detection for execinfo, so that Linux (glibc) and other platforms with libexecinfo will generate inline stack backtraces correctly. 2008-01-28 18:27 +0000 [r100627] Russell Bryant * /: Merged revisions 100626 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r100626 | russell | 2008-01-28 12:26:31 -0600 (Mon, 28 Jan 2008) | 7 lines Fix a crash in ast_masq_park_call() (issue #11342) Reported by: DEA Patches: res_features-park.txt uploaded by DEA (license 3) ........ 2008-01-28 18:24 +0000 [r100625] Jason Parker * channels/chan_zap.c, /: Merged revisions 100624 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r100624 | qwell | 2008-01-28 12:23:09 -0600 (Mon, 28 Jan 2008) | 1 line Correct a comment which made little/no sense. ........ 2008-01-28 17:21 +0000 [r100565-100582] Russell Bryant * main/channel.c, channels/chan_local.c, /, include/asterisk/channel.h: Merged revisions 100581 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r100581 | russell | 2008-01-28 11:15:41 -0600 (Mon, 28 Jan 2008) | 9 lines Make some deadlock related fixes. These bugs were discovered and reported internally at Digium by Steve Pitts. - Fix up chan_local to ensure that the channel lock is held before the local pvt lock. - Don't hold the channel lock when executing the timing function, as it can cause a deadlock when using chan_local. This actually changes the code back to be how it was before the change for issue #10765. But, I added some other locking that I think will prevent the problem reported there, as well. ........ * main/pbx.c: Clean up some formatting, and simplify a bit of code using ast_str 2008-01-28 13:57 +0000 [r100549] Joshua Colp * channels/chan_sip.c: Don't do a network byte order conversion when setting the socket's port variable to that of bindaddr's. It is already in the correct network byte order. (closes issue #11800) Reported by: hmodes 2008-01-28 04:43 +0000 [r100514-100533] Russell Bryant * main/channel.c: Make a couple more uses of ARRAY_LEN, and convert some spaces to tabs * main/channel.c: - Simplify a line with ARRAY_LEN() - Make a few little formatting changes * main/channel.c: These readlocks always fail for me on my mac, and I saw it happen again today on another mac. We ignore the return value of locking operations almost everywhere in Asterisk. So, ignore these, as well, so Asterisk will actually work on systems where this is occurring while I look into what the issue is. 2008-01-27 23:14 +0000 [r100488-100497] Tilghman Lesher * channels/chan_sip.c, include/asterisk/sched.h, channels/chan_iax2.c: With the switch to the ast_sched_replace* API in trunk, we lose the correction that was just merged from 1.4, so this is a changeover to those APIs to use the macro versions, so that we properly detect errors from ast_sched_del, instead of simply ignoring the return values. * main/cdr.c, channels/chan_misdn.c, main/dnsmgr.c, /, channels/chan_sip.c, channels/chan_h323.c, include/asterisk/sched.h, main/file.c, pbx/pbx_dundi.c, channels/chan_iax2.c, main/rtp.c, channels/chan_mgcp.c: Merged revisions 100465 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r100465 | tilghman | 2008-01-27 15:59:53 -0600 (Sun, 27 Jan 2008) | 11 lines When deleting a task from the scheduler, ignoring the return value could possibly cause memory to be accessed after it is freed, which causes all sorts of random memory corruption. Instead, if a deletion fails, wait a bit and try again (noting that another thread could change our taskid value). (closes issue #11386) Reported by: flujan Patches: 20080124__bug11386.diff.txt uploaded by Corydon76 (license 14) Tested by: Corydon76, flujan, stuarth` ........ 2008-01-25 22:54 +0000 [r100421-100422] Jason Parker * doc/tex/channelvariables.tex: Get rid of that last little bit. * build_tools/menuselect-deps.in, configs/vpb.conf.sample (removed), makeopts.in: Remove more remnants of chan_vpb 2008-01-25 22:39 +0000 [r100419-100420] Mark Michelson * channels/chan_vpb.cc (removed), configure, include/asterisk/autoconfig.h.in, configure.ac, channels/Makefile, .cleancount: Removing chan_vpb from the tree 2008-01-25 21:26 +0000 [r100379] Jason Parker * /, channels/chan_sip.c: Merged revisions 100378 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r100378 | qwell | 2008-01-25 15:24:49 -0600 (Fri, 25 Jan 2008) | 2 lines This would have never been true, since we're passing (sizeof(req.data) - 1) as the len to recvfrom(). ........ 2008-01-25 20:51 +0000 [r100361] Kevin P. Fleming * apps/app_rpt.c: correct a real problem and silence an annoying compiler warning 2008-01-25 14:53 +0000 [r100344] Mark Michelson * apps/app_queue.c: Insure that we are not going to pass a NULL pointer to add_to_interfaces. (closes issue #11840) Reported by: junky 2008-01-25 02:52 +0000 [r100325] Joshua Colp * main/dial.c, include/asterisk/dial.h: Add an API call that steals the answered channel so that a destruction of the dialing structure does not hang it up. 2008-01-24 22:58 +0000 [r100307] Tilghman Lesher * Makefile, build_tools/make_defaults_h: Use the set ASTDBDIR as the default, too 2008-01-24 22:36 +0000 [r100305-100306] Kevin P. Fleming * include/asterisk/app.h: ummm... might be good if this macro argument was actually used :-) * include/asterisk/app.h: add the ability to define a structure type for argument parsing when it would be useful to be able to pass it between functions 2008-01-24 22:02 +0000 [r100266] James Golovich * channels/chan_sip.c: Fix simple whitespace issue 2008-01-24 22:01 +0000 [r100265] Kevin P. Fleming * include/asterisk/app.h, /: Merged revisions 100264 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r100264 | kpfleming | 2008-01-24 15:57:41 -0600 (Thu, 24 Jan 2008) | 2 lines make these macros not assume that the only other field in the structure is 'argc'... this is true when someone uses AST_DECLARE_APP_ARGS, but it's perfectly reasonable to define your own structure as long as it has the right fields ........ 2008-01-24 20:32 +0000 [r100245] Joshua Colp * main/features.c: Minor cosmetic change... 2008-01-24 18:35 +0000 [r100224] James Golovich * main/astmm.c: Increase the size of filenames stored when astmm is used. If the path length was long they would be truncated and grouped together with whatever matches 2008-01-24 17:47 +0000 [r100206] Joshua Colp * configs/rtp.conf.sample, CHANGES, main/rtp.c: Merge in strictrtp branch. This adds a strictrtp option to rtp.conf which drops packets that do not come from the remote party. (closes issue #8952) Reported by: amorsen 2008-01-24 17:24 +0000 [r100169] Russell Bryant * /, main/asterisk.c: Merged revisions 100164 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r100164 | russell | 2008-01-24 11:22:09 -0600 (Thu, 24 Jan 2008) | 2 lines Update main Asterisk copyright info to 2008 ........ 2008-01-24 16:47 +0000 [r100121-100139] Jason Parker * /, res/res_phoneprov.c, main/acl.c: Merged revisions 100138 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r100138 | qwell | 2008-01-24 10:41:29 -0600 (Thu, 24 Jan 2008) | 6 lines Fix compilation on Solaris. (closes issue #11832) Patches: bug-11832.diff uploaded by snuffy (license 35) ........ * channels/chan_sip.c, main/features.c: Move chan_local dependency into places (only one) that previously depended on res_features, and used local channels 2008-01-24 15:54 +0000 [r100076-100112] Joshua Colp * channels/chan_zap.c, channels/chan_sip.c, channels/chan_iax2.c, channels/chan_mgcp.c: Remove dependency on res_features from some channel drivers. It is now part of the core and no longer exists as a module. * main/channel.c: Some more cosmetic changes. * main/channel.c: Add some spacing. * main/dial.c: Test hopefully over. * main/dial.c: Testing something... 2008-01-24 00:04 +0000 [r100057] Kevin P. Fleming * channels/chan_sip.c: fix flag bit definitions to make code from issue #11049 actually work; along the way, clarify comments and add some dummy flag definitions for other multi-bit flags to hopefully stop this from happening in the future (closes issue #11049) 2008-01-23 23:09 +0000 [r100039] Jason Parker * res/res_features.c (removed), main/Makefile, main/features.c (added), include/asterisk/_private.h, CHANGES, .cleancount, main/asterisk.c, main/loader.c, include/asterisk/features.h: Move code from res_features into (new file) main/features.c 2008-01-23 22:00 +0000 [r100021] Russell Bryant * CREDITS: Add Sergey Tamkovich to CREDITS. Thank you for your contributions! 2008-01-23 21:11 +0000 [r99979-99980] Olle Johansson * /, channels/chan_sip.c: Merged revisions 99978 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r99978 | oej | 2008-01-23 22:07:16 +0100 (Ons, 23 Jan 2008) | 7 lines Second attempt. Don't change invitestate when receiving 18x messages in CANCEL state. (issue #11736) Reported by: MVF Patch by oej. ........ * /, channels/chan_sip.c: Merged revisions 99977 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r99977 | oej | 2008-01-23 21:58:20 +0100 (Ons, 23 Jan 2008) | 9 lines Make sure we don't cancel destruction on calls in CANCEL state, even if we get 183 while waiting for answer on our CANCEL. (issue #11736) Reported by: MVF Patches: bug11736.txt uploaded by oej (license 306) Tested by: MVF ........ 2008-01-23 20:26 +0000 [r99976] Mark Michelson * /, apps/app_externalivr.c: Merged revisions 99975 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r99975 | mmichelson | 2008-01-23 14:25:00 -0600 (Wed, 23 Jan 2008) | 3 lines Fixing a typo. ........ 2008-01-23 17:48 +0000 [r99922-99924] Russell Bryant * /, apps/app_chanspy.c: Merged revisions 99923 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r99923 | russell | 2008-01-23 11:46:55 -0600 (Wed, 23 Jan 2008) | 8 lines ChanSpy issues a beep when it starts at the beginning of a list of channels to potentially spy on. However, if there were no matching channels, it would beep at you over and over, which is pretty annoying. Now, it will only beep once in the case that there are no channels to spy on, but it will still beep again once it reaches the beginning of the channel list again. (closes issue #11738, patched by me) ........ * main/tcptls.c: Fix tcptls build when openssl isn't installed (closes issue #11813) Reported by: tzafrir Patches: asterisk-tcptls.diff.txt uploaded by jamesgolovich (license 176) 2008-01-23 17:27 +0000 [r99920] Kevin P. Fleming * channels/chan_zap.c: since echo canceler parameters in Zaptel are now signed integers, allow them during parsing 2008-01-23 15:23 +0000 [r99860] Tilghman Lesher * channels/chan_h323.c: Progress messages don't work (closes issue #10497) Reported by: pj Patches: h323-announces-r99483.diff uploaded by sergee (license 138) Tested by: pj 2008-01-23 10:18 +0000 [r99839] Olle Johansson * channels/chan_sip.c: - Add a few comments to sip_xmit - Make sure that we are aware of a pending INVITE even if we're using TCP 2008-01-23 05:29 +0000 [r99696-99818] Tilghman Lesher * apps/app_voicemail.c: Coding guidelines fixups * /, apps/app_voicemail.c: Merged revisions 99777 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r99777 | tilghman | 2008-01-22 22:31:51 -0600 (Tue, 22 Jan 2008) | 8 lines When we reset the password via an external command, we should also reset the password stored in the in-memory list, too (otherwise it doesn't really take effect). (closes issue #11809) Reported by: davetroy Patches: fix_externpass.diff uploaded by davetroy (license 384) ........ * /, res/res_odbc.c: Merged revisions 99775 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r99775 | tilghman | 2008-01-22 22:20:15 -0600 (Tue, 22 Jan 2008) | 2 lines Oops, should have checked for a NULL obj, here, too ........ * res/res_config_ldap.c: Coding guidelines cleanup * /, main/acl.c: Merged revisions 99718 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r99718 | tilghman | 2008-01-22 18:56:06 -0600 (Tue, 22 Jan 2008) | 2 lines Just confirmed that all current platforms need this header file ........ * /: Oops * /, build_tools/menuselect-deps.in, configure, include/asterisk/autoconfig.h.in, doc/ldap.txt (added), configure.ac, configs/res_ldap.conf.sample (added), res/res_config_ldap.c (added), CHANGES, makeopts.in, contrib/scripts/asterisk.ldap-schema (added), contrib/scripts/asterisk.ldif (added): Add res_config_ldap for realtime LDAP engine. (closes issue #5768) Reported by: mguesdon Patches: res_config_ldap-v0.7.tar.gz uploaded by mguesdon (license 121) res_ldap.conf.sample uploaded by suretec (license 70) asterisk-v3.1.4.ldif uploaded by suretec (license 70) asterisk-v3.1.4.schema uploaded by suretec (license 70) Tested by: oej, mguesdon, suretec, cthorner 2008-01-22 21:09 +0000 [r99647-99653] Olle Johansson * /, channels/chan_sip.c: Merged revisions 99652 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r99652 | oej | 2008-01-22 21:56:09 +0100 (Tis, 22 Jan 2008) | 4 lines Thanks to Russell's education I realize that BUFSIZ has changed since I learned the C language over 20 years ago... Resetting chan_sip to the size of BUFSIZ that I expected in my old head to avoid too heavy memory allocations on some systems. ........ * doc/tex/channelvariables.tex, CHANGES: Documentation updates for BRIDGEPVTCALLID 2008-01-22 20:42 +0000 [r99646] Tilghman Lesher * /, main/acl.c: Merged revisions 99643 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r99643 | tilghman | 2008-01-22 14:34:55 -0600 (Tue, 22 Jan 2008) | 2 lines Fix the defines for OS X (and Solaris, too) ........ 2008-01-22 20:41 +0000 [r99645] Russell Bryant * main/asterisk.c: Make sure the command is not just present but is also configured to be executed 2008-01-22 20:35 +0000 [r99644] Olle Johansson * main/channel.c, channels/chan_sip.c, include/asterisk/channel.h: Add a generic function to set the bridged call PVT unique id string as a channel variable BRIDGEPVTCALLID This is important for call tracing in log files and CDRs, so that the SIP callID can be traced along servers. The CHANNEL dialplan function won't work here, since the outbound channel is gone when we need the Call-ID. Other channel drivers may now implement the same function :-), but this patch only supports chan_sip.so. Inspired by (issue #11816) Reported by: ctooley Patch by oej 2008-01-22 20:33 +0000 [r99642] Russell Bryant * configs/cli.conf.sample (added), CHANGES, main/asterisk.c: Change the Asterisk CLI startup commands feature to read commands to run from cli.conf after a discussion on the -dev list. 2008-01-22 17:46 +0000 [r99595-99596] Olle Johansson * channels/chan_local.c, /, res/res_features.c, channels/chan_agent.c, apps/app_followme.c: Merged revisions 99594 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r99594 | oej | 2008-01-22 18:41:57 +0100 (Tis, 22 Jan 2008) | 3 lines Add more dependencies on chan_local and add a note to the description of chan_local so that people don't disable it in menuselect just to clean up. ........ * apps/app_dial.c, /: Merged revisions 99592 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r99592 | oej | 2008-01-22 18:31:17 +0100 (Tis, 22 Jan 2008) | 5 lines Add dependency on chan_local to app_dial. Dial still runs without chan_local, but will be missing forwarding functionality. ........ 2008-01-22 17:15 +0000 [r99559] Tilghman Lesher * /, main/acl.c: Merged revisions 99540 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r99540 | tilghman | 2008-01-22 10:54:06 -0600 (Tue, 22 Jan 2008) | 7 lines Ensure that we can get an address even when we don't have a default route. (closes issue #9225) Reported by: junky Patches: 20080122__bug9225.diff.txt uploaded by Corydon76 (license 14) Tested by: oej, loloski, sergee ........ 2008-01-22 16:55 +0000 [r99542] Russell Bryant * channels/chan_sip.c: Point out a bug in some debug counter handling 2008-01-22 15:25 +0000 [r99464-99521] Olle Johansson * channels/chan_sip.c: Add authentication options to the SIP dialstring. Documentation follows separately (issue #11587) Reported by: sobomax Patches: chan_sip.c-trunk.diff uploaded by sobomax (license 359) * configs/sip.conf.sample: Documentation updates * doc/siptls.txt: Small fixes * main/tcptls.c, channels/chan_zap.c, main/abstract_jb.c, include/asterisk/tcptls.h: Doxygen updates 2008-01-21 23:56 +0000 [r99427] Mark Michelson * channels/chan_local.c, /: Merged revisions 99426 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r99426 | mmichelson | 2008-01-21 17:55:26 -0600 (Mon, 21 Jan 2008) | 12 lines Fixing an issue wherein monitoring local channels was not possible. During a channel masquerade, the monitors on the two channels involved are swapped. In 99% of the cases this results in the desired effect. However, if monitoring a local channel, this caused the monitor which was on the local channel to get moved onto a channel which is immediately hung up after the masquerade has completed. By swapping the monitors prior to the masquerade, we avoid the problem by tricking the masquerade into placing the monitor back onto the channel where we want it. During the investigation of the issue, the channel's monitor was the only thing that was swapped in such a manner which did not make sense to have done. All other variable swapping made sense. ........ 2008-01-21 23:25 +0000 [r99424] Jason Parker * channels/chan_zap.c: Fix distinctive ring detection. Reported by: milazzo Patches: drings.diff uploaded by milazzo (license 383) Closes issue #11799 2008-01-21 22:32 +0000 [r99406] Mark Michelson * configs/queues.conf.sample, apps/app_queue.c: Adding the QUEUENAME variable to the variables set using the setqueuevar option in queues.conf. Suggestion comes from Shaun2222 on IRC. 2008-01-21 21:11 +0000 [r99382-99384] Olle Johansson * channels/chan_console.c: Remove compiler warning for uninitialized variable * channels/chan_sip.c: Doxygen updates. The TCP/TLS code was committed without any doxygen obviously. Tss tss. * channels/chan_sip.c: Updating doxygen 2008-01-21 18:15 +0000 [r99350] Tilghman Lesher * include/asterisk/res_odbc.h, /, res/res_odbc.c, configs/res_odbc.conf.sample: Merged revisions 99341 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r99341 | tilghman | 2008-01-21 12:11:07 -0600 (Mon, 21 Jan 2008) | 8 lines Permit the user to specify number of seconds that a connection may remain idle, which fixes a crash on reconnect with the MyODBC driver. (closes issue #11798) Reported by: Corydon76 Patches: 20080119__res_odbc__idlecheck.diff.txt uploaded by Corydon76 (license 14) Tested by: mvanbaak ........ 2008-01-21 16:02 +0000 [r99302] Joshua Colp * /, channels/chan_sip.c: Merged revisions 99301 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r99301 | file | 2008-01-21 12:01:00 -0400 (Mon, 21 Jan 2008) | 4 lines Bump the buffer size for Via headers up to 512. There are some exceptionally large Via headers out there. (closes issue #11783) Reported by: ofirroval ........ 2008-01-21 07:02 +0000 [r99280] Olle Johansson * CREDITS: Update 2008-01-21 03:54 +0000 [r99265] Joshua Colp * channels/chan_sip.c: Change over to using ast_debug so these debug messages don't always show up. 2008-01-20 07:28 +0000 [r99166-99248] Russell Bryant * channels/chan_console.c: Add a "console active" CLI command, which lets you find out which console device is currently active for the Asterisk CLI, or to set it. Also, knock multiple device support off of the to-do list. * configs/console.conf.sample: correct the name of a CLI command for getting available device names * configs/console.conf.sample, channels/chan_console.c: Merge changes from team/russell/console_devices - Add support for multiple devices. All devices are configured in console.conf. - Add "console list devices" CLI command to show configured devices. Also, changed the old "list devices" to be "list available", which queries PortAudio for all audio devices that are available for use. * /, main/slinfactory.c: Merged revisions 99187 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r99187 | russell | 2008-01-19 04:05:27 -0600 (Sat, 19 Jan 2008) | 4 lines Fix a couple of memory leaks with frame handling. Specifically, ast_frame_free() needed to be called on the frame that came from the translator to signed linear. ........ * README: Add Cygwin as an "other" platform that is supported * README: Various README updates 2008-01-18 Russell Bryant * Asterisk 1.6.0-beta1 released. 2008-01-18 22:04 +0000 [r99080-99085] Russell Bryant * CREDITS, include/asterisk/http.h, main/tcptls.c (added), main/manager.c, channels/chan_sip.c, doc/siptls.txt (added), main/Makefile, main/http.c, include/asterisk/tcptls.h (added), configs/sip.conf.sample, CHANGES: Merge changes from team/group/sip-tcptls This set of changes introduces TCP and TLS support for chan_sip. There are various new options in configs/sip.conf.sample that are used to enable these features. Also, there is a document, doc/siptls.txt that describes some things in more detail. This code was implemented by Brett Bryant and James Golovich. It was reviewed by Joshua Colp and myself. A number of other people participated in the testing of this code, but since it was done outside of the bug tracker, I do not have their names. If you were one of them, thanks a lot for the help! (closes issue #4903, but with completely different code that what exists there.) * main/frame.c, /, include/asterisk/translate.h: Merged revisions 99081 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r99081 | russell | 2008-01-18 15:37:21 -0600 (Fri, 18 Jan 2008) | 9 lines Revert adding the packed attribute, as it really doesn't make sense why that would do any good. Fix the real bug, which is to do the check to see if the frame came from a translator at the beginning of ast_frame_free(), instead of at the end. This ensures that it always gets checked, even if none of the parts of the frame are malloc'd, and also ensures that we aren't looking at free'd memory in the case that it is a malloc'd frame. (closes issue #11792, reported by explidous, patched by me) ........ * /, include/asterisk/translate.h: Merged revisions 99079 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r99079 | russell | 2008-01-18 15:22:21 -0600 (Fri, 18 Jan 2008) | 4 lines Since we're relying on the offset between the frame and the beginning of the translator pvt struct, set the packed attribute to make sure we get to the right place. (potential fix for issue #11792) ........ 2008-01-18 16:58 +0000 [r99026] Terry Wilson * res/res_features.c: This should at least temporarily fix a problem where the 't' Dial option is incorrectly passed to the transferee when built-in attended transfers are used. There is still a problem with 'T', but better to fix some problems than no problems while we work on it. (closes issue #7904) Reported by: k-egg Patches: transfer-fix-trunk-r97657.diff uploaded by sergee (license 138) Tested by: sergee, otherwiseguy 2008-01-18 06:58 +0000 [r99015-99018] Tilghman Lesher * funcs/func_odbc.c: Convert func_odbc to use SQLExecDirect for speed (closes issue #10723) Reported by: mnicholson Patches: func-odbc-direct-execute1.diff uploaded by mnicholson (license 96) Tested by: Corydon76, mnicholson, falves11 * res/res_odbc.c: Permit username and password to be NULL (which enables pass-through from the layer above). Reported by: lurcher Patch by: tilghman (Closes issue #11739) * funcs/func_cut.c: Reset default CUT delimiter back to '-' 2008-01-17 23:28 +0000 [r99006-99011] Russell Bryant * channels/chan_console.c: Make the output of "console list devices" a bit prettier. * channels/chan_console.c: List which devices are inputs and outputs in "console list devices" * main/channel.c: Add AST_FORMAT_SLINEAR16 to the list for ast_best_codec() * main/frame.c, /, channels/chan_iax2.c, include/asterisk/frame.h: Merged revisions 99004 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r99004 | russell | 2008-01-17 16:37:22 -0600 (Thu, 17 Jan 2008) | 10 lines Have IAX2 optimize the codec translation path just like chan_sip does it. If the caller's codec is in our codec list, move it to the top to avoid transcoding. (closes issue #10500) Reported by: stevedavies Patches: iax-prefer-current-codec.patch uploaded by stevedavies (license 184) iax-prefer-current-codec.1.4.patch uploaded by stevedavies (license 184) Tested by: stevedavies, pj, sheldonh ........ 2008-01-17 22:22 +0000 [r99002] Mark Michelson * apps/app_voicemail.c: Fixing trunk IMAP build (closes issue #11788) Reported by: DEA Patches: vm-imap-build-fix.txt uploaded by DEA (license 3) 2008-01-17 20:51 +0000 [r98998] Jason Parker * Makefile, build_tools/cflags.xml, channels/chan_zap.c, main/dsp.c, configs/zapata.conf.sample: Add several busy detection related defines to menuselect. Allow better busy detect debugging (with BUSYDETECT_DEBUG). Remove very old BUSYDETECT and BUSY_DETECT_MARTIN defines. (closes issue #11107) Patches: busydetect_enhancement.patch uploaded by agx (license 298) busydetect-r94975.diff uploaded by sergee (license 138) Additional changes/cleanup by me. 2008-01-17 16:33 +0000 [r98993-98994] Mark Michelson * apps/app_queue.c: state_interface could be NULL, so use the never-NULL cur->state_interface for this check * apps/app_queue.c: Get the device state of the state interface instead of the interface when creating a new queue member. Thanks to Atis Lezdins for bringing this up on the Asterisk-Dev mailing list. 2008-01-17 16:21 +0000 [r98992] Jason Parker * /, configs/zapata.conf.sample: Merged revisions 98991 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 (Closes issue #11784) ........ r98991 | qwell | 2008-01-17 10:19:46 -0600 (Thu, 17 Jan 2008) | 4 lines Add a clarification about the immediate= option of zapata.conf Issue 11784, patch by klaus3000. ........ 2008-01-17 16:17 +0000 [r98989-98990] Kevin P. Fleming * channels/chan_zap.c, configs/zapata.conf.sample: major reliability and performance improvement in VWMI monitoring for FXO ports (code by markster, me and dbailey) * res/res_config_curl.c: resolve (valid) compiler warning about variable that could be used before being initialized 2008-01-17 03:09 +0000 [r98988] Terry Wilson * res/res_phoneprov.c, doc/tex/phoneprov.tex, configs/phoneprov.conf.sample: Update res_phoneprov to default to setting the SERVER variable to the IP the HTTP request for the config came in on and the SERVER_PORT to the bindport setting in sip.conf. I've left in the ability to override these options, because I can't always guess how someone might decide to do something weird with what is available to them--although needing to is pretty unlikely. Documentation was updated to reflect preference for not setting serveraddr, serveriface, or serverport. Tested on Linux and OS X. 2008-01-17 00:13 +0000 [r98987] Tilghman Lesher * cdr/cdr_adaptive_odbc.c: Change the way the new filter feature works, by allowing it to be a column NOT logged into the database. This will allow more granularity of a decision evaluated in the dialplan, then takes effect when posting the CDR. 2008-01-17 00:05 +0000 [r98986] Russell Bryant * CHANGES, main/asterisk.c: Add support for an easy way to automatically execute some Asterisk CLI commands immediately at startup. Any commands in the startup_commands file in the Asterisk config diretory will get executed. (closes issue #11781) Reported by: jamesgolovich Patches: asterisk-startupcmds.diff.txt uploaded by jamesgolovich (license 176) -- With some changes by me. 2008-01-16 23:08 +0000 [r98985] Jason Parker * configure, include/asterisk/autoconfig.h.in, configure.ac, acinclude.m4: Change AST_EXT_TOOL_CHECK to attempt to build against _LIB, per recommendations from Russell. 2008-01-16 22:36 +0000 [r98984] Tilghman Lesher * CHANGES: Info about res_config_curl 2008-01-16 22:20 +0000 [r98981] Tilghman Lesher * res/res_config_curl.c (added), main/utils.c: New module res_config_curl (closes issue #11747) Reported by: Corydon76 Patches: res_config_curl.c uploaded by Corydon76 (license 14) 20080116__bug11747.diff.txt uploaded by Corydon76 (license 14) Tested by: jmls 2008-01-16 21:53 +0000 [r98978] Russell Bryant * CREDITS, channels/chan_sip.c, configs/sip.conf.sample: Merge the changes from issue #10665 from the team/group/sip_session_timers branch. This set of changes introduces SIP session timers support (RFC 4028). In short, this prevents stuck SIP sessions that were not properly torn down due to network or endpoint failures during an established SIP session. To quote some of the documentation supplied with the patch: "The SIP Session-Timers is an extension of the SIP protocol that allows end-points and proxies to refresh a session periodically. The sessions are kept alive by sending a RE-INVITE or UPDATE request at a negotiated interval. If a session refresh fails then all the entities that support Session- Timers clear their internal session state. In addition, UAs generate a BYE request in order to clear the state in the proxies and the remote UA (this is done for the benefit of SIP entities in the path that do not support Session-Timers)." (closes issue #10665) Reported by: rjain Patches: chan_sip.c.1.diff uploaded by rjain (license 226) chan_sip.c.diff uploaded by rjain (license 226) sip.conf.sample.diff uploaded by rjain (license 226) proc_422_rsp_comment.diff uploaded by rjain (license 226) chan_sip.c.cache.diff uploaded by rjain (license 226) chan_sip.memalloc uploaded by rjain (license 226) chan_sip.memalloc.bugfix uploaded by rjain (license 226) Patches tracked in team/group/sip_session_timers, with some additional fixes by russell and oej. Tested by: jtodd, rjain, loloski 2008-01-16 19:41 +0000 [r98968-98971] Jason Parker * configure, include/asterisk/autoconfig.h.in, configure.ac: Partially revert r93898, because it broke the way netsnmp was being detected. rizzo, do you want to discuss so we can rethink this, or do you have another way? * CHANGES: Add note about new update.log to CHANGES, by request of jmls and further prodding by jsmith. * Makefile, /: Add logging for 'make update' command (also fixes updates in some places). Issue #11766, initial patch by jmls. 2008-01-16 17:51 +0000 [r98967] Joshua Colp * /, channels/chan_iax2.c: Merged revisions 98966 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r98966 | file | 2008-01-16 13:50:10 -0400 (Wed, 16 Jan 2008) | 6 lines Add missing NULLs at end of two ast_load_realtimes. (closes issue #11769) Reported by: tequ Patches: chaniax.patch uploaded by dimas (license 88) ........ 2008-01-16 17:21 +0000 [r98965] Mark Michelson * channels/chan_local.c, /: Merged revisions 98964 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r98964 | mmichelson | 2008-01-16 11:20:11 -0600 (Wed, 16 Jan 2008) | 10 lines Fix a deadlock in chan_local in local_hangup. There was contention because the local_pvt was held and it was attempting to lock a channel, which is the incorrect locking order. (closes issue #11730) Reported by: UDI-Doug Patches: 11730.patch uploaded by putnopvut (license 60) Tested by: UDI-Doug ........ 2008-01-16 16:06 +0000 [r98962] Terry Wilson * res/res_phoneprov.c: Make users list static 2008-01-16 15:09 +0000 [r98954-98961] Joshua Colp * main/dial.c, /: Merged revisions 98960 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r98960 | file | 2008-01-16 11:08:24 -0400 (Wed, 16 Jan 2008) | 6 lines Introduce a lock into the dialing API that protects it when destroying the structure. (closes issue #11687) Reported by: callguy Patches: 11687.diff uploaded by file (license 11) ........ * /, main/rtp.c: Merged revisions 98958 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r98958 | file | 2008-01-16 11:03:14 -0400 (Wed, 16 Jan 2008) | 4 lines Add two more SDP names for ulaw and alaw. (closes issue #11777) Reported by: tootai ........ * /, channels/chan_sip.c: Merged revisions 98955 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r98955 | file | 2008-01-15 23:07:24 -0400 (Tue, 15 Jan 2008) | 6 lines Don't drop the old record route information when dealing with packets related to a reinvite. (closes issue #11545) Reported by: kebl0155 Patches: reinvite-patch.txt uploaded by kebl0155 (license 356) ........ * channels/chan_sip.c: Remove DNS lookup from sip_devicestate. This seems to come from way back when and I can't think of a reason for it being here, plus it could cause needless DNS lookups. (closes issue #10983) Reported by: jtodd 2008-01-16 01:35 +0000 [r98953] Steve Murphy * main/ast_expr2.h, main/ast_expr2.y, main/ast_expr2.c: Terry found this problem with running the expr2 parser on OSX. Make the #defines come out the same between the parser & lexer. 2008-01-16 01:17 +0000 [r98952] Joshua Colp * /, build_tools/menuselect-deps.in, configure, include/asterisk/autoconfig.h.in, codecs/codec_speex.c, configure.ac, makeopts.in: Merged revisions 98951 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r98951 | file | 2008-01-15 21:13:27 -0400 (Tue, 15 Jan 2008) | 4 lines Add autoconf logic for speexdsp. Later versions use a separate library for some things so we need to use it if present in codec_speex. (closes issue #11693) Reported by: yzg ........ 2008-01-15 23:53 +0000 [r98948] Russell Bryant * /, channels/chan_sip.c: Merged revisions 98946 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r98946 | russell | 2008-01-15 17:50:10 -0600 (Tue, 15 Jan 2008) | 11 lines Change a buffer in check_auth() to be a thread local dynamically allocated buffer, instead of a massive buffer on the stack. This fixes a crash reported by Qwell due to running out of stack space when building with LOW_MEMORY defined. On a very related note, the usage of BUFSIZ in various places in chan_sip is arbitrary and careless. BUFSIZ is a system specific define. On my machine, it is 8192, but by definition (according to google) could be as small as 256. So, this buffer in check_auth was 16 kB. We don't even support SIP messages larger than 4 kB! Further usage of this define should be avoided, unless it is used in the proper context. ........ 2008-01-15 23:52 +0000 [r98947] Tilghman Lesher * cdr/cdr_adaptive_odbc.c, configs/cdr_adaptive_odbc.conf.sample: Add the "filter" keyword 2008-01-15 23:35 +0000 [r98944-98945] Russell Bryant * main/translate.c, include/asterisk/translate.h: Clean up something I did for ABI compatability in 1.4 * main/frame.c, /, main/translate.c, main/abstract_jb.c, channels/chan_iax2.c, codecs/codec_zap.c, include/asterisk/frame.h, main/rtp.c, include/asterisk/translate.h: Merged revisions 98943 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r98943 | russell | 2008-01-15 17:26:52 -0600 (Tue, 15 Jan 2008) | 25 lines Commit a fix for some memory access errors pointed out by the valgrind2.txt output on issue #11698. The issue here is that it is possible for an instance of a translator to get destroyed while the frame allocated as a part of the translator is still being processed. Specifically, this is possible anywhere between a call to ast_read() and ast_frame_free(), which is _a lot_ of places in the code. The reason this happens is that the channel might get masqueraded during this time. During a masquerade, existing translation paths get destroyed. So, this patch fixes the issue in an API and ABI compatible way. (This one is for you, paravoid!) It changes an int in ast_frame to be used as flag bits. The 1 bit is still used to indicate that the frame contains timing information. Also, a second flag has been added to indicate that the frame came from a translator. When a frame with this flag gets released and has this flag, a function is called in translate.c to let it know that this frame is doing being processed. At this point, the flag gets cleared. Also, if the translator was requested to be destroyed while its internal frame still had this flag set, its destruction has been deffered until it finds out that the frame is no longer being processed. Admittedly, this feels like a hack. But, it does fix the issue, and I was not able to think of a better solution ... ........ 2008-01-15 20:10 +0000 [r98895-98935] Joshua Colp * /, channels/chan_sip.c: Merged revisions 98934 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r98934 | file | 2008-01-15 16:08:43 -0400 (Tue, 15 Jan 2008) | 4 lines Based on the boundary found move over the correct amount. (closes issue #11750) Reported by: tasker ........ * /, channels/chan_sip.c: Merged revisions 98894 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r98894 | file | 2008-01-14 18:41:55 -0400 (Mon, 14 Jan 2008) | 4 lines Accept "; boundary=" not just ";boundary=" in the multipart mixed content type. (closes issue #11750) Reported by: tasker ........ 2008-01-14 22:19 +0000 [r98889] Jason Parker * configs/voicemail.conf.sample, CHANGES, apps/app_voicemail.c: Add backupdeleted option to app_voicemail (closes issue #10740) Reported by: ruffle Patches: app_voicemail.diff uploaded by ruffle (license 201) 10740-voicemail.diff uploaded by qwell (license 4) 20080113_bug10740.diff.txt uploaded by mvanbaak (license 7) Tested by: blitzrage, mvanbaak, qwell 2008-01-14 22:11 +0000 [r98850-98888] Mark Michelson * apps/app_directory.c: Big improvement for app_directory. This patch breaks the do_directory function up so that it is more easily parsed by the human brain. It also fixes some errors. I'll quote dimas from the original bug description: "app_directory contained some duplicate code even before addition of 'm' option. Addition of that option doubled amount of that code. Worst of all, there are minor differences between these code block and bugs caused by these differences. 1. There is a memory leak. In the 'menu' mode, result of the convert(pos) function is not freed while it should be. 2. In the 'menu' mode check for OPT_LISTBYFIRSTNAME flag ('f' option) is not negated as result, application works in the mode opposite to what user expect (checking last name when user wants the first nd vice versa). 3. select_item function plays message for user using res = func1() || func2() || func3()... construct. This construct loses the actual value returned by ast_waitstream() for example so at the end, res does not contain digit user dialed while listening to the message. 4. (also in 1.4) application announces entries from voicemail.conf/realtime separately from entries from users.conf. I see no reason why doing so instead of building combined list. 5. Alot of duplicated code as already mentioned." This was tested by dimas and I (I tested under valgrind). A word of caution: any bug fixes that happen in app_directory in 1.4 will almost certainly not merge cleanly into trunk as a result of this, but it is well worth it. Huge thanks to dimas for this wonderful submission. (closes issue #11744) Reported by: dimas Patches: dir3.patch uploaded by dimas (license 88) Tested by: putnopvut, dimas 2008-01-14 20:01 +0000 [r98830] Joshua Colp * main/manager.c: Make sure the user's manager secret exists, even if it is blank. (closes issue #11749) Reported by: srt 2008-01-14 18:42 +0000 [r98811] Terry Wilson * CHANGES: Add description of TOUPPER and TOLOWER dialplan functions to CHANGES. 2008-01-14 17:40 +0000 [r98776] Jason Parker * channels/chan_skinny.c: Add proper call forwarding (all and busy) support for chan_skinny. Note: NoAnswer support is currently not implemented, as it would take a significant amount of work to figure out how to do correctly. Closes issue #11310, patches, testing, and support by DEA, mvanbaak, and myself. 2008-01-14 17:39 +0000 [r98775] Russell Bryant * /, main/translate.c: Merged revisions 98774 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r98774 | russell | 2008-01-14 11:38:38 -0600 (Mon, 14 Jan 2008) | 3 lines Revert a change that introduces an unacceptable performance hit and is causing memory leaks ... (from rev 97973) ........ 2008-01-14 17:18 +0000 [r98773] Jason Parker * channels/chan_skinny.c: Fix for potential crash with vmexten 2008-01-14 16:36 +0000 [r98735-98738] Mark Michelson * apps/app_queue.c: Merged revisions 98737 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r98737 | mmichelson | 2008-01-14 10:35:12 -0600 (Mon, 14 Jan 2008) | 3 lines Fixing another compilation error. I'm a bit off today :( ........ * /, apps/app_queue.c: Merged revisions 98733 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r98733 | mmichelson | 2008-01-14 10:21:28 -0600 (Mon, 14 Jan 2008) | 8 lines Adding explicit defaults for missing options to init_queue. This is necessary because if a user either removes or comments one of these options and reloads their queues, the option will not reset to its default, instead maintaining the value from prior to the reload. Thanks to John Bigelow for pointing this error out to me. ........ 2008-01-14 15:07 +0000 [r98695-98714] Joshua Colp * main/pbx.c: Print out a warning when spaces are used in the variable name in Set and MSet. It is extremely hard to debug this issue so this should make it easier. (closes issue #11759) Reported by: caio1982 Patches: setvar_space_warning1.diff uploaded by caio1982 (license 22) * apps/app_meetme.c, doc/tex/qos.tex, doc/tex/realtime.tex: Update documentation. (closes issue #11763) Reported by: IgorG Patches: docupd.v1.diff uploaded by IgorG (license 20) 2008-01-14 04:53 +0000 [r98558-98676] Russell Bryant * apps/app_jack.c: Add another small option for the JACK app and JACK_HOOK function. The 'n' option tells JACK not to start jackd automatically if it is not already running. Otherwise, the default is that jackd will get started for you if it isn't running already. * CHANGES: - Break up the Misc. section a bit with a new section for Misc. New Modules - Change spacing a bit in some places for consistent indentation * CHANGES, apps/app_jack.c (added): Bring in the code from team/russell/jack/. Add a new module, app_jack, which provides interfaces to JACK, the Jack Audio Connection Kit (http://www.jackaudio.org/). Two interfaces are provided; there is a JACK() application, and a JACK_HOOK() function. Both interfaces create an input and output JACK port. The application makes these ports the endpoint of the call. The audio coming from the channel goes out the output port and whatever comes back in on the input port is what gets sent to the channel. The JACK_HOOK() function turns on a JACK audiohook on the channel. This lets you run the audio coming from a channel through JACK, and whatever comes back in is what gets forwarded on as the channel's audio. This is very useful for building custom vocoders or doing recording or analysis of the channel's audio in another application. In case anyone is curious, the platform that inspired me to write this is PureData (http://puredata.info/). I wrote these JACK interfaces so that I could use Pd to do interesting things with the audio of phone calls ... * build_tools/menuselect-deps.in, configure, include/asterisk/autoconfig.h.in, configure.ac, makeopts.in: Add configure script check for JACK. * build_tools/menuselect-deps.in, configure, include/asterisk/autoconfig.h.in, configure.ac, makeopts.in: Remove KDE configure script check that isn't used * main/audiohook.c: Remove a duplicate lock of the audiohook lock when destroying manipulate audiohooks. This causes an error when we attempt to destroy the lock later when freeing the audiohook. * main/pbx.c, CHANGES: Add a new CLI command, "core set chanvar", which allows you to set a channel variable (or function) on an active channel from the CLI. 2008-01-12 18:12 +0000 [r98536] Tilghman Lesher * main/manager.c: Conversion to load manager.conf into memory did not convert the password functions correctly. (Closes issue #11749) 2008-01-12 05:13 +0000 [r98514] Pari Nannapaneni * /, main/http.c: merging a comment added in 1.4 2008-01-12 00:20 +0000 [r98488] Kevin P. Fleming * channels/chan_zap.c, CHANGES: Add 'zap set dnd' CLI command, and ensure that the AMI DNDState event always gets generated. (closes issue #11212) Reported by: tzafrir Patches: zap_dnd.diff uploaded by tzafrir (modified by me) (license 46) 2008-01-12 00:17 +0000 [r98487] Tilghman Lesher * /, res/res_odbc.c: Merged revisions 98467 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r98467 | tilghman | 2008-01-11 18:05:08 -0600 (Fri, 11 Jan 2008) | 4 lines Add a connection timeout attribute, as that was what was intended with the login timeout, but ODBC divides it up into 2 different timeouts. (Closes issue #11745) ........ 2008-01-11 23:57 +0000 [r98454] Russell Bryant * configure, doc/tex/Makefile, configure.ac, makeopts.in: Add some extra checking to help out with a potential error when trying to run "make asterisk.pdf" when not all of the right packages are installed. (closes issue #10763) Reported by: Corydon76 Patches: 20070919__bug10763.diff.txt uploaded by Corydon76 (license 14) Tested by: Corydon76 2008-01-11 23:10 +0000 [r98436] Kevin P. Fleming * channels/chan_zap.c, CHANGES, configs/zapata.conf.sample: Add 'auto' signalling mode for Zaptel channels. (closes issue #11690) Reported by: tzafrir Patches: signaling_to_signalling.diff uploaded by tzafrir (license 46) signalling_cleanup.diff uploaded by tzafrir (license 46) zap_auto_default.diff uploaded by tzafrir (license 46) zap_no_default_sig.diff uploaded by tzafrir (license 46) zap_signal_auto.diff uploaded by tzafrir (license 46) 2008-01-11 23:09 +0000 [r98424-98435] Joshua Colp * main/event.c: Goodbye again drumkilla. * main/event.c: drumkilla ftw. * main/audiohook.c: I am no longer Rockin' * main/audiohook.c: Testing something... 2008-01-11 22:52 +0000 [r98400] Russell Bryant * /, pbx/pbx_dundi.c: Merged revisions 98390 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r98390 | russell | 2008-01-11 16:46:21 -0600 (Fri, 11 Jan 2008) | 9 lines Fix up setting the EID on BSD based systems. (closes issue #11646) Reported by: caio1982 Patches: dundi_osx_eid6.diff.txt uploaded by caio1982 (license 22) dundi_osx_eid6-1.4.diff uploaded by caio1982 (license 22) Tested by: caio1982, mvanbaak ........ 2008-01-11 19:53 +0000 [r98318-98334] Joshua Colp * /, main/rtp.c: Merged revisions 98325 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r98325 | file | 2008-01-11 15:51:10 -0400 (Fri, 11 Jan 2008) | 6 lines If the incoming RTP stream changes codec force the bridge to break if the other side does not support it. (closes issue #11729) Reported by: tsearle Patches: new_codec_patch_udiff.patch uploaded by tsearle (license 373) ........ * /, res/res_agi.c: Merged revisions 98317 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r98317 | file | 2008-01-11 15:28:30 -0400 (Fri, 11 Jan 2008) | 6 lines If the channel is hungup during RECORD FILE send a result code of -1 to be uniform with everything else. (closes issue #11743) Reported by: davevg Patches: res_agi.diff uploaded by davevg (license 209) ........ 2008-01-11 19:12 +0000 [r98316] Mark Michelson * main/channel.c, /: Merged revisions 98315 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r98315 | mmichelson | 2008-01-11 13:10:57 -0600 (Fri, 11 Jan 2008) | 5 lines Properly report the hangup cause as no answer when someone does not answer (closes issue #10574, reported by boch, patched by moy) ........ 2008-01-11 19:05 +0000 [r98270-98308] Russell Bryant * codecs/codec_resample.c: Kevin noted that the thing that I _actually_ changed here was that I converted a value from a double, to a float, back to a double. Sure enough, when I changed my interim variable back to a double, it still blows up. Switching all of these to a float fixes the problem. This seems like a compiler bug where a double passed as an argument isn't getting properly aligned, so I'll have to see if I can replicate it with a small test program. (related to issue #11725) * codecs/codec_resample.c: Fix a bus error that happened when asterisk was built with optimizations on with platforms that explode on unaligned access. I'm not exactly sure why this fixes it, but it fixed it on the machine I was testing on. If it makes sense to you, feel free to enlighten me. :) (closes issue #11725, patched by me) 2008-01-11 18:35 +0000 [r98268-98269] Tilghman Lesher * cdr/cdr_adaptive_odbc.c: Port Nick Gorham's timestamp patch to adaptive_odbc, too * cdr/cdr_odbc.c: Commit Nick Gorham's suggestion for timestamp fix 2008-01-11 17:27 +0000 [r98220] Joshua Colp * /, apps/app_followme.c: Merged revisions 98219 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r98219 | file | 2008-01-11 13:22:53 -0400 (Fri, 11 Jan 2008) | 4 lines Ensure the return value of ast_bridge_call is passed back up as the application return value. This is needed for transfers to function so the PBX core knows to continue execution. (closes issue #10327) Reported by: kkiely ........ 2008-01-11 17:17 +0000 [r98218] Russell Bryant * codecs/codec_g722.c: At one point during working on this module, I had the lin/lin16 versions of the framein callbacks different. However, they are now the same again, so remove the duplicate code and use the same functions for the lin/lin16 versions. 2008-01-11 16:08 +0000 [r98152-98193] Tilghman Lesher * /, channels/chan_sip.c: Merged revisions 98164 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r98164 | tilghman | 2008-01-11 09:52:31 -0600 (Fri, 11 Jan 2008) | 2 lines Back out changes from revision 97077, since it wasn't perfect ........ * doc/manager_1_1.txt: Documentation updates 2008-01-11 12:51 +0000 [r98124] Kevin P. Fleming * channels/chan_sip.c: Ascom phones send Flash events as SIP INFO using '!' as the 'digit' 2008-01-11 03:40 +0000 [r98081-98083] Russell Bryant * codecs/codec_g722.c, main/frame.c: - Fix the last set of places where incorrect assumptions were made about the sample length with g722. It is _2_ samples per byte, not 1. This was all over the place, and I believed it, and it is what caused me to take so long to figure out what was broken. - Update copyright information on codec_g722. 2008-01-11 00:54 +0000 [r98047] Mark Michelson * main/translate.c: Fix "core show translation" to not output information for "unknown" codecs. This fix was made in favor of the proposed patch since it doesn't involve changing a core codec define. (closes issue #11722, reported and initially patched by caio1982, final patch by me) 2008-01-11 00:38 +0000 [r98024-98027] Russell Bryant * channels/chan_sip.c, configs/sip.conf.sample, CHANGES: Add a new global and per-peer option to chan_sip, qualifyfreq, which allows you to set the qualify frequency. (closes issue #11597) Reported by: wilder Patches: qualifyfreq5.patch uploaded by wilder (license 362) -- with some mods by me * main/translate.c: Simplify this code with a suggestion from Luigi on the asterisk-dev list. Instead of using is16kHz(), implement a format_rate() function. 2008-01-10 23:40 +0000 [r97978] Tilghman Lesher * /, channels/chan_sip.c, main/translate.c: Merged revisions 97973 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r97973 | tilghman | 2008-01-10 17:08:36 -0600 (Thu, 10 Jan 2008) | 6 lines 1) When we get a translated frame out, clone it, because if the translator pvt is freed before we use the frame, bad things happen. 2) Getting a failure from ast_sched_delete means that the schedule ID is currently running. Don't just ignore it. (Closes issue #11698) ........ 2008-01-10 23:33 +0000 [r97974-97977] Russell Bryant * /, main/translate.c: Merged revisions 97976 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r97976 | russell | 2008-01-10 17:30:40 -0600 (Thu, 10 Jan 2008) | 3 lines Fix various timing calculations that made assumptions that the audio being processed was at a sample rate of 8 kHz. ........ * codecs/codec_g722.c: Fix various issues in codec_g722. - The most common fix being made here is to fix all of the places where the number of output samples and output bytes gets updated in the translator state structure. - Fix a number of other places where the number of samples provided as an initialization value to a struct was incorrect. * codecs/codec_resample.c: Fix the buffer_samples value. For signed linear, the number of samples needed to fill the buffer is half the buffer size. 2008-01-10 21:58 +0000 [r97933] Mark Michelson * /, apps/app_voicemail.c: Merged revisions 97925 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r97925 | mmichelson | 2008-01-10 15:57:06 -0600 (Thu, 10 Jan 2008) | 6 lines Let us leave a voicemail for ourself if we have logged into VoiceMailMain and chosen to leave a message. (closes issue #11735, reported and patched by jamessan) ........ 2008-01-10 21:46 +0000 [r97850-97890] Steve Murphy * /, res/ael/ael_lex.c, res/Makefile, res/ael/ael.flex: Merged revisions 97889 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r97889 | murf | 2008-01-10 14:37:10 -0700 (Thu, 10 Jan 2008) | 1 line Applied the same fixes for ael.flex as was done in 97849 for ast_expr2.fl; overrode the normally generate yyfree func with our own version that checks the pointer for non-null before passing to free(). Also takes care of a little problem with 2.5.33 and the use of the __STDC_VERSION__ macro. ........ * /, main/Makefile, main/ast_expr2f.c, main/ast_expr2.fl: Merged revisions 97849 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r97849 | murf | 2008-01-10 13:21:27 -0700 (Thu, 10 Jan 2008) | 1 line This is a fix for 2 things: a problem Terry was having in OSX with null pointers, which was my fault, as I probably forgot to run the sed script last time I made mods. So, I moved the fix into the flex input itself. Then, I found when I used flex 2.5.33, that it was using __STDC_VERSION__, and that's not real good; so I added back in a DIFFERENT sed script to fix that little mess. Tested everything, a couple different ways. Hope I did no harm, at the least. ........ 2008-01-10 20:13 +0000 [r97848] Jason Parker * /, include/asterisk/frame.h: Merged revisions 97847 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r97847 | qwell | 2008-01-10 14:12:37 -0600 (Thu, 10 Jan 2008) | 1 line Fix a comment that is no longer true. ........ 2008-01-10 20:05 +0000 [r97846] Mark Michelson * apps/app_voicemail.c: Use the appropriate line ending for the X-Asterisk-VM-Message-Type header. (closes issue #11734, reported and patched by jaroth) 2008-01-10 19:07 +0000 [r97825-97826] Terry Wilson * main/ast_expr2f.c: heh, remove patch to generated file. * main/ast_expr2f.c, main/cli.c: Check pointers before freeing (was getting WARNINGS under MALLOC_DEBUG) 2008-01-10 17:38 +0000 [r97805] Tilghman Lesher * cdr/cdr_odbc.c: Fix problem with timestr going out of scope (Closes issue #11726, closes issue #11731) 2008-01-10 17:30 +0000 [r97745-97804] Russell Bryant * formats/format_sln16.c: minor formatting changes * main/translate.c: spaces to tabs * configure, configure.ac: Use AST_EXT_TOOL_CHECK() for the GTK check again. I changed this to an inline implementation to fix a small bug, but after a discussion with rizzo, I went to change it back. Also, it turns out that the implementation of the macro already supported what was needed to fix the problem. * pbx/pbx_kdeconsole.h (removed), /, configs/modules.conf.sample, pbx/kdeconsole_main.cc (removed): Merged revisions 97753 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r97753 | russell | 2008-01-10 10:19:47 -0600 (Thu, 10 Jan 2008) | 2 lines Remove other remnants of pbx_kdeconsole ........ * /, build_tools/menuselect-deps.in, configure, include/asterisk/autoconfig.h.in, configure.ac, makeopts.in, pbx/pbx_kdeconsole.cc (removed): Merged revisions 97734 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r97734 | russell | 2008-01-10 10:10:09 -0600 (Thu, 10 Jan 2008) | 4 lines Remove pbx_kdeconsole from the tree. It hasn't worked in ages, and nobody has complained. (closes issue #11706, reported by caio1982) ........ 2008-01-10 15:12 +0000 [r97698] Joshua Colp * funcs/func_groupcount.c, /: Merged revisions 97697 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r97697 | file | 2008-01-10 11:07:12 -0400 (Thu, 10 Jan 2008) | 6 lines Don't try to copy the category from the group if no category exists. (closes issue #11724) Reported by: IgorG Patches: group_count.v1.patch uploaded by IgorG (license 20) ........ 2008-01-10 00:54 +0000 [r97657] Russell Bryant * include/asterisk.h: These prototypes are not supposed to be in asterisk.h. They are already in version.h. 2008-01-10 00:50 +0000 [r97656] Steve Murphy * include/asterisk.h, channels/console_video.c, utils/astman.c, channels/console_board.c, channels/vgrabbers.c: The fixes in this commit are mainly to allow compiling of trunk with --enable-dev-mode, mutex profiling, lock debugging, etc. Mainly, the version.c needs to be in the OBJS line; asterisk.h was chosen to have the prototypes for ast_get_version, ast_get_version_num; and the ASTERISK_FILE_VERSION macro needs to be used after including asterisk.h in a few files. I hope I did the right thing. If not, let me know. 2008-01-10 00:39 +0000 [r97655] Tilghman Lesher * main/manager.c: oops, missed the case of a 0 permission (which should mean everybody is allowed, not nobody) 2008-01-10 00:22 +0000 [r97653] Terry Wilson * res/res_phoneprov.c: Attempt at making lookup_iface work under FreeBSD. Not yet tested, but it compiles under OS X. And still works under linux. 2008-01-10 00:17 +0000 [r97652] Russell Bryant * codecs/Makefile: Fix this so it doesn't force codec_g722 to get relinked every time 2008-01-10 00:12 +0000 [r97651] Tilghman Lesher * main/pbx.c, main/manager.c, channels/chan_sip.c, res/res_features.c, pbx/pbx_realtime.c, configs/manager.conf.sample, CHANGES, channels/chan_iax2.c, include/asterisk/manager.h, apps/app_stack.c, main/db.c, apps/app_voicemail.c: Several manager changes: 1) Add the Dialplan class, for NewExten and VarSet events, which should cut down on the volume of traffic in the Call class. 2) Permit some commands to be run from multiple classes, such as allowing DBGet to be run from either the System or the Reporting class. 3) Heavily document each class in the sample config, as there were several that made no sense to be in the write= line, and two that made no sense to be in the read= line (since they controlled no permissions there). (Closes issue #10386) 2008-01-10 00:11 +0000 [r97641-97650] Russell Bryant * codecs/Makefile: Ensure that libg722.a gets rebuilt if one of the files changes * /, pbx/pbx_gtkconsole.c: Merged revisions 97645 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r97645 | russell | 2008-01-09 17:01:48 -0600 (Wed, 09 Jan 2008) | 2 lines Strip terminal sequences from the verbose messages ........ * configure: re-gen configure * configure.ac: re-add check for gtk1, which is used for pbx_gtkconsole (related to issue #11706) * /, pbx/pbx_gtkconsole.c: Merged revisions 97640 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r97640 | russell | 2008-01-09 16:26:33 -0600 (Wed, 09 Jan 2008) | 3 lines Make pbx_gtkconsole build ... but doesn't actually load on my system still (related to issue #11706) ........ 2008-01-09 21:37 +0000 [r97634] Terry Wilson * phoneprov/000000000000.cfg, phoneprov/000000000000-directory.xml, phoneprov/polycom.xml, res/res_phoneprov.c (added), funcs/func_strings.c, phoneprov/000000000000-phone.cfg, configs/modules.conf.sample, main/acl.c, include/asterisk/localtime.h, CHANGES, configs/phoneprov.conf.sample (added), Makefile, phoneprov (added), doc/tex/phoneprov.tex (added), main/stdtime/localtime.c, doc/tex/asterisk.tex: Added a new module, res_phoneprov, which allows auto-provisioning of phones based on configuration templates that use Asterisk dialplan function and variable substitution. It should be possible to create phone profiles and templates that work for the majority of phones provisioned over http. It is currently only intended to provision a single user account per phone. An example profile and set of templates for Polycom phones is provided. NOTE: Polycom firmware is not included, but should be placed in AST_DATA_DIR/phoneprov/configs to match up with the included templates. 2008-01-09 20:30 +0000 [r97620-97623] Jason Parker * /, main/cli.c: Merged revisions 97622 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 (closes issue #11718) ........ r97622 | qwell | 2008-01-09 14:28:43 -0600 (Wed, 09 Jan 2008) | 5 lines Correctly display a message if a command could not be found. Also fix a comment which may have led to this happening. Issue 11718, reported by kshumard. ........ * /, main/cli.c: Merged revisions 97618 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r97618 | qwell | 2008-01-09 14:05:45 -0600 (Wed, 09 Jan 2008) | 1 line Fix some locking and return value funkiness. We really shouldn't be unlocking this lock inside of a function, unless we locked it there too. ........ 2008-01-09 18:53 +0000 [r97577] Mark Michelson * /, apps/app_queue.c: Merged revisions 97575 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r97575 | mmichelson | 2008-01-09 12:48:15 -0600 (Wed, 09 Jan 2008) | 3 lines Part 2 of app_queue doxygen improvements. Some smaller functions this time ........ 2008-01-09 18:12 +0000 [r97532-97533] Luigi Rizzo * channels/console_gui.c: remove a wrong 'const' * images/kpad2.jpg: add annotations for the two message windows we use. 2008-01-09 18:04 +0000 [r97531] Russell Bryant * /, res/res_features.c: Merged revisions 97529 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r97529 | russell | 2008-01-09 12:02:08 -0600 (Wed, 09 Jan 2008) | 2 lines Fix saying the parking space number to the caller doing the parking ... ........ 2008-01-09 18:03 +0000 [r97530] Luigi Rizzo * channels/console_gui.c, channels/console_board.c, channels/console_video.h: Two changes: - support scrolling of message window; - simplify the code for creating a message window, and try it using a second one in the top of the keypad (where we echo the dialed number). The 'skin' that supports these two windows will be committed separately. 2008-01-09 17:30 +0000 [r97495] Kevin P. Fleming * /, codecs/codec_zap.c: Merged revisions 97491 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r97491 | kpfleming | 2008-01-09 11:21:14 -0600 (Wed, 09 Jan 2008) | 2 lines report the same message whether Zaptel does not have transcoder support loaded or no transcoders were found ........ 2008-01-09 16:59 +0000 [r97490] Philippe Sultan * /, channels/chan_gtalk.c: Merged revisions 97489 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r97489 | phsultan | 2008-01-09 17:44:24 +0100 (Wed, 09 Jan 2008) | 7 lines Set the caller id within the gtalk_alloc function. As underlined in issue #10437 by Josh, we need to prevent a possible memory leak. We only set the name part of the caller id, the number part is not relevant when dealing with JIDs. Closes issue #11549. ........ 2008-01-09 16:44 +0000 [r97488] Luigi Rizzo * channels/console_gui.c, channels/console_video.c, channels/console_board.c, channels/console_video.h: Implement keyboard handling, and use it to enter a number to dial in the 'message' area under the keypad. Now you can make calls using the keypad as a regular phone (or the keyboard for chars not present on the keypad) 2008-01-09 16:13 +0000 [r97451] Joshua Colp * /, apps/app_meetme.c: Merged revisions 97450 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r97450 | file | 2008-01-09 12:11:17 -0400 (Wed, 09 Jan 2008) | 6 lines Don't do conferencing totally in Zaptel if Monitor is running on the channel. (closes issue #11709) Reported by: BigJimmy Patches: patch-meetmerec uploaded by BigJimmy (license 371) ........ 2008-01-09 15:45 +0000 [r97421-97449] Kevin P. Fleming * channels/chan_zap.c, /: Merged revisions 97448 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r97448 | kpfleming | 2008-01-09 09:43:19 -0600 (Wed, 09 Jan 2008) | 2 lines pass the right variable to get an error string... oops ........ * channels/chan_zap.c, /: Merged revisions 97410 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r97410 | kpfleming | 2008-01-09 09:26:23 -0600 (Wed, 09 Jan 2008) | 2 lines add error number output to ioctl failure messages to help with debugging ........ 2008-01-09 12:23 +0000 [r97389-97390] Luigi Rizzo * channels/console_video.c, channels/console_video.h: implement the "console startgui" and "console stopgui" commands so you can start and stop the gui even outside of a call. This is convenient for testing, and also for using the keypad to pick up a call, and to dial a number (the latter not yet implemented, but should be close). * channels/chan_oss.c: make get_video_desc() return the active console if passed a null argument (channel). 2008-01-09 00:58 +0000 [r97364-97365] Tilghman Lesher * main/asterisk.c: New option in trunk, needs strdupa to be safe, too * /, main/editline/readline.c, main/cli.c: Merged revisions 97350 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r97350 | tilghman | 2008-01-08 18:44:14 -0600 (Tue, 08 Jan 2008) | 5 lines Allow filename completion on zero-length modules, remove a memory leak, remove a file descriptor leak, and make filename completion thread-safe. Patched and tested by tilghman. (Closes issue #11681) ........ 2008-01-09 00:18 +0000 [r97307-97309] Mark Michelson * /, apps/app_queue.c: Merged revisions 97308 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r97308 | mmichelson | 2008-01-08 18:17:40 -0600 (Tue, 08 Jan 2008) | 3 lines use the \retval doxygen command properly ........ * /, apps/app_queue.c: Merged revisions 97304 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r97304 | mmichelson | 2008-01-08 17:49:11 -0600 (Tue, 08 Jan 2008) | 5 lines Part 1 of N of adding doxygen comments to app_queue. I picked some of the most common functions used (which also happen to be some the biggest/ugliest functions too) to document first. I'm pretty new to doxygen so criticism is welcome. ........ 2008-01-08 23:51 +0000 [r97305] Tilghman Lesher * apps/app_voicemail.c: Add a new flag 'd' (with optional context) permitting any extension within that context to be entered as a new extension during the playback of a voicemail greeting. Patch inspired by bluecrow76, by tilghman. (Closes issue #7063) 2008-01-08 23:35 +0000 [r97280-97303] Luigi Rizzo * channels/console_board.c: add copyright (most of this code was written by Marta Carbone), remove some unused code, add/clarify some comments. * images/kpad2.jpg: Add the annotation for the textarea used for messages, and also change the background from white to something different to show that we can make use of fonts with transparent background. * images/font.png (added): add a font suitable for use with the console GUI. The background of this particular image is transparent so we can preserve the original background when we draw strings. * channels/console_gui.c, channels/console_video.c, channels/console_board.c (added), channels/Makefile: add support for textareas, used for various dialog windows on the gui. The main code to implement the textarea is in console_board.c, and uses a simple png image with the font, blitting characters on the designated areas of the main screen. Additionally we provide some annotations in the image used as a skin to indicate which areas are used for text messages. (images will be committed separately). At the moment the dialog area is only used to display a running counter, just as a proof of concept. 2008-01-08 21:56 +0000 [r97248] Terry Wilson * apps/app_queue.c: Initialize new variable to NULL 2008-01-08 21:28 +0000 [r97203-97208] Mark Michelson * configs/queues.conf.sample, CHANGES, apps/app_queue.c: Adding the option of specifying a second interface in a member definition for a queue. app_queue will monitor this second device's state for the member, even though it actually calls the first interface. This ability has been added for statically defined queue members, realtime queue members, and dynamic queue members added through the CLI, dialplan, or manager. (closes issue #11603, reported by acidv) 2008-01-08 21:01 +0000 [r97199-97200] Olle Johansson * channels/chan_console.c: Change reference to external library so it appears on the extref listing http://www.asterisk.org/doxygen/trunk/extref.html * res/res_jabber.c: Iksemel is alive in a new home. Release 1.3 is out with bug fixes. 2008-01-08 20:56 +0000 [r97198] Tilghman Lesher * main/autoservice.c, /, main/utils.c: Merged revisions 97194 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r97194 | tilghman | 2008-01-08 14:47:07 -0600 (Tue, 08 Jan 2008) | 3 lines Increase constants to where we're less likely to hit them while debugging. (Closes issue #11694) ........ 2008-01-08 20:52 +0000 [r97196-97197] Joshua Colp * channels/chan_sip.c: One line documentation ftw! * /, channels/chan_mgcp.c: Merged revisions 97195 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r97195 | file | 2008-01-08 16:48:20 -0400 (Tue, 08 Jan 2008) | 6 lines Fix various DTMF issues in chan_mgcp. (closes issue #11443) Reported by: eferro Patches: dtmf_control_hybrid-inband-mode.patch uploaded by eferro (license 337) ........ 2008-01-08 20:45 +0000 [r97193] Mark Michelson * /, apps/app_voicemail.c: Merged revisions 97192 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r97192 | mmichelson | 2008-01-08 14:42:07 -0600 (Tue, 08 Jan 2008) | 9 lines Making some changes designed to not allow for a corrupted mailstream for a vm_state. 1. Add locking to the vm_state retrieval functions so that no linked list corruption occurs. 2. Make sure to always grab the persistent vm_state when mailstream access is necessary. 3. Correct an incorrect return value in the init_mailstream function. (closes issue #11304, reported by dwhite) ........ 2008-01-08 20:06 +0000 [r97153-97154] Joshua Colp * channels/chan_sip.c: Move common code for setting T38 capabilities and fix a bug with fax detection in the SIP RTP read callback. It's still sort of silly... but more on that later. (closes issue #11239) Reported by: dimas Patches: sipt38prop.patch uploaded by dimas (license 88) * funcs/func_groupcount.c, /: Merged revisions 97152 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r97152 | file | 2008-01-08 15:53:52 -0400 (Tue, 08 Jan 2008) | 4 lines If no group has been provided to the GROUP_COUNT dialplan function then use the first one specific to the channel. (closes issue #11077) Reported by: m4him ........ 2008-01-08 19:06 +0000 [r97125] Tilghman Lesher * /, channels/chan_sip.c, main/asterisk.c: Merged revisions 97077 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r97077 | tilghman | 2008-01-08 12:02:13 -0600 (Tue, 08 Jan 2008) | 3 lines Apply multiple crash fixes, found in issue #11386, but not completely closing that issue. ........ 2008-01-08 18:42 +0000 [r97041-97103] Joshua Colp * /, apps/app_queue.c: Merged revisions 97093 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r97093 | file | 2008-01-08 14:36:40 -0400 (Tue, 08 Jan 2008) | 4 lines Make app_queue calls work with directed pickup. (closes issue #11700) Reported by: jbauer ........ * utils/extconf.c: Make ast_atomic_fetchadd_int_slow magically appear in extconf. (closes issue #11703) Reported by: dmartin 2008-01-07 23:03 +0000 [r96988] Luigi Rizzo * channels/console_gui.c: add support for cropping the keypad image while displaying it. This way it can contain additional elements (e.g. fonts, buttons, widgets) without having to use a zillion files to store them. 2008-01-07 22:31 +0000 [r96987] Mark Michelson * apps/app_voicemail.c: Explicitly make literal constants long where they are expected to be. 2008-01-07 21:12 +0000 [r96936] Jason Parker * main/config.c: Display a message if no config mappings are found with "core show config mappings". Closes issue #11704, patch by kshumard. 2008-01-07 21:10 +0000 [r96934-96935] Mark Michelson * apps/app_voicemail.c: Document some weird casting magic that's necessary to interface with the c-client * doc/tex/imapstorage.tex, apps/app_voicemail.c: Adding user-configurable TCP timeout settings to IMAP voicemail. This could go a long way towards preventing unexplainable hangs experienced by people. In the case of MWI hangs, this also will mean that the SIP port isn't blocked anymore. (closes issue #11665, reported by yehavi) 2008-01-07 20:48 +0000 [r96885-96933] Russell Bryant * /, configs/extensions.conf.sample: Merged revisions 96932 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r96932 | russell | 2008-01-07 14:47:52 -0600 (Mon, 07 Jan 2008) | 10 lines Merged revisions 96931 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r96931 | russell | 2008-01-07 14:46:22 -0600 (Mon, 07 Jan 2008) | 2 lines Change misery.digium.com to pbx.digium.com ........ ................ * configs/http.conf.sample: Add a note about viewing the default set of documentation using the built-in http server * Makefile: If the HTML documentation exists, install it in the static-http/docs directory so that it can be viewed through the Asterisk http server if it is turned on. * build_tools/prep_tarball: Build the HTML version of the doc files for tarballs, as well * res/res_smdi.c, /: Merged revisions 96884 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r96884 | russell | 2008-01-07 10:39:23 -0600 (Mon, 07 Jan 2008) | 3 lines Don't crash if something happens when setting up an SMDI interface and it gets destroyed before the SMDI port handling thread gets created. ........ 2008-01-07 16:17 +0000 [r96862] Kevin P. Fleming * formats/format_sln16.c (added): add a file-format driver for 16KHz signed linear... which may or may not work 2008-01-07 15:52 +0000 [r96858] Joshua Colp * main/manager.c, main/loader.c: Move ModuleLoad and ModuleCheck manager commands from loader.c to manager.c. Previously they would get registered twice because of the way manager.c operates. (closes issue #11699) Reported by: caio1982 Patches: manager_module_commands1.diff uploaded by caio1982 (license 22) 2008-01-07 15:06 +0000 [r96776-96836] Luigi Rizzo * channels/console_gui.c: update comments to reflect reality (or at least planned behaviour). minor code cleanups * channels/console_gui.c: resolve a load-time problem avoiding a call to console_do_answer. On passing, fix dialling from the keypad. 2008-01-05 23:05 +0000 [r96645-96743] Russell Bryant * res/snmp/agent.c: Convert this file over the new method of getting the Asterisk version. (I don't have this building on this machine, so caio1982 on IRC is going to test it for me. :) ) * Makefile, funcs/func_version.c, main/manager.c, channels/chan_sip.c, main/Makefile, build_tools/make_version_c (added), include/asterisk/version.h (added), res/res_agi.c, main, main/http.c, build_tools/make_version_h (removed), include/asterisk, main/asterisk.c: Now that the version.h file was getting properly regenerated every time the svn revision changed, every module that used the version was getting rebuilt after every svn update. This severly annoyed me pretty quickly, so I have improved the situation. Now, instead of generating version.h, main/version.c is generated. version.c includes the version information, as well as a couple of API calls for modules to retrieve the version. So now, only version.c will get rebuilt, and the main asterisk binary relinked, which is must faster than rebuilding http.c, manager.c, asterisk.c, relinking the asterisk binary, chan_sip.c, func_version.c, res_agi ... The only minor change in behavior here is that the version information reported by chan_sip, for example, is the version of the Asterisk core, and not necessarily the Asterisk version that the chan_sip module came from. * main/pbx.c: Print out the name of a function being registered in color, just like the name of applications when they get registered. * UPGRADE.txt: Add a note about changing modules.conf since another console channel driver is now present that can not be used at the same time as chan_alsa or chan_oss. * channels/chan_console.c: Add the URL to the home page for portaudio. Also add the location of the svn repository to check out portaudio v19. * /, main/devicestate.c: Merged revisions 96644 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r96644 | russell | 2008-01-04 20:09:19 -0600 (Fri, 04 Jan 2008) | 2 lines Don't pass an empty string as the device name. ........ 2008-01-05 01:05 +0000 [r96621] Kevin P. Fleming * channels/chan_usbradio.c: improve chan_usbradio to use indications just like chan_alsa/chan_oss do now 2008-01-04 23:12 +0000 [r96576] Tilghman Lesher * /, main/devicestate.c: Merged revisions 96575 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r96575 | tilghman | 2008-01-04 17:03:40 -0600 (Fri, 04 Jan 2008) | 7 lines Fix the problem of notification of a device state change to a device with a '-' in the name. Could probably do with a better fix in trunk, but this bug has been open way too long without a better solution. Reported by: stevedavies Patch by: tilghman (Closes issue #9668) ........ 2008-01-04 22:57 +0000 [r96574] Jason Parker * /, res/res_features.c: Merged revisions 96573 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 (Closes issue #11237) ........ r96573 | qwell | 2008-01-04 16:55:56 -0600 (Fri, 04 Jan 2008) | 4 lines Properly continue in the dialplan if using PARKINGEXTEN and the slot is full. Issue 11237, patch by me. ........ 2008-01-04 19:35 +0000 [r96547] Tilghman Lesher * /, channels/chan_sip.c: Merged revisions 96525 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r96525 | tilghman | 2008-01-04 13:27:25 -0600 (Fri, 04 Jan 2008) | 4 lines If you change the bindaddr in sip.conf to a non-bound address and reload, sip goes kablooie. Reported and patched by: one47 (Closes issue #11535) ........ 2008-01-04 17:21 +0000 [r96500] Kevin P. Fleming * channels/chan_zap.c, configure, include/asterisk/autoconfig.h.in, configure.ac, acinclude.m4: [commit message] (closes issue #10393) Reported by: tzafrir Patches: chan_alarm_asterisk.diff uploaded by tzafrir (license 46) (modified by me and added configure script support) 2008-01-04 17:19 +0000 [r96499] Philippe Sultan * res/res_jabber.c: Use SASL DIGEST-MD5 authentication over unsecured network connections only. This authentication mechanism is implemented under the iksemel API, which makes use of GnuTLS, whereas we use OpenSSL. Note : there's ongoing dicsussion at the SASL IETF WG in order to deprecate SASL DIGEST-MD5, see http://ietfreport.isoc.org/ids-wg-sasl.html. 2008-01-04 16:21 +0000 [r96450] Russell Bryant * channels/chan_zap.c, /: Merged revisions 96449 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r96449 | russell | 2008-01-04 10:19:22 -0600 (Fri, 04 Jan 2008) | 7 lines Make use of the temporary channel pointer while the pvt is unlocked. (closes issue #11675) Reported by: flefoll Patches: chan_zap.c.patch-store-owner-before-unlock uploaded by flefoll (license 244) ........ 2008-01-03 23:14 +0000 [r96397-96398] Kevin P. Fleming * Makefile: we have to *always* use a completely silent 'make' invocation for generating the module embedding rules * Makefile: there was no reason to add this define for non-Solaris platforms 2008-01-03 22:46 +0000 [r96395] Russell Bryant * /, channels/chan_iax2.c: Merged revisions 96394 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r96394 | russell | 2008-01-03 16:44:22 -0600 (Thu, 03 Jan 2008) | 3 lines Don't crash if the iax2 pvt structure has been destroyed before we get to this point (closes issue #11672, reported by snuffy, patched by me) ........ 2008-01-03 21:58 +0000 [r96301-96368] Tilghman Lesher * include/asterisk/channel.h: Document recent API addition * res/res_config_pgsql.c, /: Merged revisions 96318 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r96318 | tilghman | 2008-01-03 15:37:02 -0600 (Thu, 03 Jan 2008) | 4 lines Missed initialization caused crash. Reported and fixed by: tiziano (Closes issue #11671) ........ * main/channel.c: Allow the uniqueid to be used for searching for a channel in the list. Reported and initially patched by: michael-fig (Closes issue #11340) 2008-01-03 20:04 +0000 [r96245-96272] Kevin P. Fleming * Makefile, tests/Makefile (added), tests/test_skel.c (added), tests (added): add some simple infrastructure for modules to be used for testing parts of Asterisk * channels/answer.h (removed), channels/ring10.h (removed), channels/busy.h (removed), channels/ringtone.h (removed), channels/Makefile, channels/chan_oss.c, channels/gentone.c (removed), channels: eliminiate sound_thread() and other stuff from chan_oss since Asterisk indications can handle it remove gentone and all the headers containing tones that are no longer needed * channels/chan_alsa.c: coding guidelines cleanup remove background thread and all sound generation mechanisms, as the built-in indications can handle everything that is needed 2008-01-03 14:47 +0000 [r96221] Christian Richter * channels/chan_misdn.c, /: Merged revisions 96198 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r96198 | crichter | 2008-01-03 13:08:40 +0100 (Do, 03 Jan 2008) | 1 line when overlapdial was used and no number was dialed, the call was dropped, now we just jump into the s extension, which makes a lot more sense. ........ 2008-01-03 06:16 +0000 [r96147-96174] Tilghman Lesher * res/res_agi.c: Add coordination between AMI and AGI applications, with an asyncagi method Feature proposed and patched by: moy (Closes issue #11282) * apps/app_mp3.c, apps/app_ices.c, main/asterisk.c: Compatibility fix for OpenBSD Report and fix by: mvanbaak (Closes issue #11669) 2008-01-02 23:48 +0000 [r96103] Mark Michelson * /, apps/app_queue.c: Merged revisions 96102 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r96102 | mmichelson | 2008-01-02 17:46:02 -0600 (Wed, 02 Jan 2008) | 4 lines We need to reset the membername to NULL on each iteration of this loop, otherwise the result is that multiple members can have the same name, since the variable was not reset on each iteration of the loop. ........ 2008-01-02 23:22 +0000 [r96076-96079] Russell Bryant * channels/chan_console.c: Add support for generating a ringing sound on an incoming call. This is a bit of a hack. It just asks the core to generate the same tone that it would when you hear ringback when making an outbound call. But hey, it works, and you get the localized ring tone for the appropriate language set on the channel. * channels/chan_console.c: Note that this module doesn't actually play a ringing sound for an incoming call ... oops * channels/chan_console.c: Show the correct CLI command to answer the call 2008-01-02 22:41 +0000 [r96073] Kevin P. Fleming * channels/chan_zap.c: actually parse and store echocan parameters from zapata.conf... this *should* work 2008-01-02 22:40 +0000 [r96071] Joshua Colp * configure, include/asterisk/autoconfig.h.in, configure.ac: Don't use AST_C_DEFINE_CHECK for the two pthread things that may not actually be definitions, they could be enums for example. 2008-01-02 22:29 +0000 [r96028] Mark Michelson * channels/chan_zap.c: Add curly braces around a compound if statement so that trunk will build properly 2008-01-02 21:51 +0000 [r96019] Kevin P. Fleming * channels/chan_zap.c, configs/zapata.conf.sample: another checkpoint... chan_zap can now use the new ZT_ECHOCAN_PARAMS ioctl if it is present, but doesn't parse any supplied parameters yet (this implementation is not very memory efficient as the parameters and their values will be duplicated for each channel that has the same settings, but we can worry about that later once it is working) 2008-01-02 21:49 +0000 [r96018] Russell Bryant * main/libresample/include/libresample.h: Add doxygen documentation to libresample.h while it's still fresh on my mind 2008-01-02 21:08 +0000 [r95994] Mark Michelson * funcs/func_odbc.c, channels/chan_agent.c, funcs/func_strings.c, apps/app_rpt.c: Change instances of AST_NONSTANDARD_APP_ARGS(foo, bar, ',') to AST_STANDARD_APP_ARGS(foo, bar) (closes issue #11668, reported and patched by mvanbaak) 2008-01-02 20:26 +0000 [r95947] Joshua Colp * /, channels/chan_sip.c: Merged revisions 95946 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r95946 | file | 2008-01-02 16:24:09 -0400 (Wed, 02 Jan 2008) | 4 lines Allocate a SIP refer structure when performing a transfer using BYE with Also so that the transfer information is properly stored. (AST-2008-001) (closes issue #11637) Reported by: greyvoip ........ 2008-01-02 20:23 +0000 [r95944-95945] Mark Michelson * apps/app_queue.c: Since ',' is the standard argument separator in trunk, change app_queue to use AST_STANDARD_APP_ARGS instead of AST_NONSTANDARD_APP_ARGS for determining member data. * include/asterisk/app.h: Fix a typo in a comment. AST_STANDARD_APP_ARGS uses ',' as the separator, not '|'. 2008-01-02 19:47 +0000 [r95893-95939] Kevin P. Fleming * channels/chan_zap.c: clean up hwgain CLI command and improve docs for swgain CLI command * configure, include/asterisk/autoconfig.h.in, configure.ac, acinclude.m4: improve AC_C_DEFINE_CHECK to not try to evaluate the macro being checked for, but just check for its existence finish implementation of check for Zaptel HWGAIN support add check for Zaptel ECHOCANCEL_PARAMS support * codecs/Makefile, include/asterisk/libresample.h (added), codecs/codec_resample.c: and now just to keep the libresample party going... if the functions from libresample are going to be in the main Asterisk binary, it makes sense for the header that defines them to be available without any special CFLAGS and to out-of-tree modules building against /usr/include/asterisk * channels/chan_zap.c: umm... this did not compile on x86-64, and could not possibly have worked on any platform as it was passing string pointers to a function expecting ints 2008-01-02 18:05 +0000 [r95891] Mark Michelson * /, apps/app_queue.c: Merged revisions 95890 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r95890 | mmichelson | 2008-01-02 11:51:22 -0600 (Wed, 02 Jan 2008) | 9 lines A change to improve the accuracy of queue logging in the case where a member does not answer during the specified timeout period. Prior to this change, there was a small chance that the member name recorded in this case would be blank. Also prior to this change, if using the ringall strategy, if no one answered the call during the specified timeout, the member name listed in the queue log would randomly be one of the members that was rung. (closes issue #11498, reported and tested by hloubser, patched by me) ........ 2008-01-02 17:38 +0000 [r95888] Jason Parker * apps/app_osplookup.c: Update osplookup documentation to use commas instead of pipes. Closes issue #11666, patch by Laureano. 2008-01-02 16:20 +0000 [r95864] Russell Bryant * main/Makefile, main/translate.c: For some odd reason, the last set of libresample build changes from Kevin did not work for everyone, but it did for some. This set of changes makes trunk start again for those having problems. Instead of building libresample as a static library, it just links the object files in directly with the asterisk binary. 2008-01-02 14:53 +0000 [r95816-95841] Kevin P. Fleming * channels/Makefile: fix some long-time breakage that kept chan_misdn from being embedded * channels/Makefile: use the proper technique for including submodules so that embedding will work * CHANGES: note that chan_console requires portaudio v19 * configure, configure.ac: actually check for a function present in libiconv (don't know how this test could have worked before) and don't do the check on Linux/GNU systems because libiconv is not present there and attempting to link with '-liconv' always fails (it's not necessary as the iconv functionality is always available) * main/libresample/src/filterkit.h, main/libresample/src/resample.c, main/libresample/win/libresample.dsp, main/libresample/configure, main/libresample/Makefile.in, res/Makefile, main/libresample/configure.in, main/libresample/src, main/libresample/tests/testresample.c, main/libresample/win/libresample.vcproj, main/libresample/tests/compareresample.c, main/libresample/tests, codecs/codec_resample.c, res/res_resample.c (removed), main/libresample/README.txt, main/libresample/src/resamplesubs.c, main/libresample/tests/resample-sndfile.c, main/libresample/src/configtemplate.h, main/libresample/install-sh, main/Makefile, main/translate.c, main/libresample/include, main/libresample/src/resample_defs.h, codecs/Makefile, main/libresample/config.guess, main/libresample/config.sub, main/libresample/win, main/libresample/LICENSE.txt, main/libresample (added), main/libresample/Makefile.asterisk, build_tools/strip_nonapi, res/libresample (removed), main/libresample/src/filterkit.c, main/libresample/include/libresample.h: go back to including libresample in the main Asterisk binary, but this time including a small hack to ensure that it does get linked in (and also modify the strip_nonapi script to leave the resample_ symbols alone) 2008-01-02 11:34 +0000 [r95794] Philippe Sultan * res/res_jabber.c: Set stream flags to zero upon initialization. When the XMPP over TLS/SSL connection resets for some reason, it is wrongly believed as being secured, which makes the re-connection process endlessly fail. This was reported by mvanbaak in issue #11644. 2008-01-02 09:16 +0000 [r95771-95772] Luigi Rizzo * main/loader.c: some cleanup of this code while I am trying to debug a problem with gdb dying while debugging asterisk. The problem seems to be related with a race in the handling of module_list, which in turn is triggeded by calling dlopen() on a system which uses initializers to create locks. * include/asterisk/module.h: There are three instances of the module definition macros, which make maintaining this file very error prone. This commit merges the embedded and !embedded versions, and fixes the C++ version. Eventually we should move to a single version of the macro. Too bad C++ doesn't like the C-style struct initializers .foo = some_value 2008-01-02 04:33 +0000 [r95697-95746] Russell Bryant * res/libresample/src/resample_defs.h, res/libresample/src/resample.c: Don't make libresample print out debugging output * main/translate.c: Make the translation table show slin16 * apps/app_meetme.c: fix a spacing issue introduced in revision 95443. * main/Makefile, res/libresample/README.txt, res/Makefile, res/libresample/install-sh, res/libresample/configure, res/libresample/Makefile.in, res/libresample/include, codecs/Makefile, res/libresample/configure.in, res/libresample/src, res/libresample/config.guess, main/libresample (removed), res/libresample/config.sub, res/libresample/win, codecs/codec_resample.c, res/libresample/LICENSE.txt, res/libresample (added), res/libresample/Makefile.asterisk, res/libresample/tests, res/res_resample.c (added): Instead of linking libresample into the main Asterisk binary, build it as res_resample, and mark codec_resample as dependent upon res_resample. This prevents the linker from optimizing away libresample, and also makes it so the libresample code isn't linked in to multiple places. (I have another module in a branch that needs it, too.) 2008-01-01 23:55 +0000 [r95671-95673] Luigi Rizzo * channels/console_gui.c: call directly the cli command to implement hangup. * channels/vcodecs.c: prevent a panic when destroying a channel with no incoming video. * channels/console_video.c: remove a leftover sleep(1) used for debugging 2008-01-01 23:09 +0000 [r95648] Joshua Colp * codecs/Makefile: Fix building of codec_resample on platforms other then Cygwin. On everything else it actually gets built after codec_resample, so you can't exactly link it in since it doesn't exist. 2008-01-01 22:21 +0000 [r95624-95625] Luigi Rizzo * codecs/Makefile, codecs/codec_resample.c: make codec_resample build on __CYGWIN__, and make it load on FreeBSD (and probably other systems as well). Both need libresample.a to be specified in the linking phase, and cygwin needs as other BSD. The checks for OS-specific headers should really be moved to some common header though. * build_tools/menuselect-deps.in, configure, include/asterisk/autoconfig.h.in, configure.ac, funcs/func_iconv.c, makeopts.in: implement "configure" checks for libiconv, and add the iconv dependency for func_iconv. This fixes some build issues on CYGWIN and FreeBSD and probably other platforms where libiconv is not there by default 2007-12-31 23:44 +0000 [r95578] Mark Michelson * main/pbx.c, /: Merged revisions 95577 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r95577 | mmichelson | 2007-12-31 17:43:13 -0600 (Mon, 31 Dec 2007) | 9 lines Avoiding a potentially bad locking situation. ast_merge_contexts_and_delete writelocks the conlock, then calls ast_hint_extension, which attempts to readlock the same lock. Recursion with read-write locks is dangerous, so the inner lock needs to be removed. I did this by copying the "guts" of ast_hint_extension into ast_merge_contexts_and_delete (sans the extra lock). (this change is inspired by the locking problems seen in issue #11080, but I have no idea if this is the problematic area experienced by the reporters of that issue) ........ 2007-12-31 22:41 +0000 [r95501-95550] Russell Bryant * codecs/codec_resample.c: Use float.h to fix the build on FreeBSD. Also, add some other platforms as they are likely the same. * channels/chan_console.c: Update chan_console to natively use a 16 kHz sample rate. If it is talking to an 8 kHz endpoint, then codec_resample will automatically be used to properly resample the audio before sending it to/from chan_console. * main/libresample/src/filterkit.h, main/libresample/README.txt, main/libresample/tests/resample-sndfile.c, main/libresample/src/resamplesubs.c, main/Makefile, main/libresample/install-sh, main/libresample/src/configtemplate.h, main/libresample/src/resample.c, main/libresample/win/libresample.dsp, main/libresample/configure, main/libresample/Makefile.in, main/libresample/include, CHANGES, main/libresample/src/resample_defs.h, main/libresample/configure.in, main/libresample/src, main/libresample/config.guess, codecs/Makefile, main/libresample/tests/testresample.c, codecs/slin_resample_ex.h (added), main/libresample/config.sub, main/libresample/win, main/libresample/win/libresample.vcproj, main/libresample/LICENSE.txt, main/libresample (added), main/libresample/Makefile.asterisk, main/libresample/tests, main/libresample/tests/compareresample.c, codecs/codec_resample.c (added), main/libresample/src/filterkit.c, main/libresample/include/libresample.h: Merge changes from team/russell/codec_resample This commit imports libresample for use in Asterisk. It also adds a new codec module, codec_resample. This module uses libresample to re-sample signed linear audio between 8 kHz and 16 kHz. It also provides an alternative for converting between 16 kHz G.722 and 8 kHz signed linear when using G.722, which will likely be useful as some people have complained about volume issues when the current codec_g722 converts to 8 kHz signed linear. But, to test this, you will have to disable the g722-to-slin and g722-to-slin16 translators in codec_g722.c. 2007-12-31 20:33 +0000 [r95490] Tilghman Lesher * /, funcs/func_env.c: Merged revisions 95470 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r95470 | tilghman | 2007-12-31 14:27:26 -0600 (Mon, 31 Dec 2007) | 3 lines Allow the default "0" to be returned if the STAT fails (Closes issue #11659) ........ 2007-12-31 18:46 +0000 [r95443] Mark Michelson * apps/app_meetme.c: Fix a compiler warning (closes issue #11658, reported and patched by eliel) 2007-12-31 16:13 +0000 [r95383-95412] Russell Bryant * configs/console.conf.sample (added), configs/modules.conf.sample, channels/chan_console.c (added), CHANGES: Merge the main set of changes from team/russell/chan_console. Add a new console channel driver, chan_console, which is a console channel driver that uses portaudio as a cross platform audio interface. It was written to provide a console channel driver that works with Mac CoreAudio, but it supports a number of other audio interfaces, as well, including OSS and ALSA. It could one day be the single console channel driver, but does not yet have as many features as chan_oss. * include/asterisk/channel.h: fix a spelling error in a comment * include/asterisk/config.h: Add CV_STRINGFIELD() macro. This lets you set a config variable to a string field. (from team/russell/chan_console) * configure, include/asterisk/autoconfig.h.in: Regenerate configure script to include check for portaudio. * build_tools/menuselect-deps.in, configure.ac, makeopts.in: Add configure script checking for portaudio. 2007-12-29 02:02 +0000 [r95262-95313] Luigi Rizzo * channels/vcodecs.c, channels/console_video.c, channels/Makefile, channels/console_video.h, channels/vgrabbers.c (added): Move grabbers definitions to a separate file, vgrabbers.c, so it is easier to add more entries. This required moving struct grab_desc to the common header, and adding an entry in the Makefile. On passing, cleanup some comments and file headers (some are still missing). * channels/console_gui.c, channels/console_video.c: virtualize the interface for video grabbers, which should make it easier to add support for more grabbers (V4L2, firewire, and so on). * channels/console_video.c: Add a few entries up to 1408x1152 in the table of known video resolutions. This makes it very convenient to enlarge images using the right-click on the video window. * channels/vcodecs.c, channels/console_video.c: change the interface of video encapsulation routines, they only need the buffer and mtu as input. * channels/console_gui.c, channels/vcodecs.c, channels/console_video.c, channels/console_video.h: various rearrangements and renaming of console_video stuff 2007-12-28 18:39 +0000 [r95233] Mark Michelson * apps/app_queue.c: The diff for this change looks really bad, but all I did here was decrease the indentation of most of the queue_exec function by reversing the logic of an if statement. This change makes the function comply better with the coding guidelines. Since this change is purely a cosmetic change to the code, I am only committing the change to trunk. 2007-12-28 18:26 +0000 [r95192] Russell Bryant * /, channels/chan_sip.c: Merged revisions 95191 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r95191 | russell | 2007-12-28 12:24:59 -0600 (Fri, 28 Dec 2007) | 6 lines Remove duplicate increment of the header count in the add_header() function. (closes issue #11648) Reported by: makoto Patch provided by sergee, committed patch by me, inspired by comments from putnopvut ........ 2007-12-28 16:12 +0000 [r95167] Mark Michelson * apps/app_amd.c, CHANGES: Some changes to app_amd. The channel name is printed in verbose messages maximumWordLength option added. Duration of words that do not meet the minimum word duration will be logged The duration of pre-greeting silence will be logged Only consider us in the greeting if we actually detected a valid word duration. (closes issue #11650, reported and patched by davevg) 2007-12-28 08:57 +0000 [r95139] Luigi Rizzo * channels/console_video.c: fix a small bug in printing out geometries - wrong input. 2007-12-28 00:17 +0000 [r95096] Mark Michelson * /, apps/app_queue.c: Merged revisions 95095 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r95095 | mmichelson | 2007-12-27 18:16:15 -0600 (Thu, 27 Dec 2007) | 8 lines I found a bug while browsing the queue code and managed to reproduce it in a small setup. If a queue uses the ringall strategy, it was possible through unfortunate coincidence for a single member at a given penalty level to make app_queue think that all members at that penalty level were unavailable and cause the members at the next penalty level to be rung. With this patch, we will only move to the next penalty level if ALL the members at a given penalty level are unreachable. ........ 2007-12-27 23:32 +0000 [r95073] Luigi Rizzo * apps/app_dictate.c, apps/app_mp3.c, apps/app_voicemail.c: remove more unnecessary casts for NULL. main/say.c is a big offender in this respect. 2007-12-27 23:28 +0000 [r95070] Jason Parker * doc/asterisk.8, main/asterisk.c: Fix -s socket option, and document it as well. Closes issue #11645, patch by Laureano. 2007-12-27 23:13 +0000 [r95068-95069] Luigi Rizzo * apps/app_ices.c, apps/app_queue.c, apps/app_voicemail.c: NULL does not need to be cast to (char *) * channels/chan_oss.c: remove useless casts 2007-12-27 21:41 +0000 [r95025] Russell Bryant * main/channel.c, /: Merged revisions 95024 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r95024 | russell | 2007-12-27 15:40:02 -0600 (Thu, 27 Dec 2007) | 9 lines Don't report a syntax error when an empty string is passed to ast_get_group. Just return 0. (closes issue #11540) Reported by: tzafrir Patches: group_empty.diff uploaded by tzafrir (license 46) -- slightly changed by me ........ 2007-12-27 20:11 +0000 [r94978] Mark Michelson * /, main/io.c: Merged revisions 94977 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r94977 | mmichelson | 2007-12-27 14:09:06 -0600 (Thu, 27 Dec 2007) | 3 lines Fixing a typo in a comment. ........ 2007-12-27 17:34 +0000 [r94908-94934] Joshua Colp * /, channels/chan_h323.c: Merged revisions 94924 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r94924 | file | 2007-12-27 13:32:15 -0400 (Thu, 27 Dec 2007) | 6 lines Include types.h in chan_h323 as without it it can not be compiled on some operating systems like FreeBSD to name one. (closes issue #11585) Reported by: sobomax Patches: chan_h323.c.diff uploaded by sobomax (license 359) ........ * /, channels/chan_sip.c: Merged revisions 94905 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r94905 | file | 2007-12-27 13:27:11 -0400 (Thu, 27 Dec 2007) | 4 lines Use ast_strlen_zero to see if our_contact is set or not on the dialog. It is possible for it to be a pointer to NULL. (closes issue #11557) Reported by: FuriousGeorge ........ 2007-12-27 17:26 +0000 [r94904] Luigi Rizzo * channels/console_gui.c, channels/console_video.c: more localization of gui stuff 2007-12-27 17:18 +0000 [r94903] Mark Michelson * doc/manager_1_1.txt: Adding documentation for new manager actions and events in app_queue 2007-12-27 16:51 +0000 [r94902] Luigi Rizzo * CHANGES: clarify the type of video support in chan_oss 2007-12-27 16:11 +0000 [r94830-94877] Russell Bryant * codecs/codec_g722.c: I went looking for where we downloaded the g722 implementation and came across these two links. So, I'm adding them so they are available for reference later. * /, main/translate.c, include/asterisk/translate.h: Merged revisions 94828-94829 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r94828 | russell | 2007-12-27 08:33:21 -0600 (Thu, 27 Dec 2007) | 9 lines Change ast_translator_best_choice() to only pay attention to audio formats. This fixes a problem where Asterisk claims that a translation path can not be found for channels involving video. (closes issue #11638) Reported by: cwhuang Tested by: cwhuang Patch suggested by cwhuang, with some additional changes by me. ........ r94829 | russell | 2007-12-27 08:44:29 -0600 (Thu, 27 Dec 2007) | 2 lines Use the constant that I really meant to use here ... ........ 2007-12-27 09:13 +0000 [r94826-94827] Olle Johansson * funcs/func_dialplan.c: This function checks more than just contexts... * apps/app_pickupchan.c: - Add Copyright - Doxygen fixes Note: - This application needs better documentation and a RESULT code in the dialplan. 2007-12-27 01:03 +0000 [r94825] Kevin P. Fleming * main/manager.c, /: Merged revisions 94824 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r94824 | kpfleming | 2007-12-26 18:01:47 -0700 (Wed, 26 Dec 2007) | 2 lines make this comment explain the situation in an even more explicit fashion ........ 2007-12-27 00:48 +0000 [r94819-94823] Luigi Rizzo * channels/console_gui.c: more steps to decouple the gui from the rest of the code. * channels/console_gui.c, channels/console_video.c, channels/console_video.h: Enable building the code even if SDL is not present (similarly, SDL is also detected at runtime). Now we should be able to stream video even without a rendering device (useful for remote monitoring). * channels/console_gui.c, channels/console_video.c: more localizations around sdl_setup * channels/console_gui.c: use fread instead of mmap to read in the comment area from the keypad. fread is simpler and more portable, and there is no performance gain in using mmap. * images/kpad2.jpg: update the region description with an empty line at the beginning. 2007-12-26 22:38 +0000 [r94818] Tilghman Lesher * build_tools/cflags.xml, channels/chan_zap.c: Allow more spans than 32. Also, rearrange compiler flags so the most often used flags appear closer to the top. Reported by: tzafrir Patch by: tzafrir,tilghman (Closes issue #11528) 2007-12-26 22:29 +0000 [r94817] Luigi Rizzo * channels/console_gui.c, channels/console_video.c: another bunch of gui localizations 2007-12-26 22:14 +0000 [r94814] Jason Parker * apps/app_exec.c: Make 'else' argument to ExecIf optional. Clean up the description and usage text a bit. Closes issue #11564, patch by pnlarsson (with some extra cleanup by me). 2007-12-26 22:10 +0000 [r94810-94813] Luigi Rizzo * channels/console_gui.c, channels/console_video.c: more localization of sdl stuff * channels/console_gui.c, channels/console_video.c, channels/console_video.h: move more gui stuff into console_gui.c 2007-12-26 20:49 +0000 [r94809] Tilghman Lesher * main/manager.c, /: Merged revisions 94808 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r94808 | tilghman | 2007-12-26 14:43:38 -0600 (Wed, 26 Dec 2007) | 6 lines Workaround for what is probably a glibc bug (but we'll see this crop up again and again, if we don't add the workaround). Reported by: rolek Patch by: tilghman (Closes issue #11601, closes issue #11426) ........ 2007-12-26 20:02 +0000 [r94806] Jason Parker * pbx/pbx_loopback.c, apps/app_zapbarge.c, pbx/pbx_spool.c, apps/app_authenticate.c, apps/app_zapscan.c, apps/app_zapras.c, apps/app_alarmreceiver.c, apps/app_amd.c, pbx/pbx_realtime.c, pbx/pbx_dundi.c, apps/app_zapateller.c, pbx/pbx_config.c, pbx/pbx_gtkconsole.c, apps/app_adsiprog.c, apps/app_cdr.c: Use defined return values in load_module in more places. (closes issue #11096) Patches: pbx_config.c.patch uploaded by moy (license 222) pbx_dundi.c.patch uploaded by moy (license 222) pbx_gtkconsole.c.patch uploaded by moy (license 222) pbx_loopback.c.patch uploaded by moy (license 222) pbx_realtime.c.patch uploaded by moy (license 222) pbx_spool.c.patch uploaded by moy (license 222) app_adsiprog.c.patch uploaded by moy (license 222) app_alarmreceiver.c.patch uploaded by moy (license 222) app_amd.c.patch uploaded by moy (license 222) app_authenticate.c.patch uploaded by moy (license 222) app_cdr.c.patch uploaded by moy (license 222) app_zapateller.c.patch uploaded by moy (license 222) app_zapbarge.c.patch uploaded by moy (license 222) app_zapras.c.patch uploaded by moy (license 222) app_zapscan.c.patch uploaded by moy (license 222) 2007-12-26 20:01 +0000 [r94805] Luigi Rizzo * channels/console_gui.c, channels/vcodecs.c, channels/console_video.c, channels/console_video.h: more preparation for untangling of the various console_video stuff 2007-12-26 19:09 +0000 [r94796-94802] Russell Bryant * main/autoservice.c, /: Merged revisions 94801 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r94801 | russell | 2007-12-26 13:04:31 -0600 (Wed, 26 Dec 2007) | 4 lines Just in case the AST_FLAG_END_DTMF_ONLY flag was already set before starting autoservice, remember it and ensure that the channel has the same setting when autoservice gets stopped. (pointed out by d1mas, patched up by me) ........ * funcs/func_dialplan.c (added), CHANGES: Add a new dialplan function, DIALPLAN_EXISTS(), which allows you to check for the existence of a dialplan target. (closes issue #11579) Reported by: irroot Patches: func_dialplan2.c uploaded by irroot (license 52) -- Additional changes by me. * main/autoservice.c, /: Merged revisions 94797 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r94797 | russell | 2007-12-26 12:46:39 -0600 (Wed, 26 Dec 2007) | 4 lines When a channel is in autoservice, mark a flag on the channel that says that we only care about the END of a digit. That way, no magic digit emulation stuff will happen when all we're doing is queueing up END frames. ........ * main/channel.c: Leave a note for a minor bug that was pointed out by d1mas 2007-12-26 18:05 +0000 [r94795] Tilghman Lesher * channels/chan_zap.c: Convert raw bits for callprogress bitfield to use constants, for greater code clarity Reported by: dimas Patch by: dimas (Closes issue #11280) 2007-12-26 17:26 +0000 [r94787-94794] Russell Bryant * /, res/res_features.c: Merged revisions 94793 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r94793 | russell | 2007-12-26 11:24:17 -0600 (Wed, 26 Dec 2007) | 3 lines Don't try to send a parked call back to itself. (closes issue #11622, reported by djrodman, patched by me) ........ * Makefile, /: Merged revisions 94789 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r94789 | russell | 2007-12-26 11:00:03 -0600 (Wed, 26 Dec 2007) | 5 lines List include/asterisk/version.h as a .PHONY target because we want the commands listed for this target to be executed regardless of whether the file exists or not. This fixes having the version not up to date when running from svn. (closes issue #11619, reported by plack, fixed by me) ........ * main/autoservice.c, /: Merged revisions 94790 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r94790 | russell | 2007-12-26 11:06:26 -0600 (Wed, 26 Dec 2007) | 5 lines Don't store DTMF BEGIN frames while a channel is in autoservice. It's just going to make ast_read() do a lot of extra work when the channel comes back out of autoservice. (closes issue #11628, patched by me) ........ * channels/chan_iax2.c: Fix a bug in peer handling that caused multiple instances of a peer to end up in the peers container after a reload. Somehow, this bug doesn't exist in 1.4 ... (closes issue #11626) (reported by pnlarsson, additional info from mvanbaak, fixed by me) * utils: update svn:ignore for astcanary 2007-12-26 15:58 +0000 [r94782] Mark Michelson * configs/extconfig.conf.sample, main/logger.c, CHANGES: Adding support for storing the queue log entries in a realtime backend. (closes issue #11625, reported and patched by sergee) Thank you very much to sergee for adding this new feature! 2007-12-26 10:14 +0000 [r94774] Luigi Rizzo * channels/console_gui.c (added), channels/vcodecs.c (added), channels/console_video.c: Split console_video.c so that video codecs and gui functions are in separate files (still #include'd because of tangling in the data structures, but this is going to be cleaned up). The video grabbing functions still need to be moved to a separate file. 2007-12-25 04:10 +0000 [r94771-94773] Tilghman Lesher * apps/app_pickupchan.c (added): Add pickup by channel (Closes issue #11161) * channels/chan_zap.c, configs/zapata.conf.sample: Change the abbreviated TON from 'A' to 'V', since 'A' is a legitimate DTMF character. Also, fix the documentation to match the code. * res/res_agi.c: Add channel thread ID to the information passed to AGI. Reported by: dror99 Patch by: tilghman (Closes issue #11162) 2007-12-24 19:43 +0000 [r94764-94768] Tilghman Lesher * main/channel.c, /: Merged revisions 94767 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r94767 | tilghman | 2007-12-24 13:36:59 -0600 (Mon, 24 Dec 2007) | 5 lines Race: we need to wait to queue a NewChannel event until after the channel is inserted into the channel list. The reason is because some manager users immediately queue requests from the channel when they see that event and are confused when Asterisk reports no such channel. (Closes issue #11632) ........ * /: Merged revisions 94763 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r94763 | tilghman | 2007-12-24 09:39:56 -0600 (Mon, 24 Dec 2007) | 5 lines Another bit of bad logic in realtime_peer Reported by: dimas Patch by: dimas (Closes issue #11631) ........ 2007-12-23 14:51 +0000 [r94713-94741] Luigi Rizzo * channels/console_video.c, channels/console_video.h: support sdl_videodriver to send output to x11/aalib/console * channels/console_video.c: move reading info from the keypad to a separate function. Remove an unused keypad field and some debugging messages. Adjust formatting on config file parsing * channels/console_video.c: make sure the minimum surface depth is 16bpp so we can create YUVoverlays. With this change we can do setenv SDL_VIDEODRIVER aalib and output to an ascii window (which is still in an X11 window). If you also do unsetenv DISPLAY then the output goes into the main asterisk window, unfortunately it interferes with the normal output so you don't see much. In any case, i don't think we are very far away from having a working xterm videophone! * channels/Makefile: avoid rebuilding dependent files if the generated busy.h and ringtone.h do not change. Ths masks (but does not solve) a but that i am seeing in doing a 'gmake install' without donig a 'gmake all' first. 2007-12-23 01:38 +0000 [r94662] Tilghman Lesher * /, channels/chan_sip.c: Merged revisions 94660 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r94660 | tilghman | 2007-12-22 19:21:03 -0600 (Sat, 22 Dec 2007) | 2 lines Argh... I suppose third time's the charm. ........ 2007-12-22 22:44 +0000 [r94615-94638] Luigi Rizzo * configs/oss.conf.sample, channels/console_video.c: Change the name of config file entries for keypad regions from 'keypad_entry' to 'region'. Fix the example file accordingly. Also make some fixes in the code do reset entries on reload of the keypad. The recently committed kpad2.jpg has the correct names. * images/kpad2.jpg (added): add a sample keypad (with annotations) for console video * channels/console_video.c, channels/Makefile, channels/chan_oss.c, channels/console_video.h (added): Build console_video support by linking in, as opposed to including, console_video.c This will ease the task of splitting console_video.c into its components (V4L and X11 grabbers, various video codecs and packetizers, SDL), as well as ease future extensions (e.g. additional video sources, codecs and rendering engines). For the time being nothing changes for users: video support is off by default, and requires -DHAVE_VIDEO_CONSOLE on the command line to be included (if SDL and FFMPEG are available). 2007-12-21 21:19 +0000 [r94593] Mark Michelson * apps/app_voicemail.c: Something I've been itching to do for a while now. A minor optimization in app_voicemail. Since the dtable in base_encode always gets populated with the same values every time and never changes, make it static and const and only initialize it once. Also, there's no reason to define BASEMAXINLINE twice, so remove the redundant #define. 2007-12-21 20:50 +0000 [r94549-94551] Matthew Fredrickson * channels/chan_zap.c: We should only clear this value if we have to * channels/chan_zap.c: Commit non TCP transport part of #11506. Includes numerous additional parameters, as well as RLT support for DMS type switches 2007-12-21 20:38 +0000 [r94542-94548] Mark Michelson * res/res_config_sqlite.c: Store dates using local time instead of UTC (closes issue #11610, reported and patched by rbraun_performatique) * apps/app_queue.c: Fix a memory leak when reloading queue rules. * CHANGES: The one documentation source I forgot to update after the merge of the queue-penalty branch was the CHANGES file. No longer! * apps/app_voicemail.c: Lots of coding guidelines cleanup. * /, apps/app_voicemail.c: Merged revisions 94540 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r94540 | mmichelson | 2007-12-21 14:11:34 -0600 (Fri, 21 Dec 2007) | 8 lines Better quota support for using IMAP storage voicemail (closes issue #11415, reported by jaroth) (closes issue #11152, reported by selsky) Patch provided by jaroth ........ 2007-12-21 20:12 +0000 [r94541] Jason Parker * codecs/codec_lpc10.c, codecs/codec_g722.c, codecs/codec_adpcm.c, codecs/codec_alaw.c, codecs/codec_speex.c, codecs/codec_g726.c, codecs/codec_gsm.c, codecs/codec_ulaw.c, codecs/codec_zap.c: codecs.conf really shouldn't be mandatory.. it never had been before, so let's go back to being optional. A big "thank you" to pnlarsson on IRC for allowing me access to his system to debug this. Closes issue #11584. 2007-12-21 20:01 +0000 [r94477-94539] Mark Michelson * /, apps/app_voicemail.c: Merged revisions 94538 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r94538 | mmichelson | 2007-12-21 13:59:45 -0600 (Fri, 21 Dec 2007) | 5 lines The mail_copy c-client function does not expect a full imap mailbox string, just the name of the mailbox. (closes issue #11419, reported and patched by jaroth, with additional patchwork from me) ........ * main/dial.c: AST_LIST_REMOVE_CURRENT only takes one argument in trunk * main/dial.c, /: Merged revisions 94468 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r94468 | mmichelson | 2007-12-21 10:49:35 -0600 (Fri, 21 Dec 2007) | 6 lines Since we are freeing list elements within a list traversal, we need to use the safe traversal and remove the item from the list before freeing it. (closes issue 11612, reported by dtyoo) ........ 2007-12-21 16:12 +0000 [r94463-94465] Mark Michelson * /, apps/app_queue.c: Merged revisions 94464 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r94464 | mmichelson | 2007-12-21 10:11:44 -0600 (Fri, 21 Dec 2007) | 3 lines Removing a debug message I accidentally just committed ........ * /, main/say.c, apps/app_queue.c: Merged revisions 94420 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r94420 | mmichelson | 2007-12-21 09:45:14 -0600 (Fri, 21 Dec 2007) | 5 lines Fixing Portuguese syntax for saying dates and times. Also some coding guidelines cleanup. (closes issue #11599, reported and patched by caio1982, coding guidelines cleanup by me) ........ 2007-12-21 15:14 +0000 [r94419] Tilghman Lesher * /, main/asterisk.c: Merged revisions 94418 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r94418 | tilghman | 2007-12-21 09:07:42 -0600 (Fri, 21 Dec 2007) | 2 lines Fix for restart-as-user problem reported via the -dev list ........ 2007-12-21 01:14 +0000 [r94345-94396] Mark Michelson * apps/app_queue.c: Moved the update of the queue_ent's rule list to just before we try to call queue members. This allows for the change in penalty levels to be executed at the most logical time frame. * configs/queues.conf.sample, doc/tex/channelvariables.tex, apps/app_queue.c, configs/queuerules.conf.sample (added): Merging the queue-penalty branch. In short, this allows one to dynamically adjust the QUEUE_MAX_PENALTY and the newly introduced QUEUE_MIN_PENALTY during a call depending on the amount of time passed. The purpose is to allow the call to open up to more (or maybe just different) members without the caller's losing his place in the queue. See configs/queuerules.conf.sample for an example of how to set up queue rules and configs/queues.conf.sample for how to associate a rule with a queue. Along with the functional changes, new CLI and manager commands exist to show the rules defined and there is an additional CLI command to reload the queue rules. Future enhancements that may be made: support for realtime queue rules and support for dynamically adding a rule through the manager or CLI. Also a manager command to reload the queue rules (I'll probably write this myself very soon). * apps/app_voicemail.c: The changes to header inclusion in trunk broke compilation of app_voicemail when using IMAP storage. The reason is that c-client has its own definitions for LOG_WARNING and LOG_DEBUG, so we need to be sure to include asterisk's definitions last so that we use the proper values in app_voicemail. (closes issue #11437, reported by blitzrage, patch suggested by blitzrage) 2007-12-20 22:39 +0000 [r94320] Russell Bryant * configs/zapata.conf.sample: Add a bit more to the description of the "mwimonitor" option. 2007-12-20 22:28 +0000 [r94319] Steve Murphy * build_tools/make_buildopts_h: closes issue #11287; thanks to snuffy for this fix, which will surely make all solaris owners shout praises to his name. 2007-12-20 20:25 +0000 [r94252-94257] Russell Bryant * /, channels/chan_iax2.c: Merged revisions 94256 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r94256 | russell | 2007-12-20 14:22:22 -0600 (Thu, 20 Dec 2007) | 13 lines Merged revisions 94255 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r94255 | russell | 2007-12-20 14:21:41 -0600 (Thu, 20 Dec 2007) | 5 lines Fix another potential seg fault ... (closes issue #11606) Reported by: dimas ........ ................ * channels/chan_zap.c, /: Merged revisions 94251 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r94251 | russell | 2007-12-20 14:08:42 -0600 (Thu, 20 Dec 2007) | 10 lines Fix a deadlock in d-channel handling in chan_zap. This deadlock was introduced by the fix to ensure that channels are properly locked when handling channel variables. There were sections of this code where the channel pvt was locked before the channel lock, when in fact it _must_ be the other way around. (closes issue #11582) Reported by: bugi ........ 2007-12-20 12:56 +0000 [r94168-94191] Luigi Rizzo * channels/chan_usbradio.c, include/asterisk/config.h, channels/console_video.c, channels/chan_oss.c: add some macros to simplify parsing the config file, see description in config.h . They are a variant of the set of macros i used in chan_oss.c, structured in a way to be more robust to the presence of spurious ';' - basically, they define wrappers for 'do {' and '} while (0)', plus some helper functions to deal with simple cases such as ast_copy_string, ast_malloc, strtoul, ast_true ... The prefix (CV_ as 'Config Variable') tries to be easy to remember and has been chosen to not conflict with other existing macros in the tree. For the time being, I have only updated the three source files in the tree that used the old M_* macros. Hopefully, more files will be converted. NOTE: I understand that inventing my own dialect of C is generally wrong; however, the lack of adequate support in the language encourages lazy programming practices (such as ignoring errors, bounds, etc.) and this increases the chance of vulnerability in the code, especially because we are parsing user input here. Hopefully, these macros and the use of ast_parse_arg (in config.h) should encourage the programmer to write more robust code. * include/asterisk/paths.h, res/snmp/agent.c, utils/ael_main.c, utils/extconf.c, main/asterisk.c, utils/conf2ael.c: modify http://svn.digium.com/view/asterisk?view=rev&rev=93603 so that paths and filename are writable by asterisk.c without causing segfaults. This involves defining the variables as const char *, and having them point to as static, writable buffer defined in asterisk.c On passing, fix some errors in using these variables in some files in utils/ , and in res/snmp/agent.c which was redefining a variable without using paths.h (not applicable to 1.4) 2007-12-19 23:17 +0000 [r94123-94124] Mark Michelson * apps/app_queue.c: 1. Unify the check for a penalty < 0 into the set_member_penalty code. 2. Fix an error when checking the CLI command for setting a member's penalty. 3. Fix a logging error if the incorrect parameter was the queue name or interface. (closes issue #11544, reported and patched by Laureano) * /, res/res_monitor.c: Merged revisions 94122 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r94122 | mmichelson | 2007-12-19 17:02:22 -0600 (Wed, 19 Dec 2007) | 6 lines Sox versions 13.0.0 and newer do not have "soxmix" and instead use sox -m. res_monitor needs to use this if the user does not have soxmix. (closes issue #11589, reported by amessina, patch inspired by amessina but with a flourish from me) ........ 2007-12-19 22:51 +0000 [r94085] Russell Bryant * /, configure, include/asterisk/autoconfig.h.in, configure.ac: Merged revisions 94077 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r94077 | russell | 2007-12-19 16:48:48 -0600 (Wed, 19 Dec 2007) | 4 lines Check for the existence of the soxmix application on the target platform and have the result available in autoconfig.h. (part of issue #11589) ........ 2007-12-19 20:20 +0000 [r94052-94053] Tilghman Lesher * apps/app_voicemail.c: Add 'voicemail reload' command. Reported by: eliel Patch by: eliel (Closes issue #11365) * apps/app_waituntil.c (added): Add contributed WaitUntil app. Original code by pprindeville, updated for trunk by tilghman. (Closes issue #11487) 2007-12-19 19:29 +0000 [r94029] Russell Bryant * include/asterisk/time.h: Add a couple of new time API calls - ast_tvdiff_sec and ast_tvdiff_usec (closes issue #11270) Reported by: dimas Patches: tvdiff_us-4.patch uploaded by dimas (license 88) 2007-12-19 17:58 +0000 [r94002] Luigi Rizzo * channels/console_video.c: Add instructions on how to generate your own font. 2007-12-19 17:31 +0000 [r93956] Joshua Colp * /: Merged revisions 93955 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r93955 | file | 2007-12-19 13:29:20 -0400 (Wed, 19 Dec 2007) | 2 lines Make the 1.4 builders happy, ensure var is NULL. ........ 2007-12-19 17:13 +0000 [r93952] Tilghman Lesher * /, channels/chan_iax2.c: Merged revisions 93949 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r93949 | tilghman | 2007-12-19 11:04:13 -0600 (Wed, 19 Dec 2007) | 3 lines Avoid segfault in chan_iax when peer isn't defined (Closes issue #11602) ........ 2007-12-19 17:09 +0000 [r93925-93950] Luigi Rizzo * main/utils.c, include/asterisk/strings.h: Add a new API function, written at least twice in app_voicemail.c and likely in other places too. This is quite useful when placing mail/html stuff in config files. /*! \brief Convert some C escape sequences (\b\f\n\r\t) into the equivalent characters. \brief s The string to be converted (will be modified). \return The converted string. */ char *ast_unescape_c(char *s); * include/asterisk/config.h, main/config.c: add support for PARSE_DOUBLE, and remove identifiers for types not supported (INT16 and UINT16) 2007-12-19 09:20 +0000 [r93899] Olle Johansson * CHANGES: Reorganize CHANGES a bit. The "misc" section grew too large... 2007-12-19 08:57 +0000 [r93898] Luigi Rizzo * configure, include/asterisk/autoconfig.h.in, configure.ac, acinclude.m4, makeopts.in: Properly document AST_EXT_TOOL_CHECK() and use it to check for NETSMP and GTK (GTK is not used thoug). AST_EXT_TOOL_CHECK() could be used for checking curl status as well, perhaps with a small addition because we currently seem to require a curl version greater than X.Y.Z Add a NETSMP_INCLUDE entry in makeopts.in We don't have yet any macros for using pkg-config to check for a specific package (right now there is only gtk2+ in the category). 2007-12-19 08:57 +0000 [r93897] Olle Johansson * channels/chan_sip.c, configs/sip.conf.sample, CHANGES: Adding the ability to specify the To: header in an outbound INVITE by adding an exclamation mark to the dial string. This patch also exists for 1.4 in the fixtoheader-1.4 branch and has been in production for quite some time. 2007-12-19 08:12 +0000 [r93875] Luigi Rizzo * res/snmp/agent.c: make netsmp build under AST_DEVMODE. Description, included in the source, is below. I should note that the PACKAGE_* macros that asterisk defines in autoconfig.h are not used anywhere in the tree so they should just be removed. /* * There is some collision collision between netsmp and asterisk names, * causing build under AST_DEVMODE to fail. * * The following PACKAGE_* macros are one place. * Also netsnmp has an improper check for HAVE_DMALLOC_H, using * #if HAVE_DMALLOC_H instead of #ifdef HAVE_DMALLOC_H * As a countermeasure we define it to 0, however this will fail * when the proper check is implemented. */ No 2007-12-19 07:01 +0000 [r93854] Olle Johansson * CHANGES, main/asterisk.c, doc/asterisk.sgml: Add option for starting remote Asterisk by naming the actual runtime socket instead of pointing to configuration file with -C Reported by: sobomax Patches: asterisk.c.diff.trunk uploaded by sobomax (license 359) doc changes by committer (closes issue #11598) 2007-12-19 00:09 +0000 [r93827] Dwayne M. Hubbard * apps/app_osplookup.c: add missing header file 2007-12-18 23:38 +0000 [r93804-93805] Tilghman Lesher * main/asterisk.c: Making the canary error message a little more obvious. * utils/Makefile, utils/astcanary.c (added), main/asterisk.c: Add a canary process, for high priority mode (asterisk -p) to ensure that if Asterisk goes into a busy loop, the machine will be recoverable. We'd still need to do a restart to put Asterisk back into high priority mode, but at least a reboot won't be required. (Closes issue #11559) 2007-12-18 21:13 +0000 [r93741] Olle Johansson * channels/chan_sip.c: Move some warnings away to debug since some devices send a packet with a silly string as a NAT keepalive packet. 2007-12-18 18:39 +0000 [r93672] Tilghman Lesher * /, channels/chan_sip.c, channels/chan_iax2.c: Merged revisions 93668 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r93668 | tilghman | 2007-12-18 12:29:39 -0600 (Tue, 18 Dec 2007) | 10 lines Merged revisions 93667 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r93667 | tilghman | 2007-12-18 12:23:06 -0600 (Tue, 18 Dec 2007) | 2 lines Fixing AST-2007-027 (Closes issue #11119) ........ ................ 2007-12-18 18:20 +0000 [r93666] Luigi Rizzo * include/asterisk/paths.h: remove a leftover line with only a '#' (wonder why the compiler does not complain!) and variables that are only used in asterisk.c 2007-12-18 17:05 +0000 [r93626] Mark Michelson * main/channel.c, /: Merged revisions 93625 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r93625 | mmichelson | 2007-12-18 11:02:48 -0600 (Tue, 18 Dec 2007) | 6 lines Rework deadlock avoidance used in ast_write, since it meant that agent channels which were being monitored had one audio file recorded and one empty audio file saved. (closes issue #11529, reported by atis patched by me) ........ 2007-12-18 10:24 +0000 [r93558-93603] Luigi Rizzo * include/asterisk/paths.h, channels/chan_sip.c, res/res_crypto.c, utils/ael_main.c, utils/extconf.c, main/asterisk.c, res/res_monitor.c, utils/conf2ael.c: make configuration variable const so they are not accidentally modified. This requires casting the strings in asterisk.c when writing to them, so we do it through a macro to do it consistently. * channels/chan_unistim.c, res/res_crypto.c, main/astmm.c, apps/app_ices.c, utils/extconf.c, channels/chan_iax2.c, main/asterisk.c, main/config.c, main/db.c, apps/app_adsiprog.c, cdr/cdr_csv.c: remove unnecessary (char *) casts for ast_config_AST_* variables. There are some left in the .flex files, left to the maintainer... * build_tools/make_defaults_h, main/asterisk.c: Rename the macros in defaults.h - they are not meant to be globally visible. Document the fact that DEFAULT_TMP_DIR cannot be overridden from the default configuration (this needs to be fixed, as you could have a totally different spooldir configured at runtime, and yet DEFAULT_TMP_DIR keeps the compile-time default). Remove two unused entries for sounds and images. * Makefile.moddir_rules: make the code match documentation - now you can specify multiple words in MODULE_PREFIX. * CREDITS: Name the people responsible for some recent contributions to the tree. * Makefile: Two small changes: + document the difference between "A=foo make ..." and "make A=foo ..." and suggest using COPTS/LDOPTS if you need to use the second form to pass compiler and loader flags; + define only in one place the environment used to build stuff in menuselect/ 2007-12-18 07:56 +0000 [r93557] Olle Johansson * doc/CODING-GUIDELINES: A minor update, caused by a recent bug report ;-) 2007-12-18 07:22 +0000 [r93536] Luigi Rizzo * doc/CODING-GUIDELINES: small documentation update (nothing important). 2007-12-18 02:57 +0000 [r93514] Joshua Colp * channels/chan_unistim.c: You... will... build! I say so and therefore you will. 2007-12-18 02:42 +0000 [r93493] Kevin P. Fleming * channels/chan_unistim.c, include/asterisk/threadstorage.h: minor cleanups 2007-12-17 23:10 +0000 [r93464] Luigi Rizzo * channels/chan_unistim.c: fix building under cygwin. At this point WINARCH should go away. 2007-12-17 22:54 +0000 [r93405] Luigi Rizzo * channels/chan_unistim.c: remove some unnecessary includes 2007-12-17 22:50 +0000 [r93390] Jason Parker * /, main/translate.c: Merged revisions 93381 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r93381 | qwell | 2007-12-17 16:45:57 -0600 (Mon, 17 Dec 2007) | 4 lines What was I thinking when I wrote this masterpiece? -1 + 1 = 0.. who woulda thunk it?. ........ 2007-12-17 22:38 +0000 [r93380] Luigi Rizzo * channels/chan_oss.c: surprising as it may be, chan_oss compiles correctly under cygwin as well, provided you look for soundcard.h in the right place... 2007-12-17 22:29 +0000 [r93378] Joshua Colp * /, main/utils.c: Merged revisions 93377 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r93377 | file | 2007-12-17 18:28:09 -0400 (Mon, 17 Dec 2007) | 7 lines Do not try to access information about a lock when printing out a trylock attempt. It is possible for the lock that it references to no longer be valid. This would have caused segfaults or deadlocks. (issue #BE-263) (closes issue #11080) Reported by: callguy (closes issue #11100) Reported by: callguy ........ 2007-12-17 21:14 +0000 [r93337] Tilghman Lesher * /, include/asterisk/time.h: Merged revisions 93336 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r93336 | tilghman | 2007-12-17 15:12:42 -0600 (Mon, 17 Dec 2007) | 6 lines Today is tomorrow's yesterday, and yesterday's tomorrow is today, and tomorrow's tomorrow is the day after tomorrow, so who cares if you recycle anyway? If this confuses you, that's nothing compared to what this fixes. ;-) ........ 2007-12-17 21:12 +0000 [r93335] Olle Johansson * channels/chan_zap.c, /, channels/chan_sip.c, apps/app_queue.c, channels/chan_iax2.c, channels/chan_mgcp.c: Merged revisions 93182 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r93182 | oej | 2007-12-17 08:15:13 +0100 (MÃ¥n, 17 Dec 2007) | 8 lines Issue 11574: Add dependencies on res_monitor and res_features. I wonder if Asterisk can run at all without res_features. My guess is that there's propably a lot of more modules and the core that depends on it. Reported by: caio1982 (closes issue #11574) ........ 2007-12-17 20:42 +0000 [r93293-93297] Mark Michelson * apps/app_queue.c: Removing some leftover debug messages from a while back. * /, apps/app_voicemail.c: Merged revisions 93291 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r93291 | mmichelson | 2007-12-17 13:53:48 -0600 (Mon, 17 Dec 2007) | 6 lines We need to create the directory for a voicemail user even if they are using IMAP storage since greetings are stored in the filesystem. (closes issue #11388, reported by spditner, patch by me inspired by a patch by spditner) ........ 2007-12-17 18:07 +0000 [r93252] Joshua Colp * channels/chan_zap.c, /: Merged revisions 93250 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r93250 | file | 2007-12-17 14:05:55 -0400 (Mon, 17 Dec 2007) | 6 lines If a call is received with a called number IE containing nothing go to the 's' extension. (closes issue #9099) Reported by: kb1_kanobe2 Patches: 20070906__9099.diff.txt uploaded by Corydon76 (license 14) ........ 2007-12-17 17:16 +0000 [r93191-93224] Kevin P. Fleming * utils: all created files need to be listed in the ignore property * channels/chan_unistim.c, build_tools/menuselect-deps.in, configure, configure.ac, channels/Makefile, channels/chan_oss.c: make the configure script detect that it is running on a Windows platform, and report that information so that menuselect can use it (all information that is used to decide whether to build modules or not must be fed to menuselect so the user knows what will be built and why... don't make module build decisions in the makefiles, please) * Makefile: make using PRINT_DIR a little easier 2007-12-17 15:18 +0000 [r93187-93190] Joshua Colp * channels/chan_sip.c: Fix usage of rtptimeout. It can be used without rtpkeepalive, and the value can not be accessed directly in the SIP pvt structure. All RTP related timeouts have to be retrieved using the ast_rtp_* function calls. (closes issue #11562) Reported by: ibc * channels/chan_unistim.c: If no timezone is available use the default message. (closes issue #11576) Reported by: junky * channels/chan_unistim.c: Make chan_unistim actually be able to unload. When creating a thread that you want to pthread_join you have to explicitly create it as joinable, and also if using pthread_cancel you have to have a pthread_testcancel to see if it has been called. 2007-12-17 07:27 +0000 [r93184-93185] Kevin P. Fleming * codecs, /, build_tools/make_version, include/asterisk/autoconfig.h.in, configure.ac, apps, Makefile.moddir_rules, res/Makefile, pbx/Makefile, build_tools/prep_moduledeps (removed), channels/Makefile, cdr, formats, Makefile, codecs/Makefile, funcs, apps/Makefile, configure, build_tools/embed_modules.xml, cdr/Makefile, build_tools/prep_tarball, makeopts.in, formats/Makefile, res, pbx, channels, funcs/Makefile: Merged revisions 93180 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r93180 | kpfleming | 2007-12-16 22:44:51 -0800 (Sun, 16 Dec 2007) | 23 lines In http://lists.digium.com/pipermail/asterisk-dev/2007-December/031145.html, rizzo brought up some issues related to the way that the metadata required for menuselect and the rest of the build system is extracted from the source files. Since I had a few hours to kill on an airplane today, I decided to improve this situation... so now the system caches the extracted metadata and uses it to build the menuselect 'tree' as much as it can. The result of this is that when a single source file is changed, only the metadata for that file needs to be extracted again, and the rest is used from the cache files. I also reduced the number of forked processes required to do the metadata extraction; it was actually possible to do most of what we needed in the Makefiles themselves without using any shell scripts at all! On my laptop, these changes resulted in an 80% decrease in the time required for the 'menuselect.makeopts' automatic check to occur after editing a single source file. While doing this work I also cleaned up a few minor things in the Makefiles, adding a check for 'awk' to the configure script and changed all remaining places we use 'grep' or 'awk' to use the ones found by the configure script, and changed the 'prep_tarball' script to build the menuselect metadata so that tarballs of Asterisk will include it and won't require the user to wait while it is extracted after unpacking. ........ 2007-12-16 19:06 +0000 [r93173] Luigi Rizzo * Makefile: menuselect.makeopts is not a .PHONY target 2007-12-16 13:38 +0000 [r93163-93167] Olle Johansson * pbx/pbx_dundi.c: Convert from LOG_DEBUG etc to ast_debug. Thanks, dimas! (closes issue #11572) Reported by: dimas Patches: dundilog-trunk.patch uploaded by dimas (license 88) * main/manager.c, CHANGES: Adding a new CLI command for "manager reload", which is important now that you need to reload after changes. Thanks YS. Reported by: ys Patches: trunk93163_manager_reload.c.diff uploaded by ys (license 281) (related to issue #11414) * main/manager.c, CHANGES: Change manager so that registered accounts are stored in memory. This opens for a manager realtime implementation. If you change accounts in manager.conf, you now need to reload to activate the changes (deletions, additions). This was not the case with 1.4. Reported by: ys Patches: trunk93163_manager_reload.c.diff uploaded by ys (license 281) (closes issue #11414) * CHANGES: Adding console_video to CHANGES. It's important that we keep this file up to date, even with experimental stuff. * channels/chan_unistim.c, main/udptl.c, configs/dundi.conf.sample, channels/chan_sip.c, include/asterisk/rtp.h, include/asterisk/netsock.h, channels/iax2-provision.c, UPGRADE.txt, doc/tex/qos.tex, configs/skinny.conf.sample, CHANGES, channels/chan_iax2.c, main/rtp.c, main/netsock.c, configs/h323.conf.sample, configs/iax.conf.sample, channels/chan_skinny.c, configs/mgcp.conf.sample, configs/unistim.conf.sample, channels/chan_h323.c, configs/iaxprov.conf.sample, pbx/pbx_dundi.c, configs/sip.conf.sample, channels/chan_mgcp.c: HUGE improvements to QoS/CoS handling by IgorG - Refer to the proper documentation - Implement separate signalling/media QoS/CoS in many channels using RTP - Improve warnings and verbose messages - Deprecate some old settings Minor modifications by me, a big effort from IgorG. Thanks! Reported by: IgorG Patches: qoscleanup-89394-4-trunk.patch uploaded by IgorG (license 20) Tested by: IgorG (closes issue #11145) 2007-12-16 10:34 +0000 [r93162] Luigi Rizzo * Makefile: use a simpler idiom for 'cmp -s ...' 2007-12-16 09:37 +0000 [r93152-93161] Olle Johansson * main/asterisk.c: Don't drop the first character randomly in long listings in the CLI. Reported by: slavon Patches: asterisk-consolerefresh2.diff.txt uploaded by jamesgolovich (license 176) Tested by: eliel (closes issue #9325) * configs/sip.conf.sample, CHANGES: Update documentation * channels/chan_sip.c, configs/sip.conf.sample: Make more timers settable in SIP so that we can force timeout earlier on non-responsive SIP servers. Thanks, jcmoore, for the patch! Reported by: jcmoore Patches: peer_t1_timerb_trunk_v3.patch.txt uploaded by jcmoore (license 9) (closes issue #9771) * include/asterisk/file.h: Typo fixed earlier, that wasn't a typo after all. Didn't a clever guy once say "Compile before you commit" ? :-) 2007-12-15 08:10 +0000 [r93151] Russell Bryant * include/asterisk/file.h: fix a typo from revision 93138 2007-12-15 00:44 +0000 [r93138-93145] Luigi Rizzo * configs/oss.conf.sample: configuration options related to video support. * channels/console_video.c (added): Bring in video console support for chan_oss (and later chan_alsa too). This is disabled in the default build, you need to explicitly enable it compiling with make COPTS=-DHAVE_VIDEO_CONSOLE In return, you will be able to do a video call with chan_oss, using the webcam (or X11 grabbing) as local source, and rendering the incoming stream on your screen. Currently supported formats are h261, h263, h263+, h264, mpeg4 (all through the avcodec lib, part of ffmpeg). Incoming video is on the left, outgoing video is on the right, while the center displays a keypad (if configured so). Right clicking on the video windows increases the size, center clicking reduces the size. Dragging the mouse (with the left key) on the right window while the X11 grabber is active moves the grab area. This is the result of work by Sergio Fadda, Marta Carbone and myself, all properly disclaimed to digium. Note, there is a lot of work left to do in this module, including adding support for Video4LinuxV2 (I have patches from Matteo Brancaleoni which should be integrated), and making the GUI a lot more friendly than it is now (e.g. supporting merging or switching among multiple sources, a text window, and more). * channels/chan_oss.c: remove some redundant headers * include/asterisk/file.h: include mmap header if detected by configure 2007-12-14 22:02 +0000 [r93094-93115] Mark Michelson * apps/app_voicemail.c: Resolve a compiler warning * apps/app_voicemail.c: Change places where the name "INBOX" was hardcoded to use the imapfolder setting from voicemail.conf instead. This commit will help to get issue #11415 moving towards commitment. 2007-12-14 21:09 +0000 [r93090] Tilghman Lesher * Makefile, channels/chan_unistim.c, codecs/ilbc/iLBC_define.h: Solaris compat fixes Reported by: snuffy Patch by: snuffy,tilghman (Closes issue #11315) 2007-12-14 19:31 +0000 [r93067] Russell Bryant * pbx/pbx_dundi.c: make something static 2007-12-14 19:27 +0000 [r93066] Tilghman Lesher * apps/app_privacy.c, UPGRADE.txt, CHANGES, configs/privacy.conf.sample (removed): Remove use of privacy.conf by the Privacy app. Reported by: eliel Patch by: eliel (Closes issue #11344) 2007-12-14 19:19 +0000 [r93042-93065] Mark Michelson * main/pbx.c, main/manager.c, funcs/func_timeout.c: I needed to increment the numbers used on the VERBOSITY_ATLEAST calls by 1. Thanks to kpfleming for pointing this out. * include/asterisk/logger.h, main/pbx.c, main/manager.c, funcs/func_timeout.c: Changed VERBOSITY_LEVEL to VERBOSITY_ATLEAST to be more accurate. * include/asterisk/logger.h, main/pbx.c, main/manager.c, funcs/func_timeout.c, main/logger.c: After reading Russell's e-mail to the dev list stating that checking option_verbose is not equivalent to the check done by ast_verb, I wrote a macro, VERBOSITY_LEVEL, which does this check. I did a quick look in the source and used this macro in some places where option_verbose was used. I also converted some verbose messages in logger.c to use ast_verb instead of ast_verbose. 2007-12-14 18:24 +0000 [r93041] Tilghman Lesher * apps/app_meetme.c: gcc 4.1.3 wants a union used here. 2007-12-14 17:49 +0000 [r93001-93004] Russell Bryant * main/config.c: Print an error message if a #included file does not exist 2007-12-14 17:29 +0000 [r92999] Tilghman Lesher * res/res_agi.c: Publish the AGI events to manager. Reported by: moy Patch by: moy,tilghman (Closes issue #11337) 2007-12-14 15:59 +0000 [r92976] Mark Michelson * funcs/func_timeout.c: Reintroduce an optimization that was lost when converting trunk to use ast_verb. 2007-12-14 15:49 +0000 [r92939] Tilghman Lesher * main/editline/sys.h: If malloc.h is included in a Solaris build, the compilation breaks. Reported by: snuffy Patch by: snuffy (Closes issue #11313) 2007-12-14 15:18 +0000 [r92938] Joshua Colp * /, channels/chan_sip.c: Merged revisions 92937 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r92937 | file | 2007-12-14 11:16:15 -0400 (Fri, 14 Dec 2007) | 4 lines Up the length of the format on the SIP channel since it can now be rather long. (closes issue #11552) Reported by: francesco_r ........ 2007-12-14 15:14 +0000 [r92936] Tilghman Lesher * /, res/res_agi.c: Merged revisions 92933 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r92933 | tilghman | 2007-12-14 09:01:10 -0600 (Fri, 14 Dec 2007) | 5 lines Change help documentation to match actual behavior (FAILURE vs FAILED). Reported by: angeloxx-sir Patch by: tilghman (Closes issue #11548) ........ 2007-12-14 15:08 +0000 [r92935] Christian Richter * channels/chan_misdn.c, /: Merged revisions 92934 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r92934 | crichter | 2007-12-14 16:05:28 +0100 (Fr, 14 Dez 2007) | 1 line fixed the sequencing of WAITING_4DIGS state setting and overlap_task thread starting. ........ 2007-12-14 14:48 +0000 [r92913] Tilghman Lesher * apps/app_dial.c, main/pbx.c, main/srv.c, channels/chan_skinny.c, res/res_features.c, apps/app_minivm.c, apps/app_amd.c, res/snmp/agent.c, apps/app_chanspy.c, apps/app_mixmonitor.c, main/asterisk.c, main/netsock.c, apps/app_voicemail.c: Convert ast_verbose to ast_verb. Reported by: snuffy Patch by: snuffy (Closes issue #11547) 2007-12-14 01:25 +0000 [r92876] Mark Michelson * /, include/asterisk/lock.h: Merged revisions 92875 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r92875 | mmichelson | 2007-12-13 19:24:06 -0600 (Thu, 13 Dec 2007) | 7 lines When compiling with DETECT_DEADLOCKS, don't spam the CLI with messages about possible deadlocks. Instead just print the intended single message every five seconds. (closes issue 11537, reported and patched by dimas) ........ 2007-12-13 23:10 +0000 [r92816-92855] Tilghman Lesher * apps/app_meetme.c: When working with dates, use numeric form whenever possible, as it's faster. Also, a bunch of coding guidelines fixes. * channels/chan_zap.c, /: Merged revisions 92815 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r92815 | tilghman | 2007-12-13 15:28:39 -0600 (Thu, 13 Dec 2007) | 5 lines Properly initialize polarity statuses, so that they are detected properly. Reported by: julianjm Patch by: julianjm (Closes issue #10238) ........ 2007-12-13 20:23 +0000 [r92811] Joshua Colp * include/asterisk/app.h, include/asterisk/module.h, res/res_agi.c, apps/app_rpt.c: Move usage of the old LOCAL_USER_* macros to the new ast_module_user_* functions in a few documentation places. (closes issue #11533) Reported by: IgorG Patches: oldmacroclean.v1.diff uploaded by IgorG (license 20) 2007-12-13 20:14 +0000 [r92810] Jason Parker * main/pbx.c, /: Merged revisions 92809 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r92809 | qwell | 2007-12-13 14:13:48 -0600 (Thu, 13 Dec 2007) | 1 line Make application help text a little more clear about the use of extensions in a filename. ........ 2007-12-13 20:12 +0000 [r92806-92808] Mark Michelson * /, apps/app_voicemail.c: Merged revisions 92807 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r92807 | mmichelson | 2007-12-13 14:03:20 -0600 (Thu, 13 Dec 2007) | 3 lines Prevent another potential fd leak ........ * /, apps/app_voicemail.c: Merged revisions 92803 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r92803 | mmichelson | 2007-12-13 13:49:55 -0600 (Thu, 13 Dec 2007) | 3 lines Prevent a possible fd leak. ........ 2007-12-13 17:46 +0000 [r92779] Tilghman Lesher * cdr/cdr_adaptive_odbc.c: Don't use backslash as an escape character, unless it really is an escape character. 2007-12-13 16:23 +0000 [r92758] Jason Parker * channels/chan_sip.c: Remove remnants of a poorly merged commit. (92697) 2007-12-13 15:40 +0000 [r92737] Doug Bailey * apps/app_voicemail.c: Tag voicemails with UTC time as opposed to local time zone 2007-12-13 00:18 +0000 [r92697] Jason Parker * /, channels/chan_sip.c, channels/chan_h323.c, main/config.c: Merged revisions 92696 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 (closes issue #10690) ........ r92696 | qwell | 2007-12-12 18:11:09 -0600 (Wed, 12 Dec 2007) | 7 lines If a typo is found in a config file, we previous continued on with what was already loaded. We do not want to do this (see bug below for details). This makes it so that if a [ is found without a ], the entire config will fail, and nothing in it will be loaded. Issue 10690. ........ 2007-12-12 23:44 +0000 [r92676] Russell Bryant * channels/chan_iax2.c: Revert an "optimization" that I added in revision 89887, as the user who reported issue #11449 has demonstrated that it actually was a performance hit on his machine. I think that it is possible that it could still be a benefit on systems under higher load, especially SMP systems, but I don't have enough time or interest to find out at the moment. (closes issue #11449) 2007-12-12 21:22 +0000 [r92618] Jason Parker * /, apps/app_meetme.c, channels/ringtone.h: Merged revisions 92617 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 (closes issue #11048) ........ r92617 | qwell | 2007-12-12 15:15:45 -0600 (Wed, 12 Dec 2007) | 4 lines Don't increment user count until after name has been recorded (if enabled). Issue 11048, tested by pep. ........ 2007-12-12 20:05 +0000 [r92594] Tilghman Lesher * apps/app_dial.c, main/logger.c, main/utils.c, apps/app_mixmonitor.c: Conversions of free to ast_free, where applicable, and several other formatting fixes. Reported by: eliel Patch by: eliel,tilghman (Closes issue #11209) 2007-12-12 19:50 +0000 [r92562] Russell Bryant * res/res_features.c: Merged revisions 92556 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r92556 | russell | 2007-12-12 13:40:02 -0600 (Wed, 12 Dec 2007) | 1 line resolve compiler warning ........ 2007-12-12 17:51 +0000 [r92511-92526] Mark Michelson * res/res_features.c: Same change to trunk as revision 92510. I'm not sure why I merged this way, but I did. 2007-12-12 17:15 +0000 [r92476-92507] Tilghman Lesher * main/asterisk.c: Correctly handle possible memory allocation failure Reported by: eliel Patch by: eliel (Closes issue #11512) * /, configure, include/asterisk/autoconfig.h.in, configure.ac: Merged revisions 92463 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r92463 | tilghman | 2007-12-12 10:52:56 -0600 (Wed, 12 Dec 2007) | 4 lines Test directly for the API that fixed AST-2007-026, to ensure that older versions of PostgreSQL are no longer acceptable. (Closes issue #11526) ........ 2007-12-12 16:11 +0000 [r92444] Mark Michelson * /, apps/app_queue.c: Merged revisions 92443 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r92443 | mmichelson | 2007-12-12 10:08:55 -0600 (Wed, 12 Dec 2007) | 3 lines Removing an unused variable. ........ 2007-12-11 22:20 +0000 [r92423] Olle Johansson * include/asterisk/term.h, channels/misdn/isdn_msg_parser.c, channels/ringtone.h, include/asterisk/ulaw.h, include/jitterbuf.h, include/asterisk/manager.h, include/asterisk/transcap.h, channels/misdn/isdn_lib.c, channels/gentone.c, include/asterisk/zapata.h, channels/misdn/isdn_lib.h, include/asterisk/doxyref.h, channels/DialTone.h, channels/misdn/ie.c, channels/misdn/chan_misdn_config.h, channels/iax2.h, channels/misdn/portinfo.c, include/asterisk/udptl.h, main/cygload.c, include/asterisk/translate.h: Doxygen updates, formatting. misdn stuff needs a lot of doxygenification (Hello, Qwell :-) ) 2007-12-11 22:10 +0000 [r92422] Mark Michelson * channels/chan_zap.c, configure, include/asterisk/autoconfig.h.in, configure.ac: Trunk build would fail due to the nonexistence of zaptel hwgain structures missing. Patched configure to check for this stuff and put a #ifdef around the offending code in chan_zap. Thanks to file for overseeing this. 2007-12-11 21:58 +0000 [r92421] Jason Parker * channels/chan_sip.c: We need to set the address we want to match against before we actually do the match.. Closes issue #11518. 2007-12-11 21:46 +0000 [r92402] Mark Michelson * res/res_musiconhold.c: Removing a pointless memset. The memory was just calloc'd, so the memory is already zeroed out 2007-12-11 21:17 +0000 [r92401] Jason Parker * apps/app_controlplayback.c: Add variable to show which key was pressed to stop playback. Issue #11377, initial patch by johan. 2007-12-11 20:06 +0000 [r92364-92365] Joshua Colp * res/res_monitor.c: Only look to see if options are set if some have been provided. (closes issue #11505) Reported by: Mike Anikienko * main/global_datastores.c, /: Merged revisions 92363 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r92363 | file | 2007-12-11 15:51:40 -0400 (Tue, 11 Dec 2007) | 6 lines Fix potential memory leak with the dialed interfaces list if another memory allocation fails. (closes issue #11507) Reported by: eliel Patches: global_datastores.c.patch uploaded by eliel (license 64) ........ 2007-12-11 17:44 +0000 [r92324] Mark Michelson * /, apps/app_queue.c: Merged revisions 92323 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r92323 | mmichelson | 2007-12-11 11:42:25 -0600 (Tue, 11 Dec 2007) | 10 lines Fixing autofill to be more accurate. Specifically, if calls ahead of the current caller were ringing members (but not yet bridged) there could be available members and waiting callers who would not get matched up. The member availability checker was correctly determining the number of available members in this scenario, but the queue itself did not parallelly reflect this status on the pending calls. This commit corrects the issue. (closes issue #11459, reported by equissoftware, patched by me) ........ 2007-12-11 16:29 +0000 [r92305] Russell Bryant * include/asterisk/unaligned.h, main/event.c: * In unaligned.h, remove some unnecessary casts and mark the arg of the get_unaligned functions as const * In event.c, use get_unaligned_uint32() in a couple of places to fix issues on architectures that don't allow unaligned access 2007-12-11 14:17 +0000 [r92267-92285] Olle Johansson * include/asterisk/devicestate.h, include/asterisk/agi.h, include/asterisk/astobj2.h, include/asterisk/extconf.h, include/asterisk/io.h, include/asterisk/cdr.h, include/asterisk/aes.h, include/asterisk/_private.h, include/asterisk/localtime.h, include/asterisk/hashtab.h, include/asterisk/callerid.h, include/asterisk/logger.h, include/asterisk/doxyref.h, include/asterisk/app.h, include/asterisk/adsi.h, include/asterisk/event.h, include/asterisk/causes.h, include/asterisk/alaw.h, include/asterisk/ast_expr.h, include/asterisk/dsp.h, include/asterisk/mod_format.h, include/asterisk/ael_structs.h, include/asterisk/astdb.h: A lot of doxygen updates * include/asterisk/frame.h: Doxygen updates 2007-12-10 20:18 +0000 [r92243] Doug Bailey * channels/chan_zap.c: Add CLI commands to dynamically set hw and sw gains 2007-12-10 16:48 +0000 [r92205-92206] Joshua Colp * utils/check_expr.c: Add ast_atomic_fetchadd_int_slow to check_expr for platforms that need it. (closes issue #11484) Reported by: snuffy * /, main/rtp.c: Merged revisions 92204 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r92204 | file | 2007-12-10 12:36:15 -0400 (Mon, 10 Dec 2007) | 6 lines Add G729A as another possible payload name for G729. Some devices use this instead of G729, which is perfectly normal since the payload number itself is defined and can't be used by anything else so the name doesn't matter that much. (closes issue #11483) Reported by: revolution Patches: rtp.diff uploaded by revolution (license 346) ........ 2007-12-10 16:30 +0000 [r92203] Mark Michelson * /, apps/app_queue.c: Merged revisions 92202 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r92202 | mmichelson | 2007-12-10 10:29:44 -0600 (Mon, 10 Dec 2007) | 7 lines If there are no members in a queue, then the loop where the datastore for detecting duplicate dialed numbers will be skipped, meaning the datastore isn't created. This means that when we try to free it, there's a crash. This stops that crash from occurring. (closes issue #11499, reported by slavon, patched by eliel) ........ 2007-12-10 16:15 +0000 [r92199-92201] Joshua Colp * res/res_agi.c: Only send a SIGHUP if the pid is greater than -1, otherwise all PIDs greater than -1 will get the SIGHUP... and that is bad. (closes issue #11453) Reported by: alanmcmillan 2007-12-10 14:18 +0000 [r92140-92160] Olle Johansson * channels/chan_sip.c: Removing some LOG_DEBUG items * /, channels/chan_sip.c: Merged revisions 92158 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r92158 | oej | 2007-12-10 15:04:44 +0100 (MÃ¥n, 10 Dec 2007) | 16 lines Avoid reinvite race situations with two Asterisks trying to reinvite each other in 1.4 and trunk. This patch implements support for the 491 error code that Asterisk 1.4 generates on situations where we get an incoming INVITE and already has one in progress. Thanks to mavetju for reporting and to Raj Jain for an excellent explanation of the problem. Patch by myself. Tested with 8 Asterisk servers connected to each other in a training network. Closes issue #10481 ........ * doc/manager_1_1.txt, apps/app_voicemail.c: Add a few extra headers in the voicemail users listing in manager 1.1. Update documentation too. (closes issue #11495) Reported by: caio1982 Patches: extra_vm_manager_info1.diff uploaded by caio1982 (license 22) 2007-12-10 09:00 +0000 [r91929-92122] Luigi Rizzo * build_tools/make_version, build_tools/make_version_h: simplify/cleanup the scripts * utils/Makefile: remove relative paths and use ASTTOPDIR instead. Give a default value to ASTTOPDIR if unset so we can at least do a 'make clean' without too much trouble. The proper fix, however, is to partition the top level Makefile in a 'setup' and a 'main' part, in a way that the 'setup' part can be included from subdirs' Makefiles and allow targets to be built without going through the top level Makefile. * utils/clicompat.c: simplify this file * doc/CODING-GUIDELINES: add a bit of info on the build infrastructure * Makefile: Fix the detection of modules installed from this build. You can now add the path of local module subdirs from the command line with make LOCAL_MOD_SUBDIRS= .... * codecs/Makefile, apps/Makefile, Makefile.moddir_rules, cdr/Makefile, pbx/Makefile, res/Makefile, channels/Makefile, formats/Makefile, funcs/Makefile: Put into Makefile.moddir_rules the common instructions used to generate loadable and embedded module lists. Individual Makefiles now are a lot simpler, possibly as simple as this: -include $(ASTTOPDIR)/menuselect.makeopts $(ASTTOPDIR)/menuselect.makedeps MODULE_PREFIX=cdr_ all: _all include $(ASTTOPDIR)/Makefile.moddir_rules and also more flexible because in a single directory we can combine various types of modules (app_, cdr_, func_, ... ) by simply listing them in the MODULE_PREFIX variable. The individual Makefiles can also create list of modules to be excluded by listing them in the variablel MODULE_EXCLUDE (see an example in channels/Makefile). With this change it becomes trivial to integrate a directory with locally created/modified sources into the main build. * Makefile, Makefile.moddir_rules: make the install target a bit less noisy * Makefile: document usage of several exported variables * utils/Makefile: add hashtab.c to the list of files deleted * Makefile.moddir_rules: another place where ../ should have been ASTTOPDIR * codecs/Makefile, utils/Makefile, apps/Makefile, cdr/Makefile, pbx/Makefile, res/Makefile, channels/Makefile, formats/Makefile, funcs/Makefile: normalize subdirs' Makefile by using ASTTOPDIR and not .. to reference the top level directory. * Makefile: Implement the outcome of a discussion on the -dev list re. the use of DESTDIR and INSTALL_PATH - many thanks to Tzafrir Cohen and Simon Perreault for extremely useful feedback: 1. comment out the [directories] section the created asterisk.conf ; you can set the correct defaults at build time using INSTALL_PATH, so the repetition here is redundant and often wrong. (The next step now is move asterisk.conf outside the Makefile to asterisk.conf.sample, because there is little if anything here that needs to be constructed at build/install time). 2. use DESTDIR?=$(INSTALL_PATH) so you only need to specify a path once if the two coincide. This should have no ill side effects, because if you don't specify DESTDIR, you really need INSTALL_PATH="" to set the correct defaults, and if you specify DESTDIR the value is not overridden. The second part required moving the 'export DESTDIR' right after the assignment to prevent DESTDIR getting set by the export (this is documented in the Makefile).o hopefully avoid the mistake)$ With this change you can now do something like this from your source tree: make INSTALL_PATH=/some/place install samples and then main/asterisk -vdc which will pick up the correct config files and libraries from /some/place - i.e. great for developers! * main/config.c: remove unused code, and simplify the logic for #include/#exec (still a lot of cleanup needed here). * main/config.c: Implement comment_buffer and lline_buffer in terms of the ast_str_*() API. I don't know if comment_buffers etc are actually used at all... * main/config.c: unify some common code * main/config.c: normalize formatting * main/config.c: document a nice technique to exit from a block in case of errors. * main/config.c: a little bit of documentation on how lines are parsed. * utils/ael_main.c: normalize header order, and add a comment on the need to clean up this file. * include/asterisk/network.h: some platforms (e.g. FreeBSD4) need netinet/in.h to be included before arpa/inet.h 2007-12-07 23:32 +0000 [r91832-91891] Jason Parker * /, main/dsp.c: Merged revisions 91890 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 (closes issue #11273) ........ r91890 | qwell | 2007-12-07 17:29:01 -0600 (Fri, 07 Dec 2007) | 4 lines We need to make sure we free the input frame if we return a different frame in ast_dsp_process. Issue 11273, pointed out by dimas, with a patch by eliel. ........ * pbx/pbx_lua.c, configs/extensions.lua.sample: Update documentation for pbx_lua. Closes issue #11492, patch by mnicholson. 2007-12-07 21:25 +0000 [r91784-91831] Russell Bryant * /, main/utils.c: Merged revisions 91830 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r91830 | russell | 2007-12-07 15:24:33 -0600 (Fri, 07 Dec 2007) | 5 lines Make the lock protecting each thread's list of locks it currently holds recursive. I think that this will fix the situation where some people have said that "core show locks" locks up the CLI. (related to issue #11080) ........ * /, include/asterisk/lock.h: Merged revisions 91828 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r91828 | russell | 2007-12-07 15:17:24 -0600 (Fri, 07 Dec 2007) | 6 lines Fix another bug in the DEBUG_THREADS code. The ast_mutex_init() function had the mutex attribute object marked as static. This means that multiple threads initializing locks at the same time could step on each other and end up with improperly initialized locks. (found when tracking down locking issues related to issue #11080) ........ * /, include/asterisk/lock.h: Merged revisions 91826 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r91826 | russell | 2007-12-07 15:11:08 -0600 (Fri, 07 Dec 2007) | 6 lines I love fixing lock related errors in the lock debugging code. That's about as ironic as it gets in Asterisk programming land. Anyway, I spotted this bug while trying to track down why systems are locking up and acting weird in issue #11080. The mutex attribute object was marked as static in this function when it should not have been. ........ * apps/app_dial.c, /: Merged revisions 91783 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r91783 | russell | 2007-12-07 10:38:48 -0600 (Fri, 07 Dec 2007) | 6 lines * Add channel locking around datastore operations that expect the channel to be locked. * Document why we don't record Local channels in the dialed interfaces list. * Remove the dialed variable as it isn't needed. * Restructure some code for clarity and coding guidelines stuff ........ 2007-12-07 16:37 +0000 [r91782] Jason Parker * channels/chan_sip.c: Fix a small typo in a comment. Closes issue #11490 2007-12-07 16:28 +0000 [r91781] Russell Bryant * /, apps/app_queue.c: Merged revisions 91780 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r91780 | russell | 2007-12-07 10:25:25 -0600 (Fri, 07 Dec 2007) | 7 lines * Add channel locking around datastore operations that expect the channel to be locked. * Document why we don't record Local channels in the dialed interfaces list. * Handle memory allocation failure. * Remove the dialed variable, as it wasn't actually needed. * Tweak some formatting to conform to coding guidelines. ........ 2007-12-07 16:11 +0000 [r91779] Jason Parker * doc/asterisk-mib.txt, main/pbx.c, res/snmp/agent.c, include/asterisk/pbx.h, main/cli.c: Add count of total number of calls processed by asterisk during it's lifetime. Add number of total calls and current calls to SNMP. Closes issue #10057, patch by jcmoore. 2007-12-07 16:11 +0000 [r91778] Russell Bryant * main/autoservice.c, /: Merged revisions 91777 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r91777 | russell | 2007-12-07 10:08:35 -0600 (Fri, 07 Dec 2007) | 6 lines * Add a bit more of a verbose comment as to why a hangup frame needs to be queued up if autoservice gets a NULL return from ast_read(). * Make the process of queueing the hangup frame more efficient by putting the frame where it is going to end up and avoiding some locking and extra memory allocations and freeing. ........ 2007-12-07 15:40 +0000 [r91738] Mark Michelson * main/autoservice.c, /: Merged revisions 91737 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r91737 | mmichelson | 2007-12-07 09:39:58 -0600 (Fri, 07 Dec 2007) | 7 lines Hangups that happen during autoservice were not processed appropriately. This is because a hangup actually causes a NULL frame to be received, not a hangup frame. Queueing a hangup if we receive a NULL frame during autoservice corrects this problem (closes issue #11467, reported by jmls, patched by me) ........ 2007-12-07 02:52 +0000 [r91676-91700] Russell Bryant * apps/app_dial.c, /: Merged revisions 91693 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r91693 | russell | 2007-12-06 20:51:22 -0600 (Thu, 06 Dec 2007) | 2 lines Don't unlock the dialed_interfaces list until we're done messing with the iterator. ........ * apps/app_dial.c, /, apps/app_queue.c: Merged revisions 91677 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r91677 | russell | 2007-12-06 20:38:40 -0600 (Thu, 06 Dec 2007) | 4 lines Allow dialing local channels from Queue() and Dial() again. There was a slight flaw in the code to prevent call forwards from looping that caused this problem. (related to issue #11486) ........ * /, apps/app_queue.c: Merged revisions 91675 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r91675 | russell | 2007-12-06 20:19:45 -0600 (Thu, 06 Dec 2007) | 7 lines Fix in an issue in the call forwarding handling code that was causing crashes on every call into a queue. I'm not entirely sure about the logic in this part of the code, so I want to look at it some more tomorrow. However, this makes it safe and keeps it from crashing. (closes issue #11486, reported by adamg, patched by me) ........ 2007-12-07 00:58 +0000 [r91617-91638] Tilghman Lesher * /, main/rtp.c: Merged revisions 91637 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r91637 | tilghman | 2007-12-06 18:52:17 -0600 (Thu, 06 Dec 2007) | 5 lines At the end of a call, when we're reporting, RTCP may already be partially torn down, so check for NULL dereference Reported by: blitzrage Patch by: tilghman (Closes issue #11450) ........ * channels/chan_zap.c: Add a manager event for PRI events: this will help manager users detect when a D-channel goes down * main/cdr.c: If duration or billsec are not yet calculated, calculate them on demand. 2007-12-06 21:57 +0000 [r91598] Jason Parker * cdr/cdr_sqlite3_custom.c: Fix a problem with quoting in sqlite3 cdr module.. Closes issue #11070, patch by seanbright. 2007-12-06 21:03 +0000 [r91579] Mark Michelson * apps/app_voicemail.c: Handle allocation failure of the heard and deleted arrays of the vm_state. (closes issue #11408, reported and patched by jaroth) 2007-12-06 20:52 +0000 [r91561] Tilghman Lesher * /, cdr/cdr_pgsql.c: Merged revisions 90166,90736,90753 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r90166 | tilghman | 2007-11-29 13:48:10 -0600 (Thu, 29 Nov 2007) | 3 lines Properly escape cdr->src and cdr->dst and ensure we use thread-safe escaping (Fixes AST-2007-026) ........ r90736 | tilghman | 2007-12-03 17:23:55 -0600 (Mon, 03 Dec 2007) | 5 lines If both dbhost and dbsock were not set, a NULL deref could result Reported by: xrg Patch by: tilghman (Closes issue #11387) ........ r90753 | tilghman | 2007-12-03 17:50:51 -0600 (Mon, 03 Dec 2007) | 5 lines Solaris requires the inclusion of sys/loadavg.h for getloadavg(). Reported by: snuffy Patch by: snuffy,tilghman (Closes issue #11430) ........ 2007-12-06 16:54 +0000 [r91472] Matthew Fredrickson * channels/chan_zap.c: Make sure we clear these flags when libpri is not installed 2007-12-06 16:51 +0000 [r91440-91458] Joshua Colp * main/udptl.c, /: Merged revisions 91450 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r91450 | file | 2007-12-06 12:49:42 -0400 (Thu, 06 Dec 2007) | 6 lines Fix various in the udptl implementation. It could return empty modem frames, have an incorrect sequence number on packets, and display the wrong sequence number in the debug messages. (closes issue #11228) Reported by: Cache Patches: udptl-4.patch uploaded by dimas (license 88) ........ * /, channels/chan_sip.c: Merged revisions 91439 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r91439 | file | 2007-12-06 12:14:26 -0400 (Thu, 06 Dec 2007) | 4 lines Add support for accepting and sending T.38 in the initial INVITE. (closes issue #9402) Reported by: thdei ........ 2007-12-06 15:56 +0000 [r91347-91438] Olle Johansson * doc/manager_1_1.txt (added), UPGRADE.txt: Adding documentation for the massive manager changes to manager version 1.1 - hopefully a more consistent manager interface. * main/manager.c: - The Ping Action - Now use Response: success - New header "Ping: pong" :-) - The Events action - Now use Response: Success - The new status is reported as "Events: On" or "Events: Off" - Report if manager is enabled in the reload event Small cleanups... From moremanager * main/channel.c: Changes to manager events in channel.c - Newstate event - Now has "CalleridNum" for numeric caller id, like Newchannel - The event does not send "" for unknown caller IDs just an empty field - Newstate and Newchannel events - these have changed headers "State" -> ChannelStateDesc Text based channel state -> ChannelState Numeric channel state - The events does not send "" for unknown caller IDs just an empty field - Newstate event - Now has "CalleridNum" for numeric caller id, like Newchannel - The event does not send "" for unknown caller IDs just an empty field - Link and Unlink events - The "Link" and "Unlink" bridge events in channel.c are now renamed to "Bridge" - The link state is in the bridgestate: header as "Link" or "Unlink" - For channel.c bridges, "Bridgetype: core" is added. This opens up for bridge events in rtp.c and channel drivers - The "Rename" manager event has a renamed header, to use the same terminology for the current channel as other events - Oldname -> Channel (Moremanager) * main/cdr.c: New manager event when a channel changes account code. Maybe belongs in the new cdr category? ---moremanager--- Event: NewAccountCode Modules: cdr.c Purpose: To report a change in account code for a live channel Example: Event: NewAccountCode Privilege: call,all Channel: SIP/olle-01844600 Uniqueid: 1177530895.2 AccountCode: Stinas account 1234848484 OldAccountCode: Olles Account 12345 * apps/app_dial.c: - Dial event - Event Dial has new headers, to comply with other events - Source -> Channel Channel name (caller) - SrcUniqueID -> UniqueID Uniqueid (new) -> Dialstring Dialstring in app data (moremanager) * apps/app_meetme.c: Adding small missing but important comma... * apps/app_meetme.c: A big oops... * apps/app_meetme.c: The MeetmeJoin now has caller ID name and Caller ID number fields (like MeetMeLeave) (Moremanager) * channels/chan_zap.c: Update ZapShowChannels so that you can specify one channel. Action ZapShowChannels Header changes - Channel: -> ZapChannel For active channels, the Channel: and Uniqueid: headers are added You can now add a "ZapChannel: " argument to zapshowchannels actions to only get information about one channel. From the moremanager branch * main/logger.c: Doxygen updates * include/asterisk/logger.h, /, main/logger.c, main/loader.c: Merged revisions 91366 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r91366 | oej | 2007-12-06 13:54:11 +0100 (Tor, 06 Dec 2007) | 4 lines Make sure logger is reloaded at general reload in the cli. (Discovered during Asterisk training in Portugal) ........ * main/manager.c: Change description of new manager command * main/manager.c, CHANGES: Add manager command for showing all current channels. Thanks, eliel, for writing the original patch. Modified by me to follow other manager events and the new "moremanager" style. (closes issue #11478) Reported by: eliel Patches: manager.c.patch uploaded by eliel (license 64) 2007-12-06 04:37 +0000 [r91328] Joshua Colp * main/channel.c: Instead of iterating through the entire epoll events array just look at the ones that will actually contain data. (props to eliel on IRC for this) 2007-12-05 22:57 +0000 [r91291-91293] Mark Michelson * /, apps/app_voicemail.c: Merged revisions 91292 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r91292 | mmichelson | 2007-12-05 16:57:13 -0600 (Wed, 05 Dec 2007) | 3 lines Reverting extra stuff I didn't mean to commit ........ * apps/app_dial.c, /, apps/app_voicemail.c: Merged revisions 91273 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r91273 | mmichelson | 2007-12-05 16:35:52 -0600 (Wed, 05 Dec 2007) | 11 lines The 'G' option for Dial() did not properly handle the case where only a label was provided. This was due to the fact that the answering channel did not have an extension set, so ast_parseable_goto would fail. This fix eliminates the call to ast_parseable_goto on the answering channel since it is a wasteful call. The answering channel and the calling channel are both directed to the same extension and context, just different priorities, so we can just copy the values from the calling channel to the answering channel and increment the answering channel's priority. (closes issue #11382, reported by jon, patch by me with correction by jon) ........ 2007-12-05 21:46 +0000 [r91238] Tilghman Lesher * /, sounds/Makefile: Merged revisions 91237 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r91237 | tilghman | 2007-12-05 15:38:13 -0600 (Wed, 05 Dec 2007) | 2 lines Upgrade to the latest version of extra sounds ........ 2007-12-05 17:49 +0000 [r91193-91197] Russell Bryant * /, main/threadstorage.c: Merged revisions 91192 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r91192 | russell | 2007-12-05 11:31:42 -0600 (Wed, 05 Dec 2007) | 10 lines Make the lock in the threadstorage debugging code untracked to avoid a deadlock on thread destruction. (closes issue #11207) Reported by: ys Patches: threadstorage.c.diff uploaded by ys (license 281) Also fixes an open bug report: (closes issue #11446) ........ * apps/app_directory.c: Resolve compiler warnings. 2007-12-05 16:46 +0000 [r91172-91173] Tilghman Lesher * main/manager.c, UPGRADE.txt, configs/manager.conf.sample, CHANGES, include/asterisk/manager.h, cdr/cdr_manager.c: Change cdr_manager to use a "CDR" level, rather than the (overcrowded) "call" level. (Closes issue #11015) * CHANGES, apps/app_directory.c: Added multiple name listing. (Closes issue #10413) 2007-12-05 16:14 +0000 [r91171] Joshua Colp * configs/http.conf.sample: Remove second prefix line. Only need it documented once in the same file. (closes issue #11472) Reported by: eserra Patches: http.conf.sample.diff uploaded by eserra (license 45) 2007-12-05 13:09 +0000 [r91151-91152] Olle Johansson * channels/chan_sip.c, UPGRADE.txt, configs/sip.conf.sample: Rename "username" to "defaultuser" to match with "defaultip". "Username" still works, but is deprecated. * channels/chan_sip.c: Remove the cseqs from "sip show channel" and make more place for the call ID. 2007-12-05 03:48 +0000 [r91133] Kevin P. Fleming * channels/chan_zap.c: revert part of my changes from earlier today since this code is no longer dependent on libpri.h 2007-12-05 03:34 +0000 [r91029-91131] Russell Bryant * res/res_odbc.c: Use ast_free() instead of free(). (closes issue #11309) Reported by: Laureano Patches: res_odbc.c.patch uploaded by Laureano (license 265) * /, include/asterisk/lock.h: Merged revisions 91070 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r91070 | russell | 2007-12-04 18:35:31 -0600 (Tue, 04 Dec 2007) | 11 lines Fix some crashes in chan_iax2 that were reported as happening on Mac systems. It turns out that the problem was the Mac version of the ast_atomic_fetchadd_int() function. The Mac atomic add function returns the _new_ value, while this function is supposed to return the old value. So, the crashes happened on unreferencing objects. If the reference count was decreased to 1, ao2_ref() thought that it had been decreased to zero, and called the destructor. However, there was still an outstanding reference around. (closes issue #11176) (closes issue #11289) ........ * /, main/utils.c: Merged revisions 91074 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r91074 | russell | 2007-12-04 18:48:47 -0600 (Tue, 04 Dec 2007) | 4 lines When DEBUG_THREADS is enabled, we only have the details about who is holding a lock that we are waiting on for a mutex, not rwlocks. This should fix the problem where people have reported "core show locks" crashing sometimes. ........ * channels/chan_zap.c: Fix mwimonitornotify on reload ... again. This option was only read at startup so a reload would erase it and not reset it. (pointed out by tzafrir) * utils/astman.c: Fix the build of astman. Any file that includes any asterisk sub-headers needs to first include asterisk.h. (closes issue #11394) 2007-12-04 22:44 +0000 [r91012] Matthew Fredrickson * channels/chan_zap.c: Don't error when we don't have libpri installed with libss7 support. Also, print the debug message anyway if we can't find the right PRI 2007-12-04 22:07 +0000 [r91010-91011] Russell Bryant * main/pbx.c, /: Merged revisions 90967 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r90967 | russell | 2007-12-04 13:57:39 -0600 (Tue, 04 Dec 2007) | 7 lines Make some changes to some additions I made recently for doing channel autoservice when looking up extensions. This code was added to handle the case where a dialplan switch was in use that could block for a long time. However, the way that I added it, it did this for all extension lookups. However, lookups in the in-memory tree of extensions should _not_ take long enough to matter. So, move the autoservice stuff to be only around executing a switch. ........ * channels/chan_zap.c: Fix resetting mwimonitornotify on reload. I guess I only added this line in my head. (thanks to tzafrir for pointing it out) 2007-12-04 21:46 +0000 [r90993] Tilghman Lesher * channels/chan_usbradio.c: Coding guidelines fixups (Closes issue #11412) 2007-12-04 21:23 +0000 [r90991] Jason Parker * channels/chan_sip.c, CHANGES: Add manager action 'sipshowregistry'. Closes issue #11464, patch by eliel. 2007-12-04 19:08 +0000 [r90949] Russell Bryant * include/asterisk/callerid.h, channels/chan_zap.c, main/callerid.c, CHANGES, configs/zapata.conf.sample: Add support for monitoring MWI on FXO lines. This introduces two new options for zapata.conf: mwimonitor and mwimonitornotify. The mwimonitor option enables MWI monitoring. When the MWI state on a line changes, then the script specified by mwimonitornotify will be executed for custom handling of the state change, similar to the externnotify option of voicemail.conf. Also, when the MWI state on an FXO line changes, an internal Asterisk event is generated to indicate the new state of the associated mailbox. That may, any module that cares about MWI information will get notified and can handle it just as if app_voicemail had sent this notification. (BE-253, original patch from markster, with some minor modifications by me to add comments, documentation, and internal event support) 2007-12-04 18:29 +0000 [r90930] Mark Michelson * apps/app_voicemail.c: Kevin suggested doing the reverse of my last commit, since imap_retrieve_file does not modify the contents of the "mailbox" string. In other words, I'm changing the imap_retrieve_file function to take a const char* as the third argument so that I don't need to cast const char*'s as char*'s to suppress compiler warnings. 2007-12-04 18:15 +0000 [r90929] Jason Parker * Makefile: Add Makefile alias target 'pdf' which does the same thing as asterisk.pdf. Issue 11452, reported by blitzrage. 2007-12-04 18:14 +0000 [r90928] Mark Michelson * apps/app_voicemail.c: Suppress a compiler warning due to discarding a "const" qualifier 2007-12-04 18:09 +0000 [r90927] Jason Parker * main/global_datastores.c: Fix build, that some people aren't seeing for some reason. 2007-12-04 17:51 +0000 [r90899] Mark Michelson * apps/app_queue.c: Wrong locking style got merged from 1.4 to trunk. My mistake. 2007-12-04 17:40 +0000 [r90880] Kevin P. Fleming * channels/chan_zap.c: fix build of this module when libpri and/or libss7 are or are not present 2007-12-04 17:38 +0000 [r90879] Jason Parker * main/channel.c, /: Merged revisions 90876 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 (closes issue #11454) ........ r90876 | qwell | 2007-12-04 11:28:08 -0600 (Tue, 04 Dec 2007) | 4 lines If we fail to create a channel after allocating a timing fd, we need to make sure to close it. Issue 11454, patch by eliel. ........ 2007-12-04 17:36 +0000 [r90878] Russell Bryant * main/Makefile: Fix a silly little typo :) 2007-12-04 17:35 +0000 [r90877] Jason Parker * apps/app_dial.c: Fix build in trunk. This was fixed in 1.4, but blocked in trunk since this hadn't been merged yet. 2007-12-04 17:08 +0000 [r90873] Mark Michelson * apps/app_dial.c, main/global_datastores.c (added), channels/chan_local.c, /, main/Makefile, include/asterisk/channel.h, include/asterisk/global_datastores.h (added), apps/app_queue.c: Merged revisions 90735 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r90735 | mmichelson | 2007-12-03 17:12:17 -0600 (Mon, 03 Dec 2007) | 22 lines A big one... This is the merge of the forward-loop branch. The main change here is that call-forwards can no longer loop. This is accomplished by creating a datastore on the calling channel which has a linked list of all devices dialed. If a forward happens, then the local channel which is created inherits the datastore. If, through this progression of forwards and datastore inheritance, a device is attempted to be dialed a second time, it will simply be skipped and a warning message will be printed to the CLI. After the dialing has been completed, the datastore is detached from the channel and destroyed. This change also introduces some side effects to the code which I shall enumerate here: 1. Datastore inheritance has been backported from trunk into 1.4 2. A large chunk of code has been removed from app_dial. This chunk is the section of code which handles the call forward case after the channel has been requested but before it has been called. This was removed because call-forwarding still works fine without it, it makes the code less error-prone should it need changing, and it made this set of changes much less painful to just have the forwarding handled in one place in each module. 3. Two new files, global_datastores.h and .c have been added. These are necessary since the datastore which is attached to the channel may be created and attached in either app_dial or app_queue, so they need a common place to find the datastore info. This approach was taken in case similar datastores are needed in the future, there will be a common place to add them. ........ 2007-12-04 15:16 +0000 [r90852-90854] Olle Johansson * apps/app_queue.c: (closes issue #11431) Reported by: Laureano Patches: app_queue.c.patch uploaded by Laureano (license 265) * main/pbx.c, CHANGES: (closes issue #11422) Reported by: eliel Patches: core.show.hint.patch uploaded by eliel (license 64) * CHANGES: (closes issue #11462) Reported by: eliel Patches: CHANGES.patch uploaded by eliel (license 64) 2007-12-04 15:01 +0000 [r90851] Tilghman Lesher * res/res_agi.c: Pass the Asterisk version to AGI scripts as part of the initial dump of info Reported by: acunningham Patch by: acunningham (Closes issue #11398) 2007-12-04 11:50 +0000 [r90834] Luigi Rizzo * res/Makefile: fix build on cygwin 2007-12-03 23:52 +0000 [r90760] Tilghman Lesher * /, include/asterisk/compat.h: Merged revisions 90753 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r90753 | tilghman | 2007-12-03 17:50:51 -0600 (Mon, 03 Dec 2007) | 5 lines Solaris requires the inclusion of sys/loadavg.h for getloadavg(). Reported by: snuffy Patch by: snuffy,tilghman (Closes issue #11430) ........ 2007-12-03 23:49 +0000 [r90746] Steve Murphy * main/hashtab.c: A small fix from snuffy 2007-12-03 23:48 +0000 [r90738] Jason Parker * res/res_monitor.c: Add manager events for when a monitor is started or stopped. Closes issue #10191, patch by dgradecak. 2007-12-03 23:29 +0000 [r90737] Tilghman Lesher * res/res_config_pgsql.c, /: Merged revisions 90736 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r90736 | tilghman | 2007-12-03 17:23:55 -0600 (Mon, 03 Dec 2007) | 5 lines If both dbhost and dbsock were not set, a NULL deref could result Reported by: xrg Patch by: tilghman (Closes issue #11387) ........ 2007-12-03 22:07 +0000 [r90697] Jason Parker * /, apps/app_meetme.c: Merged revisions 90696 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 (Closes issue #11383) ........ r90696 | qwell | 2007-12-03 16:06:36 -0600 (Mon, 03 Dec 2007) | 4 lines Make sure we always close the conference fd if we have an open one. Issue 11383, reported by markmhy, patch by eliel. ........ 2007-12-03 21:24 +0000 [r90670] Mark Michelson * apps/app_voicemail.c: Replacing some calls to free() with ast_free(). (closes issue #11448, reported and patched by jaroth) 2007-12-03 21:03 +0000 [r90656] Joshua Colp * include/asterisk/agi.h, res/res_agi.c, CHANGES: Add AGI commands for speech recognition. These mirror the dialplan applications mostly but present the information in a nicer fashion. The SPEECH RECOGNIZE command for example will return the results instead of having to query the dialplan functions. 2007-12-03 21:00 +0000 [r90644] Mark Michelson * /, channels/chan_mgcp.c: Merged revisions 90639 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r90639 | mmichelson | 2007-12-03 14:59:51 -0600 (Mon, 03 Dec 2007) | 5 lines Changing some bad logic when calculating the interdigit timeout. (closes issue #11402, reported and patched by eferro) ........ 2007-12-03 20:58 +0000 [r90631] Jason Parker * /, res/res_features.c: Merged revisions 90607 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 (Closes issue #11436) ........ r90607 | qwell | 2007-12-03 14:51:17 -0600 (Mon, 03 Dec 2007) | 4 lines Fix crash in ParkAndAnnounce application. Issue #11436, reported by lytledd, patch by eliel. ........ 2007-12-03 20:30 +0000 [r90591] Tilghman Lesher * main/channel.c: Avoid an additional function call. Reported by: eliel Patch by: eliel (Closes issue #11438) 2007-12-03 20:07 +0000 [r90550-90589] Joshua Colp * /, main/rtp.c: Merged revisions 90588 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r90588 | file | 2007-12-03 16:05:42 -0400 (Mon, 03 Dec 2007) | 2 lines Do not create a smoother for G723.1 frames, they need to be left alone to their native 20/24 byte size. ........ * main/channel.c, /, include/asterisk/channel.h, .cleancount: Merged revisions 90548 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r90548 | file | 2007-12-03 14:40:56 -0400 (Mon, 03 Dec 2007) | 2 lines Preserve the indication currently playing on a channel when a masquerade operation happens. (issue #BE-88) ........ 2007-12-03 16:46 +0000 [r90528] Mark Michelson * configs/queues.conf.sample: Updating sample queues.conf file to show how multiple periodic announcements may be specified since this was not documented previously (closes issue #11432, reported and patched by Laureano) 2007-12-03 14:14 +0000 [r90508] Joshua Colp * apps/app_dial.c: Remove the file descriptors from the main poll channel when the channel is hung up during the dialing attempt, and make sure a channel exists before trying to remove it at the end. (closes issue #11441) Reported by: blitzrage 2007-12-02 18:20 +0000 [r90471] Russell Bryant * /, apps/app_queue.c: Merged revisions 90470 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r90470 | russell | 2007-12-02 12:18:52 -0600 (Sun, 02 Dec 2007) | 6 lines The other day when I went through making changes as a result of the ao2_link() change, I added some code to set pointers to NULL after they were unreferenced. This pointed out that in this place, the object was unreferenced before the code was done using it. So, move the unref down a little bit. (crash reported by jmls on IRC) ........ 2007-12-02 09:42 +0000 [r90433] Tilghman Lesher * main/autoservice.c, /: Merged revisions 90432 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r90432 | tilghman | 2007-12-02 03:34:23 -0600 (Sun, 02 Dec 2007) | 7 lines Clarify the return value on autoservice. Specifically, if you started autoservice and autoservice was already on, it would erroneously return an error. Reported by: adiemus Patch by: dimas (Closes issue #11433) ........ 2007-12-01 01:37 +0000 [r90410] Jason Parker * res/res_adsi.c: Only reload if the config file has changed. Closes issue #11281, patch by eliel. 2007-11-30 21:19 +0000 [r90388] Mark Michelson * apps/app_dial.c, include/asterisk/app.h, include/asterisk/audiohook.h, res/res_features.c, include/asterisk/channel.h, main/audiohook.c, apps/app_queue.c, configs/features.conf.sample: Adding support for the "automixmonitor" dial and queue options. This works in much the same way as the automonitor, except that instead of using the monitor app, it uses the mixmonitor app. By providing an 'x' or 'X' as a dial or queue option, a DTMF sequence may be entered (as defined in features.conf) to start the one-touch mixmonitor. This patch also introduces some new API calls to the audiohooks code for searching for an audiohook by type and for searching for a running audiohook by type. Big thanks to joetester for writing the initial patch, testing it and patiently waiting for it to be committed. (closes issue #10185, reported and patched by xmarksthespot) 2007-11-30 19:34 +0000 [r90311-90351] Russell Bryant * main/manager.c, /, include/asterisk/astobj2.h, apps/app_queue.c, channels/chan_iax2.c, main/astobj2.c, main/config.c: Merged revisions 90348 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r90348 | russell | 2007-11-30 13:26:04 -0600 (Fri, 30 Nov 2007) | 8 lines Change the behavior of ao2_link(). Previously, in inherited a reference. Now, it automatically increases the reference count to reflect the reference that is now held by the container. This was done to be more consistent with ao2_unlink(), which automatically releases the reference held by the container. It also makes it so it is no longer possible for a pointer to be invalid after ao2_link() returns. ........ * /, include/asterisk/astobj2.h: Merged revisions 90310 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r90310 | russell | 2007-11-30 12:46:46 -0600 (Fri, 30 Nov 2007) | 2 lines Add some notes on the behavior of ao2_unlink() after a discussion with Tilghman ........ 2007-11-30 14:45 +0000 [r90270] Joshua Colp * /, channels/chan_sip.c: Merged revisions 90269 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r90269 | file | 2007-11-30 10:43:15 -0400 (Fri, 30 Nov 2007) | 6 lines Fix locking issues under one legged replaces scenarios. (closes issue #11420) Reported by: irroot Patches: chan_sip_oneleg.patch uploaded by irroot (license 52) ........ 2007-11-30 00:16 +0000 [r90164-90232] Mark Michelson * /, channels/chan_mgcp.c: Merged revisions 90231 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r90231 | mmichelson | 2007-11-29 18:16:04 -0600 (Thu, 29 Nov 2007) | 5 lines Clear the DTMF buffer if the call times out. (closes issue #11418, reported and patched by eferro) ........ * /, apps/app_queue.c: Merged revisions 90163 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r90163 | mmichelson | 2007-11-29 13:38:39 -0600 (Thu, 29 Nov 2007) | 6 lines This patch handles the case where a queue member with a negative penalty is added via the manager. If a negative value is submitted for a member penalty, we set it to 0. (closes issue #11411, reported and patched by Laureano) ........ 2007-11-29 19:35 +0000 [r90156-90162] Tilghman Lesher * res/res_config_pgsql.c, /: Merged revisions 90160 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r90160 | tilghman | 2007-11-29 13:24:11 -0600 (Thu, 29 Nov 2007) | 2 lines Properly escape input buffers (Fixes AST-2007-025) ........ * /, formats/format_wav.c, formats/format_pcm.c, formats/format_ogg_vorbis.c, main/file.c, include/asterisk/mod_format.h, formats/format_h263.c, formats/format_h264.c, formats/format_wav_gsm.c, formats/format_g726.c: Merged revisions 90155 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r90155 | tilghman | 2007-11-29 11:29:59 -0600 (Thu, 29 Nov 2007) | 5 lines Use of "private" as a field name in a header file messes with C++ projects Reported by: chewbacca Patch by: casper (Closes issue #11401) ........ * include/asterisk/lock.h: Fix build of trunk * /, sounds/Makefile: Merged revisions 90154 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r90154 | tilghman | 2007-11-29 11:18:09 -0600 (Thu, 29 Nov 2007) | 2 lines Upgrade the core sounds release version ........ 2007-11-29 13:38 +0000 [r90149-90150] Kevin P. Fleming * utils/Makefile, utils/hashtest.c: let's try this again... *all* compilation and linking in Asterisk should be done using the standard compilation rules, not manually created ones. changing hashtest.c to use these rules caused the compiler to notice a large number of coding guidelines violations, so those are fixed too. * main/manager.c: restore behavior from the 1.4 branch... manager users created via users.conf should default to *all* permissions, not none 2007-11-29 00:37 +0000 [r90139-90148] Russell Bryant * main/channel.c, /, include/asterisk/channel.h, funcs/func_callerid.c: Merged revisions 90145 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r90145 | russell | 2007-11-28 18:20:34 -0600 (Wed, 28 Nov 2007) | 5 lines This set of changes is to make some callerID handling thread-safe. The ast_set_callerid() function needed to lock the channel. Also, the handlers for the CALLERID() dialplan function needed to lock the channel when reading or writing callerid values directly on the channel structure. ........ * include/asterisk/file.h, /, main/file.c: Merged revisions 90142 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r90142 | russell | 2007-11-28 18:06:08 -0600 (Wed, 28 Nov 2007) | 4 lines Merge a change from team/russell/chan_refcount ... This makes ast_stopstream() thread-safe. ........ * include/asterisk/audiohook.h: Merge another small doxygen change from team/russell/chan_refcount to indicate that a channel doesn't need to be locked before calling a certain function. * include/asterisk/channel.h: Merge some channel.h doxygen updates from team/russell/chan_refcount This was mostly to note whether a channel needed to be locked or not before calling these functions. However, I added some other things, too. 2007-11-28 23:03 +0000 [r90102] Joshua Colp * /, res/res_musiconhold.c, apps/app_queue.c: Merged revisions 90101 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r90101 | file | 2007-11-28 18:59:28 -0400 (Wed, 28 Nov 2007) | 6 lines Fix a few memory leaks. (closes issue #11405) Reported by: eliel Patches: load_realtime.patch uploaded by eliel (license 64) ........ 2007-11-28 22:44 +0000 [r90100] Kevin P. Fleming * configs/users.conf.sample, main/manager.c, /: Merged revisions 90098 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r90098 | kpfleming | 2007-11-28 16:30:46 -0600 (Wed, 28 Nov 2007) | 2 lines it is impossible to set permissions for manager accounts created by users.conf (reported internally, patched by me) ........ 2007-11-28 22:32 +0000 [r90099] Joshua Colp * main/cli.c: file says... compile before you commit! 2007-11-28 22:17 +0000 [r90060-90061] Mark Michelson * main/pbx.c: Removing a pointless check of option_debug * main/pbx.c, /: Merged revisions 90059 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r90059 | mmichelson | 2007-11-28 16:08:50 -0600 (Wed, 28 Nov 2007) | 13 lines Removing some seemingly pointless code. This sets a channel variable for every priority executed in the dialplan if you have debug set to anything non-zero. This seems pointless due to the fact that these channel variables are not referenced anywhere else in the code and their names are esoteric enough that they would not be practical to reference in the dialplan. Plus the fact that this behavior isn't documented anywhere means that the change is not likely to cause any disruption. If anything, this may actually cause a slight performance increase if running with debug on. The motivating influence for this code change is the eventwhencalled option for queues. If set to vars, all channel variables will be output to the manager. These unnecessary channel variables make the output a lot more difficult to deal with. ........ 2007-11-28 20:33 +0000 [r90039] Steve Murphy * main/ast_expr2f.c, main/ast_expr2.fl: Made expr2 parser 8-bit transparent 2007-11-28 20:27 +0000 [r90038] Jason Parker * main/pbx.c, res/res_crypto.c, include/asterisk/cli.h, main/cli.c: Remove "old"-style CLI handler, since nothing uses it anymore. Closes issue #11403, patch by eliel. This also completes the janitor project. 2007-11-28 15:48 +0000 [r89981-89982] Joshua Colp * main/cli.c: Hide CLI commands starting with _ from tab completion as was done previously. (closes issue #11395) Reported by: eliel Patches: cli.c.patch uploaded by eliel (license 64) * main/abstract_jb.c, res/res_agi.c: Fix a few log messages. (closes issue #11396) Reported by: IgorG Patches: spell.v1.diff uploaded by IgorG (license 20) 2007-11-28 00:49 +0000 [r89947] Russell Bryant * apps/app_voicemail.c: Merge some little changes from team/russell/chan_refcount to help reduce the diff to trunk. This just removes some checks on the return value of alloca(), as behavior is undefined if it runs out of stack space, and we don't check it anywhere else. 2007-11-28 00:47 +0000 [r89946] Mark Michelson * configs/musiconhold.conf.sample, configs/extconfig.conf.sample, res/res_musiconhold.c, CHANGES: Adding support for realtime music on hold. The following are the main points: 1. When moh is started, we search first in memory to find the class. If we do not find it in memory, we search realtime instead. 2. When moh is restarted (as in, it had been started on this particular channel, stopped, and now we're starting it again), if using the "files" mode, then realtime will always be rechecked. If you are using other modes, however, we will simply reattach to the external running process which was playing moh earlier in the call. This is a necessary compromise so that we don't end up with too many background processes. 3. musiconhold.conf has a general section now. It has one option: cachertclasses. If set to yes, then moh classes found in realtime will be added to the in-memory list. This has the advantage of not requiring database lookups each time moh is started, but it has the disadvantage of not truly being realtime. I have tested this for functionality, and it passes. I also tested this under valgrind and there are no memory problems reported under typical use. Special thanks to Sergee for implementing this feature and enduring my complaints on the bugtracker! (closes issue #11196, reported and patched by sergee) 2007-11-28 00:24 +0000 [r89840-89915] Russell Bryant * main/pbx.c, /, include/asterisk/pbx.h: Merged revisions 89893 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r89893 | russell | 2007-11-27 18:20:13 -0600 (Tue, 27 Nov 2007) | 4 lines - update documentation for some of the goto functions to note that they handle locking the channel as needed - update ast_explicit_goto() to lock the channel as needed ........ * include/asterisk/channel.h: Document that the channel is not locked when the send_digit_begin and end callbacks get called. * main/autoservice.c, /: Merged revisions 89886 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r89886 | russell | 2007-11-27 17:47:28 -0600 (Tue, 27 Nov 2007) | 2 lines Don't do frame processing if ast_read() returned NULL. ........ * channels/chan_iax2.c: Merge changes from team/russell/iax2_frame_queue This patch is an optimization for chan_iax2. This module is now heavily multi-threaded. However, there is still a good number of globally shared resources that prevent things from happen asynchronously. One of those things was the global IAX frame queue. This queue was used to hold frames that have been deferred for transmitting by another thread, and frames that may need to get retransmitted. I changed the frame queue to be per-call, since almost all of the frame queue handling only cares about frames specific to a call number. * /, apps/app_queue.c: Merged revisions 89844 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r89844 | russell | 2007-11-27 17:21:13 -0600 (Tue, 27 Nov 2007) | 3 lines Instead of depending on the return value of ast_true(), explicitly set the eventwhencalled variable to 1. ........ * main/pbx.c, /: Merged revisions 89839 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r89839 | russell | 2007-11-27 17:16:00 -0600 (Tue, 27 Nov 2007) | 2 lines Don't start/stop autoservice in pbx_extension_helper() unless a channel exists ........ 2007-11-27 23:11 +0000 [r89838] Mark Michelson * /, apps/app_queue.c: Merged revisions 89837 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r89837 | mmichelson | 2007-11-27 17:10:05 -0600 (Tue, 27 Nov 2007) | 12 lines Two changes with regards to the 'eventwhencalled' option of queues.conf 1) Due to some signed vs. unsigned silliness, setting 'eventwhencalled' to 'vars' or 'yes' did exactly the same thing. Thus the sign change of the ast_true call. 2) The vars2manager function overwrote a \n for every channel variable it parsed, resulting in bizarre output for the channel variables. This patch remedies this. (related to issue #11385, however I'm not sure if this will actually be enough to close it) ........ 2007-11-27 22:42 +0000 [r89835] Russell Bryant * channels/chan_misdn.c: Bring in a small change from team/russell/chan_refcount This replaces tab completion code with the use of a public function that does the same thing 2007-11-27 22:14 +0000 [r89792] Steve Murphy * main/pbx.c, pbx/pbx_config.c: closes issue #11294; missed the conditional unlock of the contexts when the hash table is used instead; also, used the ast_free_ptr as advised. 2007-11-27 22:05 +0000 [r89791] Russell Bryant * main/autoservice.c, main/pbx.c, /: Merged revisions 89790 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r89790 | russell | 2007-11-27 15:45:51 -0600 (Tue, 27 Nov 2007) | 41 lines Merge changes from team/russell/autoservice_1.4 This set of changes fixes an issue that was reported to me on IRC yesterday. The user, d1mas, was using chan_zap for incoming calls and was having DTMF recognition issues in some situations. Specifically, he noticed that the problem occurred when using DISA or WaitExten. He also noticed that when using Read, the problem did not occur. His system also used DUNDi for dialplan lookups. So, he theorized that if the DUNDi lookups blocked for some period of time, that audio from the zap channel could get lost. If the audio got lost, then it wouldn't be run through the DTMF detector, and digits could get lost. He was correct, and the following set of changes fixes the problem. However, the changes go a little bit further than what was necessary to fix this exact problem. 1) I updated pbx_extension_helper() to autoservice the associated channel to handle cases where extension lookups may take a long time. This would normally be a dialplan switch that does some lookup over the network, such as the DUNDi or IAX2 switches. This ensures that even while a DUNDi lookup is blocking, the channel will be continuously serviced. 2) I made a change to the autoservice code. This is actually something that has bothered me for a long time. When a channel is in autoservice, _all_ frames get thrown away. However, some frames really shouldn't be thrown away. The most notable examples are signalling (CONTROL) frames, and DTMF. So, this patch queues up important frames while a channel is in autoservice. When autoservice is stopped on the channel, the queued up frames get stuck back on the channel so that they can get processed instead of thrown away. 3) I made another change to the autoservice code to handle the case where autoservice is started on channels recursively. Previously, you could call ast_autoservice_start() multiple times on a channel, and it would stop the first time ast_autoservice_stop() gets called. Now, it will ensure that autoservice doesn't actually stop until the final call to ast_autoservice_stop(). ........ 2007-11-27 21:10 +0000 [r89769-89772] Olle Johansson * main/dnsmgr.c, res/res_jabber.c, main/enum.c, main/asterisk.c: A few more "moremanager" fixes * include/asterisk.h, main/asterisk.c, main/loader.c: More "moremanager" fixes. Manager commands to check module status. * include/asterisk/manager.h: More "moremanager" changes - doxygen docs and changing manager version (finally) before making more dramatic changes. * channels/chan_iax2.c: More additions from the "moremanager" branch, this time for IAX2. 2007-11-27 20:21 +0000 [r89721] Kevin P. Fleming * /, main/app.c: Merged revisions 89709 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r89709 | kpfleming | 2007-11-27 14:16:56 -0600 (Tue, 27 Nov 2007) | 2 lines on second thought... revert all the other changes i've made in app options parsing leaving only one: if an empty argument is supplied for an option, set that argument pointer to point to an empty string rather than NULL, so that the application can do normal checks on it without worrying about it being NULL ........ 2007-11-27 20:17 +0000 [r89710] Russell Bryant * channels/chan_sip.c: remove a duplicate manager event 2007-11-27 19:50 +0000 [r89706] Olle Johansson * channels/chan_gtalk.c: Manager events from the "moremanager" branch 2007-11-27 19:47 +0000 [r89704] Kevin P. Fleming * /, main/app.c: Merged revisions 89701 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r89701 | kpfleming | 2007-11-27 13:36:55 -0600 (Tue, 27 Nov 2007) | 2 lines generate a warning when an application option that requires an argument is ignored due to lack of an argument ........ 2007-11-27 19:45 +0000 [r89698-89702] Olle Johansson * channels/chan_sip.c: Starting to merge changes from the "moremanager" branch. Documentation will follow. * /, channels/chan_sip.c, include/asterisk/rtp.h, main/rtp.c: The following patch with updates for trunk. Works much better in trunk. Also by accident fixed a bad typo by a previous committer, which actually made video calls not work fully... Merged revisions 89630 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r89630 | oej | 2007-11-27 16:23:17 +0100 (Tis, 27 Nov 2007) | 12 lines If we get a codec offer using a well-known payload type, but using it for another codec that we don't know, Asterisk did not remove that codec from the list. With this patch, we remove the codec from audio and video rtp objects and deny it ever existed. Thanks to lasse for testing. (closes issue #11376) Reported by: lasse Patches: bug11376.txt uploaded by oej (license 306) Tested by: lasse ........ 2007-11-27 19:12 +0000 [r89683] Jason Parker * include/asterisk/strings.h: Add an S_COR macro, which is similar to the existing S_OR macro, except with an additional boolean arg. A hack such as: foo ? S_OR(bar, "baz") : "baz" becomes: S_COR(foo, bar, "baz") 2007-11-27 18:50 +0000 [r89682] Steve Murphy * res/ael/ael.y, pbx/ael/ael-test/ref.ael-test11, pbx/ael/ael-test/ref.ael-test20, pbx/ael/ael-test/ref.ael-test14, pbx/ael/ael-test/ref.ael-test15, pbx/ael/ael-test/ref.ael-ntest9, pbx/ael/ael-test/ref.ael-test16, pbx/ael/ael-test/ref.ael-test18, pbx/ael/ael-test/ref.ael-test19, pbx/ael/ael-test/ref.ael-ntest10, res/ael/ael.tab.c, pbx/ael/ael-test/ref.ael-test1, pbx/ael/ael-test/ref.ael-ntest12, pbx/ael/ael-test/ref.ael-test2, pbx/ael/ael-test/ref.ael-ntest22, res/ael/ael_lex.c, pbx/ael/ael-test/ref.ael-test3, pbx/ael/ael-test/ref.ael-test4, pbx/ael/ael-test/ref.ael-test5, pbx/ael/ael-test/ref.ael-test6, res/ael/ael.flex, pbx/ael/ael-test/ref.ael-test7, pbx/ael/ael-test/ref.ael-test8: made AEL 8-bit transparent; mainly the lexer was tossing chars with the hi-order bit set. Not nice. Also, allow @ in extension names, and a backslash, also. 2007-11-27 17:01 +0000 [r89637] Joshua Colp * main/utils.c: Ensure the value returned from ast_random is between 0 and RAND_MAX on 64-bit platforms. (closes issue #11348) Reported by: sperreault 2007-11-27 16:13 +0000 [r89635] Russell Bryant * /, configs/voicemail.conf.sample: Merged revisions 89634 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r89634 | russell | 2007-11-27 10:12:33 -0600 (Tue, 27 Nov 2007) | 3 lines Add a note to the sample voicemail config noting that when using IMAP storage, only the first format specified will be attached to the message. ........ 2007-11-27 15:41 +0000 [r89632] Tilghman Lesher * /, funcs/func_env.c: Merged revisions 89631 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r89631 | tilghman | 2007-11-27 09:38:03 -0600 (Tue, 27 Nov 2007) | 3 lines Default result of STAT should be "0" not "". Reported via the -users mailing list, fixed by me. ........ 2007-11-27 07:36 +0000 [r89625] Olle Johansson * /, configs/sip.conf.sample: Merged revisions 89624 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r89624 | oej | 2007-11-27 08:34:19 +0100 (Tis, 27 Nov 2007) | 6 lines Clarify limitonpeers=yes (closes issue #11304) Reported by: pj ........ 2007-11-27 06:47 +0000 [r89623] Steve Murphy * apps/app_dial.c, main/cdr.c, /, configs/cdr.conf.sample, include/asterisk/cdr.h: Merged revisions 89622 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r89622 | murf | 2007-11-26 23:24:02 -0700 (Mon, 26 Nov 2007) | 1 line closes issue #11379; OK, this is an attempt to make both sides happy. To the cdr.conf file, I added the option 'unanswered', which defaults to 'no'. In this mode, you will see a cdr for a call, whether it was answered or not. The disposition will be NO ANSWER or ANSWERED, as appropriate. The src is as you'd expect, the destination channel will be one of the channels from the Dial() call, usually the last in the list if more than one chan was specified. With unanswered set to 'yes', you will still see this cdr entry in both cases. But in the case where the dial timed out, you will also see a cdr for each line attempted, marked NO ANSWER, with no destination channel name. The new option defaults to 'no', so you don't see the pesky extra cdr's by default, and you will not see the irritating 'not posted' messages. ........ 2007-11-26 23:15 +0000 [r89617-89621] Mark Michelson * pbx/ael/ael-test/ael-test19/extensions.ael, pbx/ael/ael-test/ael-vtest13/extensions.ael, doc/osp.txt, pbx/ael/ael-test/ael-test3/extensions.ael, pbx/ael/ael-test/ref.ael-vtest13, pbx/ael/ael-test/ael-test7/extensions.ael: Change all instances of "CALLERID(number)" to "CALLERID(num)" for consistency's sake (closes issue #11381, reported and patched by jon) * /, apps/app_playback.c: Merged revisions 89618 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r89618 | mmichelson | 2007-11-26 17:10:49 -0600 (Mon, 26 Nov 2007) | 7 lines After issuing a "say load new", if a caller hangs up during the middle of playback of a number, app_playback will continue to try to play the remaining files. With this change, no more files will be played back upon hangup. (closes issue #11345, reported and patched by IgorG) ........ 2007-11-26 22:52 +0000 [r89615] Russell Bryant * configure, configure.ac: Update the configure script check for libpri to check for the newest function that was just added. Cresl1n, please keep this in mind when making these changes to libpri or libss7. 2007-11-26 21:23 +0000 [r89613] Olle Johansson * channels/chan_sip.c, configs/sip.conf.sample: Rename "limitonpeer" to "counteronpeer" since the call-limit is deprecated. Both still works in this version. 2007-11-26 21:14 +0000 [r89612] Joshua Colp * main/dial.c, /: Merged revisions 89610 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r89610 | file | 2007-11-26 17:10:29 -0400 (Mon, 26 Nov 2007) | 2 lines Fix issues with async dialing with an application executing. The application has to be terminated and control returned to the thread before hanging things up. (issue #BE-252) ........ 2007-11-26 21:12 +0000 [r89606-89611] Olle Johansson * channels/chan_sip.c: Formatting, doxygenification * channels/chan_sip.c: Formatting changes, cleaning up some code * include/asterisk/doxyref.h, channels/chan_sip.c: Start using Doxygen groupings to group variables and defines. * apps/app_meetme.c, UPGRADE.txt, CHANGES, main/cli.c: - Mark "concise" as deprecated - Restructure other changes to UPGRADE.txt and CHANGES We're still looking for scripts that replace asterisk -rx "show shannels concise" by using the manager interface, but still produces the same output. Anyone? 2007-11-26 18:11 +0000 [r89600-89602] Joshua Colp * res/res_features.c, apps/app_queue.c: Perform some module use counting audits. This is now done outside the scope of the application/dialplan function so they do not need to worry about it. * /, res/res_features.c: Merged revisions 89599 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r89599 | file | 2007-11-26 14:02:56 -0400 (Mon, 26 Nov 2007) | 6 lines Add module counting removal for error conditions. (closes issue #11333) Reported by: Laureano Patches: res_features_v2.c.patch uploaded by Laureano (license 265) ........ 2007-11-26 17:49 +0000 [r89596] Russell Bryant * main/pbx.c, /: Merged revisions 89594 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r89594 | russell | 2007-11-26 11:41:04 -0600 (Mon, 26 Nov 2007) | 3 lines Add channel locking to a function that needed to be doing it. This is just a little something I noticed while working on a completely unrelated issue. ........ 2007-11-26 17:46 +0000 [r89595] Steve Murphy * utils/ael_main.c, utils/conf2ael.c, utils/check_expr.c: closes issue #11341; made changes to make utils again right with the MTX_PROFILE world. 2007-11-26 17:38 +0000 [r89593] Joshua Colp * /, pbx/pbx_config.c: Merged revisions 89592 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r89592 | file | 2007-11-26 13:36:45 -0400 (Mon, 26 Nov 2007) | 6 lines Use ast_free to free memory, or else we shall implode if MALLOC_DEBUG is enabled. (closes issue #11347) Reported by: ys Patches: pbx.pbx_config.c.diff uploaded by ys (license 281) ........ 2007-11-26 17:26 +0000 [r89591] Steve Murphy * main/hashtab.c: closes issue #11356; Many thanks to snuffy for his code review and changes to cut down duplication. I tested this against hashtest, and it passes. I reviewed the changes, and they look reasonable. I had to remove a few const decls to make things compile on my workstation, 2007-11-26 17:25 +0000 [r89590] Russell Bryant * Makefile: make sure we check to see if the configure script has been executed on a new checkout or after a distclean 2007-11-26 17:23 +0000 [r89589] Joshua Colp * /, apps/app_mixmonitor.c: Merged revisions 89587 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r89587 | file | 2007-11-26 13:20:58 -0400 (Mon, 26 Nov 2007) | 6 lines Close the audio file before sending it to the post processing application. (closes issue #11357) Reported by: reformed Patches: mixmonitor.patch uploaded by reformed (license 330) ........ 2007-11-26 17:21 +0000 [r89588] Kevin P. Fleming * /, main/app.c, apps/app_controlplayback.c: Merged revisions 89586 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r89586 | kpfleming | 2007-11-26 11:20:36 -0600 (Mon, 26 Nov 2007) | 2 lines when parsing application options that take arguments, don't indicate that the option was supplied unless a non-zero-length argument was found for it ........ 2007-11-26 16:24 +0000 [r89583] Steve Murphy * main/pbx.c, CHANGES, configs/extensions.conf.sample: Thanks to pnlarsson for noting the spelling error in the cli commands. Also, added some verbage about the new algorithm to CHANGES. 2007-11-26 16:20 +0000 [r89582] Joshua Colp * main/utils.c: Revert change for 11348 until it can be looked at even more. 2007-11-26 15:50 +0000 [r89581] Mark Michelson * /, apps/app_voicemail.c: Merged revisions 89580 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r89580 | mmichelson | 2007-11-26 09:48:06 -0600 (Mon, 26 Nov 2007) | 6 lines Revert vmu->email back to an empty string if it was empty when imap_store_file was called. This prevents sending a duplicate e-mail. (closes issue #11204, reported by spditner, patched by me) ........ 2007-11-26 15:36 +0000 [r89570-89578] Joshua Colp * main/channel.c, /: Merged revisions 89577 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r89577 | file | 2007-11-26 11:34:38 -0400 (Mon, 26 Nov 2007) | 6 lines If channel allocation fails because the alert pipe could not be created also free the scheduler context. (closes issue #11355) Reported by: eliel Patches: main.channel.c.patch uploaded by eliel (license 64) ........ * main/utils.c: Make the behavior of using /dev/urandom for random numbers the same as random(). (closes issue #11348) Reported by: sperreault Patches: ast_random2.diff uploaded by sperreault (license 252) * channels/chan_sip.c: Instead of printing out one codec in sip show channels print out all of the native ones (this is for video). (closes issue #11366) Reported by: ovi * /, apps/app_meetme.c: Merged revisions 89571 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r89571 | file | 2007-11-26 10:41:03 -0400 (Mon, 26 Nov 2007) | 4 lines When unloading app_meetme destroy any auto created contexts created by SLA. (closes issue #11367) Reported by: eliel ........ * apps/app_controlplayback.c: Don't crash if the 'o' option of ControlPlayback is used without any value. (closes issue #11375) Reported by: johan 2007-11-25 21:12 +0000 [r89564-89566] Olle Johansson * channels/chan_usbradio.c: Formatting changes * main/channel.c, include/asterisk/channel.h: Try to get channel.h and channel.c aligned in regards to ast_set_callerid as well as change name of variables to follow the rest of the naming. 2007-11-25 17:50 +0000 [r89560-89561] Tilghman Lesher * include/asterisk/res_odbc.h, res/res_config_odbc.c, /, res/res_odbc.c, configs/res_odbc.conf.sample: Merged revisions 89559 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r89559 | tilghman | 2007-11-25 11:17:10 -0600 (Sun, 25 Nov 2007) | 14 lines We previously attempted to use the ESCAPE clause to set the escape delimiter to a backslash. Unfortunately, this does not universally work on all databases, since on databases which natively use the backslash as a delimiter, the backslash itself needs to be delimited, but on other databases that have no delimiter, backslashing the backslash causes an error. So the only solution that I can come up with is to create an option in res_odbc that explicitly specifies whether or not backslash is a native delimiter. If it is, we use it natively; if not, we use the ESCAPE clause to make it one. Reported by: elguero Patch by: tilghman (Closes issue #11364) ........ * channels/chan_sip.c: Typo (someone needs to test compile before committing his changes) 2007-11-25 12:18 +0000 [r89551-89557] Olle Johansson * channels/chan_sip.c: More doxygen changes * channels/chan_sip.c: Housekeeping * channels/chan_sip.c: Formatting, doxygen updates * channels/chan_sip.c, configs/sip.conf.sample, CHANGES: - Deprecate "call-limit" in chan_sip. No other channel driver enforces call-limits and we now have the groupcount system to implement call-limits in the dialplan. You can use the "setvar" option in realtime/sip.conf to set limits per device. - Implement "callcounter" as a new option to enable the call counting we need to report device status to queue, manager and SIP subscriptions. The call counter setting is now enabled in the code by setting the device call-limit to 999. When we remove the call limit, we can simply enable this with a boolean setting. * channels/chan_sip.c, include/asterisk/channel.h: Housekeeping... - Fix typo in chan_sip - Remove changes to caller ID structure, moving it to branch (russellb) 2007-11-24 21:00 +0000 [r89547] Steve Murphy * main/pbx.c, include/asterisk/pbx.h, pbx/pbx_config.c, configs/extensions.conf.sample: closes issue #11363; where the pattern _20x. buried in an included context, didn't match 2012; There were a small set of problems to fix: 1. I needed NOT to score patterns unless you are at the end of the data string. 2. Capital N,X,Z and small n,x,z are OK in patterns. I canonicalize the patterns in the trie to caps. 3. When a pattern ends with dot or exclamation, CANMATCH/MATCHMORE should always report this pattern, no matter the length. With this commit, I also supplied the wish of Luigi, where the user can select which pattern matching algorithm to use, the old (legacy) pattern matcher, or the new, trie based matcher. The OLD matcher is the default. A new [general] section variable, extenpatternmatchnew, is added to the extensions.conf, and the example config has it set to false. If true, the new matcher is used. In all other respects, the context/exten structs are the same; the tries and hashtabs are formed, but in the new mode the tries are not used. A new CLI command 'dialplan set extenpatternmatch true/false' is provided to allow switching at run time. I beg users that are forced to return to the old matcher to please report the reason in the bug tracker. Measured the speed benefit of the new matcher against an impossibly large context with 10,000 extensions: the new matcher is 374 times faster. 2007-11-24 17:07 +0000 [r89546] Tilghman Lesher * /, res/res_adsi.c: Merged revisions 89545 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r89545 | tilghman | 2007-11-24 10:59:59 -0600 (Sat, 24 Nov 2007) | 5 lines Free some frames that would otherwise leak on error. Reported by: Laureano Patch by: Laureano,tilghman (Closes issue #11351) ........ 2007-11-24 16:53 +0000 [r89544] Steve Murphy * main/app.c: Added include to allow trunk to compile. Hope this doesn't louse thing up. 2007-11-24 13:57 +0000 [r89542-89543] Luigi Rizzo * channels/chan_h323.c: remove a DEBUG_THREADS message that accesses private lock fields. If needed, the code to extract this information should be implemented in some generic header or library and the function called here. (closed bug #11362) * main/acl.c, main/http.c, main/app.c: remove some unnecessary includes 2007-11-24 06:24 +0000 [r89535-89541] Tilghman Lesher * /, main/app.c, apps/app_voicemail.c: Merged revisions 89540 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r89540 | tilghman | 2007-11-24 00:19:23 -0600 (Sat, 24 Nov 2007) | 9 lines Currently, zero-length voicemail messages cause a hangup in VoicemailMain. This change fixes the problem, with a multi-faceted approach. First, we do our best to avoid these messages from being created in the first place, and second, if that fails, we detect when the voicemail message is zero-length and avoid exiting at that point. Reported by: dtyoo Patch by: gkloepfer,tilghman (Closes issue #11083) ........ * main/manager.c, /: Merged revisions 89536 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r89536 | tilghman | 2007-11-23 11:18:26 -0600 (Fri, 23 Nov 2007) | 10 lines Up until this point, the XML output of the manager has been technically invalid, due to the repetition of certain parameters in a single event. This caused various issues for XML parsers, some of which refused to parse at all, given the invalidity of the rendered XML. So this commit fixes the XML output, ensuring that each entity parameter has a unique name, thus ensuring valid XML. Reported by: msetim Patch by: tilghman (Closes issue #10220) ........ * res/res_config_odbc.c, /: Merged revisions 89534 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r89534 | tilghman | 2007-11-23 11:05:10 -0600 (Fri, 23 Nov 2007) | 5 lines Use ESCAPE clause for the first parameter, not just 2nd-Nth parameters. Reported by: apsaras Patch by: tilghman (Closes issue #11353) ........ 2007-11-23 15:54 +0000 [r89532-89533] Luigi Rizzo * channels/chan_oss.c: put in the necessary hooks for video support in the console. This is a NOP as far as the current code is concerned, but there is already support in ./configure and the Makefiles for the various libraries used by console_video.c (not yet in the tree) so addition is trivial. * channels/chan_sip.c: set rtpmap video info according to what is read from SDP; make the format explicit in a debug message; print the audio instead of aggregated peer capability in a debugging msg. 2007-11-23 09:40 +0000 [r89531] Olle Johansson * include/asterisk/channel.h: Let's start with implementing the base architecture for UTF8 caller ID's so we can handle multiple formats properly. This is not carved in stone, but a proposal to start with. We need to add support for transliterations as well as UTF8 handling, propably with libiconv. Murf is looking into that for the dialplan. 2007-11-23 09:03 +0000 [r89530] Luigi Rizzo * include/asterisk/image.h, formats/format_jpeg.c: formatting cleanup on the header, normalization of the assignment of descriptor fields. 2007-11-23 02:37 +0000 [r89529] Russell Bryant * configs/agents.conf.sample, /: Merged revisions 89527 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r89527 | russell | 2007-11-22 12:29:41 -0500 (Thu, 22 Nov 2007) | 3 lines mvanbaak pointed out a spelling error in this sample configuration file. While I was at it, I went ahead and tweaked it a little bit more. ........ 2007-11-22 07:10 +0000 [r89514-89526] Luigi Rizzo * doc/CODING-GUIDELINES: new info on the management of headers * apps/app_echo.c, apps/app_sendtext.c, apps/app_verbose.c, apps/app_milliwatt.c: more header removal * include/asterisk/channel.h: formatting cleanup * include/asterisk.h, apps/app_read.c, apps/app_record.c, apps/app_echo.c, apps/app_readexten.c, include/asterisk/channel.h, apps/app_system.c, apps/app_transfer.c, res/ael/pval.c, include/asterisk/app.h, apps/app_dumpchan.c, include/asterisk/module.h, apps/app_url.c, include/asterisk/pbx.h, apps/app_senddtmf.c, pbx/pbx_config.c, apps/app_mixmonitor.c, apps/app_stack.c, apps/app_verbose.c, apps/app_milliwatt.c, apps/app_cdr.c, apps/app_while.c: shuffle a little bit the content of header files to reduce dependencies. In this commit: - move the ast_register/unregister_app functions to module.h to avoid the need to include pbx.h for the simpler apps; - move the ast_group structure to channel.h to remove the dependency of app.h on linkedlists.h Note, this is a long process that I am doing in small steps. The main difficulty is that now for each subsystem we have a single header (e.g. channel.h) included by the subsystem provider (usually one file, e.g. channel.c) and by its clients (dozens of them, e.g. we have some 70+ apps and 30+ functions). This requires the clients to include all the extra headers required by the provider (eg. lock.h, linkedlists.h, definitions of substructures...) even though many of the clients would be just happy with opaque struct declarations and function prototypes. The long term plan is to eventually rectify this structure so that the compilation can become faster, and also APIs are more stable. * funcs/func_md5.c, funcs/func_module.c, funcs/func_blacklist.c, apps/app_url.c, funcs/func_sha1.c, funcs/func_global.c: remove some useless includes * include/asterisk/audiohook.h, apps/app_dictate.c, apps/app_readexten.c, apps/app_directory.c, apps/app_senddtmf.c, apps/app_mixmonitor.c, apps/app_stack.c, apps/app_controlplayback.c: more removal of redundant headers * apps/app_read.c, apps/app_echo.c, apps/app_record.c, apps/app_userevent.c, apps/app_image.c, apps/app_system.c, apps/app_verbose.c, apps/app_milliwatt.c, apps/app_playback.c, apps/app_while.c: remove redundant headers * main/file.c, main/netsock.c: more removal of fcntl.h and other system headers * codecs/codec_lpc10.c, codecs/codec_g722.c, codecs/codec_a_mu.c, codecs/codec_speex.c, codecs/codec_alaw.c, codecs/codec_adpcm.c, res/res_crypto.c, codecs/codec_g726.c, apps/app_test.c, formats/format_ogg_vorbis.c, codecs/codec_gsm.c, res/res_agi.c, apps/app_mp3.c, main/app.c, codecs/codec_ulaw.c, codecs/codec_ilbc.c: remove a number of #include which are either useless or done elsewhere * formats/format_sln.c, formats/format_wav.c, formats/format_ogg_vorbis.c, include/asterisk/_private.h, formats/format_wav_gsm.c, formats/format_ilbc.c, include/asterisk/file.h, formats/format_vox.c, formats/format_pcm.c, main/file.c, formats/format_h263.c, formats/format_g723.c, formats/format_h264.c, include/asterisk/frame.h, formats/format_jpeg.c, formats/format_g726.c, formats/format_gsm.c, formats/format_g729.c: implement the split of file.h and mod_format.h * include/asterisk/mod_format.h (added): Add a specific header for providers of file and format handling routines, moving here structs and function declarations formerly in file.h 2007-11-21 23:54 +0000 [r89513] Steve Murphy * apps/app_dial.c, channels/chan_sip.c, channels/chan_skinny.c, res/res_features.c, apps/app_queue.c, channels/chan_iax2.c: closes issue #11285, where an unload of a module that creates a dialplan context, causes a crash when you do a 'dialplan show' of that context. This is because the registrar string is defined in the module, and the stale pointer is traversed. The reporter offered a patch that would always strdup the registrar string, which is practical, but I preferred to destroy the created contexts in each module where one is created. That seemed more symmetric. There were only 6 place in asterisk where this is done: chan_sip, chan_iax2, chan_skinny, res_features, app_dial, and app_queue. The two apps destroyed the context, but left the contexts. All is fixed now and unloads should be dialplan friendly. 2007-11-21 23:24 +0000 [r89511-89512] Luigi Rizzo * funcs/func_rand.c, cdr/cdr_sqlite3_custom.c, apps/app_readfile.c, channels/chan_local.c, apps/app_record.c, funcs/func_strings.c, apps/app_sayunixtime.c, apps/app_test.c, apps/app_alarmreceiver.c, cdr/cdr_adaptive_odbc.c, apps/app_image.c, apps/app_chanisavail.c, apps/app_ices.c, channels/chan_iax2.c, apps/app_exec.c, pbx/pbx_loopback.c, pbx/pbx_spool.c, channels/chan_skinny.c, apps/app_dumpchan.c, apps/app_zapscan.c, apps/app_zapras.c, pbx/pbx_realtime.c, channels/chan_alsa.c, apps/app_amd.c, apps/app_url.c, apps/app_externalivr.c, cdr/cdr_odbc.c, apps/app_dial.c, funcs/func_timeout.c, apps/app_page.c, apps/app_privacy.c, channels/chan_agent.c, apps/app_disa.c, apps/app_morsecode.c, channels/iax2-provision.c, funcs/func_cut.c, apps/app_talkdetect.c, apps/app_transfer.c, apps/app_db.c, apps/app_playback.c, funcs/func_curl.c, channels/chan_misdn.c, apps/app_zapbarge.c, apps/app_waitforring.c, apps/app_sendtext.c, channels/chan_features.c, apps/app_macro.c, funcs/func_iconv.c, pbx/pbx_config.c, apps/app_mixmonitor.c, apps/app_chanspy.c, apps/app_voicemail.c, channels/chan_unistim.c, channels/chan_vpb.cc, apps/app_meetme.c, apps/app_authenticate.c, apps/app_readexten.c, funcs/func_vmcount.c, channels/chan_gtalk.c, cdr/cdr_pgsql.c, apps/app_followme.c, cdr/cdr_radius.c, apps/app_controlplayback.c, cdr/cdr_csv.c, channels/chan_phone.c, funcs/func_enum.c, apps/app_osplookup.c, funcs/func_odbc.c, apps/app_mp3.c, apps/app_minivm.c, apps/app_rpt.c, channels/chan_mgcp.c, apps/app_parkandannounce.c, apps/app_while.c, apps/app_adsiprog.c, apps/app_nbscat.c, funcs/func_version.c, funcs/func_db.c, channels/chan_zap.c, apps/app_read.c, channels/chan_sip.c, apps/app_festival.c, apps/app_waitforsilence.c, funcs/func_lock.c, pbx/pbx_lua.c, apps/app_system.c, apps/app_getcpeid.c, apps/app_queue.c, channels/chan_oss.c, cdr/cdr_tds.c, funcs/func_realtime.c, channels/chan_jingle.c, channels/chan_usbradio.c, apps/app_channelredirect.c, apps/app_flash.c, apps/app_directed_pickup.c, funcs/func_blacklist.c, channels/chan_h323.c, pbx/pbx_dundi.c, apps/app_sms.c, channels/chan_nbs.c, apps/app_senddtmf.c, funcs/func_callerid.c, apps/app_verbose.c, apps/app_stack.c, pbx/pbx_gtkconsole.c: remove another set of redundant #include "asterisk/options.h" * main/udptl.c, main/autoservice.c, main/frame.c, res/res_snmp.c, main/say.c, res/res_features.c, main/devicestate.c, main/utils.c, res/res_musiconhold.c, res/res_jabber.c, main/indications.c, main/enum.c, res/res_config_sqlite.c, main/config.c, main/loader.c, main/term.c, main/cli.c, main/io.c, main/channel.c, main/cdr.c, main/dial.c, res/res_smdi.c, res/res_config_odbc.c, main/manager.c, res/res_agi.c, main/http.c, main/logger.c, res/res_realtime.c, main/app.c, main/image.c, main/dns.c, main/db.c, res/res_speech.c, main/sched.c, main/pbx.c, res/res_config_pgsql.c, main/dnsmgr.c, main/translate.c, res/res_crypto.c, res/res_adsi.c, main/jitterbuf.c, main/acl.c, formats/format_ogg_vorbis.c, res/res_ael_share.c, res/res_monitor.c, main/rtp.c, main/netsock.c, main/srv.c, main/hashtab.c, main/privacy.c, main/adsistub.c, main/abstract_jb.c, main/file.c, main/callerid.c, main/astmm.c, main/audiohook.c, formats/format_g726.c, main/asterisk.c, res/res_odbc.c, main/dsp.c: remove a bunch of useless #include "options.h" 2007-11-21 22:37 +0000 [r89509-89510] Matthew Fredrickson * channels/chan_zap.c: Remove unneccessary explicit case for BRI * channels/chan_zap.c: Take some debug code out :-) 2007-11-21 22:20 +0000 [r89508] Luigi Rizzo * main/cygload.c: add a missing return 2007-11-21 22:07 +0000 [r89507] Matthew Fredrickson * channels/chan_zap.c: Add BRI support to chan_zap 2007-11-21 21:30 +0000 [r89506] Luigi Rizzo * utils/Makefile, configure, configure.ac: enable support for stack backtrace for stuff built in utils/ (this was present in the main tree but forgotten here). 2007-11-21 20:38 +0000 [r89505] Steve Murphy * main/pbx.c: closes issue #11290; the proposed patch was a good guess, and would solve the bug to some extent, but was really masking the real issue, that there were bad entries in the table. This fix removes the condition that the hashtab updates be done on exten removal only when the pattern_tree was present, which is silly. The operations that apply to the pattern tree are instead made conditional. Also, threw back in routines that kpfleming deleted because of probs in the 64-bit world. Tested on both 32 and 64-bit machines (compile). Tested the reload problem with over 20 reloads, and no problems. If you find more problems, please reopen 11290. 2007-11-21 20:22 +0000 [r89504] Terry Wilson * res/res_features.c: Simplify comparison in parking fix 2007-11-21 19:28 +0000 [r89494-89496] Mark Michelson * /, apps/app_queue.c: Merged revisions 89495 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r89495 | mmichelson | 2007-11-21 13:27:51 -0600 (Wed, 21 Nov 2007) | 3 lines Fix a small error I made in my previous commit ........ * /, apps/app_queue.c: Merged revisions 89493 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r89493 | mmichelson | 2007-11-21 13:24:22 -0600 (Wed, 21 Nov 2007) | 5 lines Changing an inaccurate debug message to be less inaccurate. Under the circumstances, this message would always report that there were 0 members available, even though that may not be true. ........ 2007-11-21 19:20 +0000 [r89492] Terry Wilson * /, res/res_features.c: Merged revisions 89491 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r89491 | twilson | 2007-11-21 12:59:27 -0600 (Wed, 21 Nov 2007) | 4 lines If a channel gets masqueraded in the middle of a park, don't play the announcement to the masqueraded channel, and dial back to the original channel on timeout. ........ 2007-11-21 18:52 +0000 [r89490] Russell Bryant * main/dsp.c: Remove obsolete OLD_DSP_ROUTINES code. Also, remove the FAX_DETECT define and only do the calculations if fax detection is enabled on the dsp. (closes issue #11331) Reported by: dimas Patches: dsp.patch uploaded by dimas (license 88) 2007-11-21 18:38 +0000 [r89489] Tilghman Lesher * apps/app_read.c, UPGRADE.txt, CHANGES: Change Read to set READSTATUS as an indication of the result Also, some cleanup to CHANGES. Reported by: michael-fig Patch by: michael-fig,tilghman (Closes issue #11004) 2007-11-21 18:24 +0000 [r89488] Russell Bryant * channels/chan_iax2.c: fix a small gramatical error in a comment 2007-11-21 18:19 +0000 [r89487] Mark Michelson * main/utils.c: There existed about a 1 in 4 billion chance that reading from /dev/urandom would return LONG_MIN (1 in 9 quintillion if using 64-bit longs). Since there is no positive equivalent of LONG_MIN, the result of labs() in this case is unpredictable. This fixes that situation. (closes issue #11336, reported and patched by sperreault) 2007-11-21 16:24 +0000 [r89484] Russell Bryant * channels/chan_unistim.c: Fix some code that was supposed to ensure that a buffer was terminated, but was writing to the wrong byte. Also, remove some non-thread safe test code. (closes issue #11317) Reported by: IgorG Patches: unistim-2.patch uploaded by IgorG (license 20) - additional changes by me 2007-11-21 16:08 +0000 [r89483] Mark Michelson * main/pbx.c: I introduced a deadlock avoidance into 1.4, which I attempted to port to trunk as well. Unfortunately, since trunk uses read/write locks for the context lock, it means that I have actually *introduced* a deadlock condition since they are not recursive. Removing this change for now and will look into introducing a different one. 2007-11-21 16:07 +0000 [r89480-89482] Kevin P. Fleming * include/asterisk.h, include/asterisk/compat.h, utils/ael_main.c, utils/conf2ael.c: move these forward declarations back to asterisk.h where they belong... even though asterisk.h includes compat.h, these declarations have nothing to do with the being platform-compatible and are directly related to being part of Asterisk * channels/chan_usbradio.c: get this to actually compile... * main/pbx.c: remove some debugging code that doesn't compile on 64-bit platforms 2007-11-21 15:17 +0000 [r89478-89479] Steve Murphy * res/res_features.c: OOOps! All the debug stuff I inserted was accidentally committed. I hereby revert it. * main/hashtab.c, res/res_features.c: closes issue #11265; Thanks to snuffy for his work on neatening up the code and removing duplicated code. 2007-11-21 08:28 +0000 [r89475-89477] Luigi Rizzo * channels/gentone-ulaw.c (removed): remove this file, it is not used anywhere. * main/astmm.c: add missing paths.h * configure, include/asterisk/autoconfig.h.in, configure.ac: add check for video4linux 2007-11-21 01:09 +0000 [r89474] Steve Murphy * main/pbx.c: A free in add_pri was ultimately the source of the grief we were having with parking. This set of changes fixes that problem, and introduces some more error messages, and puts debugs into ifdefs for what could be short-term usage. Txs to Terry W. for his help, guidance, and especially patience. 2007-11-21 00:23 +0000 [r89472-89473] Luigi Rizzo * main/sha1.c, agi/eagi-test.c, utils/smsq.c, utils/hashtest2.c, main/minimime/mm.h, utils/check_expr.c: more header removal/normalization * configure, include/asterisk/autoconfig.h.in, configure.ac: X11 checks (at least some - for other platforms with unusual X11 locations you might need to add more directories) 2007-11-21 00:21 +0000 [r89470] Russell Bryant * apps/app_meetme.c, CHANGES: Merge changes from team/russell/sla_trunk_moh ... * Added the ability to specify the music on hold class used to play into the conference when there is only one member and the M option is used. * Added the ability to specify a music on hold class to play instead of ringing for the SLATrunk application. (patched by me, and tested internally) 2007-11-21 00:20 +0000 [r89469] Luigi Rizzo * makeopts.in: complete support for X11 2007-11-20 23:29 +0000 [r89467-89468] Tilghman Lesher * apps/app_meetme.c, cdr/cdr_sqlite.c, pbx/pbx_lua.c: Make trunk build again * main/say.c: Add support for new recorded character sounds Closes issue #5208 2007-11-20 23:16 +0000 [r89465-89466] Luigi Rizzo * channels/chan_unistim.c, cdr/cdr_sqlite3_custom.c, apps/app_dictate.c, apps/app_test.c, apps/app_ices.c, apps/app_followme.c, channels/chan_iax2.c, main/config.c, main/loader.c, main/cli.c, cdr/cdr_csv.c, main/channel.c, main/manager.c, pbx/pbx_spool.c, include/asterisk/compat.h, res/res_agi.c, apps/app_minivm.c, main/logger.c, main/http.c, main/app.c, main/image.c, apps/app_directory.c, main/db.c, cdr/cdr_custom.c, apps/app_adsiprog.c, apps/app_dial.c, include/asterisk/utils.h, include/asterisk.h, main/pbx.c, channels/chan_sip.c, res/res_crypto.c, include/asterisk/channel.h, res/res_monitor.c, include/asterisk/paths.h, main/file.c, apps/app_sms.c, include/asterisk/ael_structs.h, pbx/pbx_config.c, apps/app_mixmonitor.c, apps/app_chanspy.c, apps/app_voicemail.c: move asterisk/paths.h outside asterisk.h and into those files who really need it. * main/pbx.c, include/asterisk.h, main/frame.c, main/dnsmgr.c, main/threadstorage.c, main/devicestate.c, include/asterisk/_private.h (added), main/astobj2.c, main/loader.c, main/term.c, main/cli.c, main/channel.c, main/manager.c, main/logger.c, build_tools/strip_nonapi, main/event.c, main/asterisk.c, main/db.c: move internal function declarations to include/asterisk/_private.h 2007-11-20 19:29 +0000 [r89464] Russell Bryant * configure, configure.ac: i got a little carried away with commas ... 2007-11-20 19:28 +0000 [r89463] Kevin P. Fleming * include/asterisk/module.h, build_tools/make_buildopts_h, main/loader.c: switch compile-time option checking to string storage mode in this branch too 2007-11-20 19:11 +0000 [r89460] Russell Bryant * configure, configure.ac: fix the zaptel configure script check 2007-11-20 18:20 +0000 [r89459] Luigi Rizzo * acinclude.m4: the 'version' is now $7 not $6 (wait a bit before regenerating configure, i have more changes) 2007-11-20 17:59 +0000 [r89458] Mark Michelson * main/pbx.c, /: Merged revisions 89457 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r89457 | mmichelson | 2007-11-20 11:50:31 -0600 (Tue, 20 Nov 2007) | 9 lines According to comments in main/pbx.c, it is essential that if we are going to lock the conlock as well as the hints lock, it must be locked in that respective order. In order to prevent a potential deadlock, we need to lock the conlock prior to locking the hints lock in ast_hint_state_changed (see the call stack example on issue #11323 for how this can happen). (closes issue #11323, reported by eelcob, suggestion for patch by eelcob, patch by me) ........ 2007-11-20 17:11 +0000 [r89454-89455] Luigi Rizzo * makeopts.in: prepare to support console_video * apps/Makefile, Makefile.moddir_rules, pbx/Makefile, res/Makefile, channels/Makefile: Fix building of modules under cygwin. After this commit we can actually load modules under windows, and we can start debugging more interesting problems related to the load order and functionality of modules. 2007-11-20 16:11 +0000 [r89453] Mark Michelson * configs/sip.conf.sample: Changed occurrences of "busy-level" to "busylevel" in sip.conf.sample in light of commit 89441. Thanks to pj for pointing out the need for this (closes issue #11307, reported by pj) 2007-11-20 15:39 +0000 [r89452] Luigi Rizzo * configure, configure.ac, acinclude.m4: add an argument for extra headers to AC_EXT_LIB_CHECK, and on passing simplify the code. Too bad that every time we need to regenerate configure... 2007-11-20 15:30 +0000 [r89451] Steve Murphy * /, doc/tex/queues-with-callback-members.tex: Merged revisions 89450 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r89450 | murf | 2007-11-20 08:22:08 -0700 (Tue, 20 Nov 2007) | 1 line closes issue #11324; break statements missing in switch cases. ........ 2007-11-20 15:00 +0000 [r89449] Joshua Colp * main/translate.c: Minor documentation tweak and if an incorrect parameter is given to core show translation return the usage information. (closes issue #11316) Reported by: eliel Patches: translate.c.patch uploaded by eliel (license 64) 2007-11-20 14:54 +0000 [r89448] Luigi Rizzo * configure, acinclude.m4: comment a bit the code in acinclude.m4 There is still a lot of code to clean up there, but hopefully this should clarify what goes on in there. 2007-11-20 14:49 +0000 [r89447] Joshua Colp * channels/h323/ast_h323.cxx: Include the compatibility header file in ast_h323.cxx for compatibility reasons. (closes issue #11311) Reported by: falves11 2007-11-20 14:44 +0000 [r89444-89446] Olle Johansson * channels/chan_sip.c: Fix sip show history. Closes issue #11312 * channels/chan_sip.c: Change terminology a bit for CLI commands handling SIP channels/calls/dialogs/whatever. Closes issue #11312 2007-11-20 07:42 +0000 [r89443] Luigi Rizzo * Makefile, main/Makefile, Makefile.moddir_rules: initial makefile changes to build loadable modules under cygwin (not complete yet - still need to sort out dependecies on res_*) 2007-11-20 00:17 +0000 [r89442] Steve Murphy * main/pbx.c: Get rid of some debug messages in pbx.c 2007-11-19 23:24 +0000 [r89441] Mark Michelson * channels/chan_sip.c, CHANGES: Changed the "busy-level" option in sip.conf to "busylevel" to be more parallel with the SIPPEER() argument of the same name. The deprecation procedure is not being used here since this is a trunk-only option. (closes issue #11307, reported by pj, patched by me) 2007-11-19 23:03 +0000 [r89439-89440] Russell Bryant * include/asterisk/module.h: Be a bit more pedantic about the type for holding the md5 sum for the build options. Also, doxygenify the comment. * funcs/func_sysinfo.c: Make the SYSINFO documentation reflect which options were compiled in 2007-11-19 22:55 +0000 [r89438] Steve Murphy * main/pbx.c: These changes were made in response to niklas@tese.se's letter of 11-17-2007, where he had 20 and 201 in two different contexts, included in the same context. In that particular case, we were behaving the same as 1.4, but after experimenting, I quickly found that if 20 and 201 were in the same extension, 1.4 would return 201, and this code returns 20. These changes now enable the current code to replicate the behavior of 1.4 in respect to MATCHMORE in cases like this. 2007-11-19 21:18 +0000 [r89430-89433] Luigi Rizzo * channels/chan_vpb.cc, channels/misdn_config.c, main/dsp.c: another few errno.h removals * pbx/pbx_loopback.c, apps/app_zapbarge.c, pbx/pbx_spool.c, apps/app_meetme.c, pbx/pbx_ael.c, pbx/pbx_lua.c, pbx/pbx_realtime.c, pbx/pbx_dundi.c, apps/app_externalivr.c, apps/app_directory.c, apps/app_system.c, pbx/pbx_config.c, apps/app_milliwatt.c: more errno.h removal * funcs/func_sysinfo.c: remove unnecessary headers * funcs/func_base64.c, funcs/func_volume.c: remove some unnecessary includes. 2007-11-19 20:13 +0000 [r89429] Tilghman Lesher * channels/chan_sip.c: Change delimiter of SIPPEER to be comma (instead of pipe) and further deprecate the old ':' delimiter Reported by: pj Patch by: tilghman Closes issue #11305 2007-11-19 19:51 +0000 [r89424-89428] Luigi Rizzo * codecs/codec_lpc10.c, codecs/codec_a_mu.c, codecs/codec_g722.c, codecs/codec_adpcm.c, codecs/codec_alaw.c, codecs/codec_speex.c, codecs/codec_g726.c, codecs/codec_gsm.c, codecs/codec_ulaw.c, codecs/codec_ilbc.c, codecs/codec_zap.c: remove some useless includes from codecs * formats/format_ilbc.c, formats/format_sln.c, formats/format_vox.c, formats/format_wav.c, formats/format_pcm.c, formats/format_ogg_vorbis.c, formats/format_g723.c, formats/format_h263.c, formats/format_h264.c, formats/format_wav_gsm.c, formats/format_g726.c, formats/format_jpeg.c, formats/format_gsm.c, formats/format_g729.c: format handlers don't need network, lock, channel and scheduler headers * include/asterisk.h, include/asterisk/compat.h, include/asterisk/lock.h, utils/extconf.c, include/asterisk/abstract_jb.h: move the declaration of struct ast_channel ast_frame and ast_module to compat.h so it is always available - hopefully this will let us reduce the number of inclusions of channel.h and frame.h * main/udptl.c, main/autoservice.c, funcs/func_rand.c, cdr/cdr_sqlite3_custom.c, main/frame.c, funcs/func_module.c, main/threadstorage.c, main/say.c, funcs/func_env.c, funcs/func_strings.c, main/devicestate.c, cdr/cdr_adaptive_odbc.c, main/indications.c, main/config.c, main/loader.c, main/term.c, main/cli.c, funcs/func_shell.c, main/http.c, cdr/cdr_odbc.c, main/db.c, cdr/cdr_manager.c, main/sched.c, main/pbx.c, funcs/func_timeout.c, funcs/func_math.c, funcs/func_cut.c, main/chanvars.c, main/netsock.c, funcs/func_curl.c, main/srv.c, main/privacy.c, funcs/func_cdr.c, funcs/func_channel.c, main/audiohook.c, funcs/func_iconv.c, main/alaw.c, main/asterisk.c, funcs/func_base64.c, funcs/func_md5.c, funcs/func_sysinfo.c, main/utils.c, funcs/func_sha1.c, cdr/cdr_pgsql.c, funcs/func_logic.c, cdr/cdr_radius.c, main/enum.c, funcs/func_uri.c, main/io.c, cdr/cdr_csv.c, main/ulaw.c, main/channel.c, main/cdr.c, funcs/func_enum.c, main/dial.c, funcs/func_groupcount.c, main/manager.c, main/tdd.c, funcs/func_odbc.c, cdr/cdr_sqlite.c, main/logger.c, main/app.c, main/image.c, main/dns.c, cdr/cdr_custom.c, funcs/func_version.c, funcs/func_db.c, main/dnsmgr.c, main/translate.c, main/slinfactory.c, funcs/func_lock.c, main/acl.c, main/rtp.c, cdr/cdr_tds.c, funcs/func_realtime.c, main/hashtab.c, funcs/func_blacklist.c, main/abstract_jb.c, main/cryptostub.c, main/adsistub.c, main/file.c, main/callerid.c, main/astmm.c, funcs/func_callerid.c, main/dsp.c: another bunch of include removals (errno.h and asterisk/logger.h) * channels/chan_local.c, apps/app_record.c, apps/app_alarmreceiver.c, apps/app_chanisavail.c, apps/app_ices.c, apps/app_exec.c, channels/chan_iax2.c, channels/chan_skinny.c, formats/format_pcm.c, apps/app_dumpchan.c, apps/app_zapras.c, formats/format_h263.c, codecs/codec_g722.c, formats/format_wav.c, apps/app_softhangup.c, codecs/codec_g726.c, formats/format_ogg_vorbis.c, apps/app_morsecode.c, apps/app_talkdetect.c, apps/app_db.c, apps/app_speech_utils.c, apps/app_sendtext.c, formats/format_g726.c, apps/app_mixmonitor.c, res/res_odbc.c, apps/app_voicemail.c, channels/chan_vpb.cc, formats/format_sln.c, res/res_snmp.c, apps/app_dictate.c, apps/app_authenticate.c, apps/app_readexten.c, codecs/codec_gsm.c, apps/app_userevent.c, channels/chan_gtalk.c, res/res_jabber.c, apps/app_setcallerid.c, res/res_config_odbc.c, apps/app_osplookup.c, apps/app_mp3.c, apps/app_minivm.c, res/res_realtime.c, formats/format_h264.c, apps/app_directory.c, apps/app_rpt.c, channels/chan_mgcp.c, apps/app_adsiprog.c, codecs/codec_lpc10.c, res/res_config_pgsql.c, apps/app_read.c, channels/chan_sip.c, codecs/codec_alaw.c, res/res_adsi.c, res/res_crypto.c, channels/chan_jingle.c, apps/app_channelredirect.c, apps/app_forkcdr.c, formats/format_vox.c, apps/app_sms.c, formats/format_g723.c, apps/app_verbose.c, apps/app_stack.c, apps/app_readfile.c, res/res_features.c, codecs/codec_adpcm.c, apps/app_sayunixtime.c, apps/app_test.c, apps/app_image.c, formats/format_wav_gsm.c, res/res_smdi.c, include/asterisk/compat.h, apps/app_skel.c, apps/app_zapscan.c, channels/chan_alsa.c, apps/app_url.c, apps/app_externalivr.c, formats/format_jpeg.c, formats/format_gsm.c, apps/app_milliwatt.c, apps/app_dial.c, apps/app_page.c, apps/app_privacy.c, codecs/codec_speex.c, apps/app_echo.c, channels/chan_agent.c, apps/app_disa.c, channels/iax2-provision.c, res/res_ael_share.c, apps/app_transfer.c, res/res_monitor.c, apps/app_playback.c, channels/chan_misdn.c, apps/app_waitforring.c, apps/app_zapbarge.c, channels/chan_features.c, apps/app_macro.c, apps/app_zapateller.c, res/res_indications.c, codecs/codec_ilbc.c, apps/app_chanspy.c, channels/chan_unistim.c, apps/app_meetme.c, res/res_musiconhold.c, apps/app_followme.c, codecs/codec_zap.c, res/res_config_sqlite.c, channels/misdn_config.c, apps/app_controlplayback.c, formats/format_ilbc.c, channels/chan_phone.c, res/res_agi.c, main/logger.c, apps/app_ivrdemo.c, apps/app_parkandannounce.c, res/res_clioriginate.c, apps/app_while.c, include/asterisk.h, apps/app_nbscat.c, channels/chan_zap.c, codecs/codec_a_mu.c, res/res_limit.c, apps/app_festival.c, apps/app_waitforsilence.c, res/res_convert.c, apps/app_getcpeid.c, apps/app_system.c, apps/app_queue.c, channels/chan_oss.c, channels/chan_usbradio.c, apps/app_flash.c, apps/app_directed_pickup.c, channels/chan_h323.c, codecs/codec_ulaw.c, channels/chan_nbs.c, apps/app_senddtmf.c, formats/format_g729.c: include "logger.h" and errno.h from asterisk.h - usage shows that they were included almost everywhere. Remove some of the instances. 2007-11-19 17:18 +0000 [r89422] Steve Murphy * main/pbx.c: a correction to code involved in an extension removal 2007-11-19 16:29 +0000 [r89421] Mark Michelson * funcs/func_sysinfo.c (added), CHANGES: Adding SYSINFO() dialplan function for retrieval of system information 2007-11-19 15:55 +0000 [r89417-89420] Joshua Colp * /, res/res_features.c: Merged revisions 89419 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r89419 | file | 2007-11-19 11:53:32 -0400 (Mon, 19 Nov 2007) | 6 lines Print out the correct filename (features.conf) in the log message when parkpos options are incorrect. (closes issue #11295) Reported by: Laureano Patches: res_features.c.patch uploaded by Laureano (license 265) ........ * /, doc/tex/localchannel.tex: Merged revisions 89416 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r89416 | file | 2007-11-19 11:24:12 -0400 (Mon, 19 Nov 2007) | 4 lines Clarify documentation a bit, include that a frame has to pass through the core in order for the Local channel optimization to happen. (closes issue #11246) Reported by: jon ........ 2007-11-19 14:36 +0000 [r89412] Luigi Rizzo * include/asterisk/logger.h: revert inclusion of options.h 2007-11-19 14:03 +0000 [r89410] Joshua Colp * apps/app_playback.c: Change warning messages (which are really debug messages) into debug messages. (closes issue #11288) Reported by: IgorG Patches: saydebug-89394-1-trunk.patch uploaded by IgorG (license 20) 2007-11-19 09:16 +0000 [r89404-89407] Olle Johansson * CHANGES: Update CHANGES * channels/chan_sip.c: Adding busy-level to the SIP_PEER() dialplan function. With this, you can control the peer in the dialplan, so you avoid placing outbound calls when the device has reached busy-level. Reported by pj. Closes bug #11180 * main/acl.c: Add some debugging to the routines that finds our local IP address. Related to bug #9225 * channels/chan_sip.c: Make some notes about a problem I found with the OPTIONs handler while working with the bug tracker. Since we don't authenticate devices (peers/users) on OPTIONS we don't have the proper context set for the user/peer. However, we might not want to process an authentication for every OPTIONS, so we could have a config option for this, "optionsforceok" to always answer 200 OK on the request and not check device or destination, nor add a SDP. If Asterisk sends the OPTIONs request, it doesn't care about the reply. Some devices use OPTIONs to discover capabilities, since we should answer like an INVITE from the device and we need to support that properly too, which we don't today. So much to do :-) 2007-11-18 21:50 +0000 [r89394-89399] Joshua Colp * build_tools/make_buildopts_h: Add OSX into the logic that uses md5 instead of md5sum. * include/asterisk/compat.h: Use the easy way that rizzo mentioned, only include malloc.h on the Windows platform. * include/asterisk/compat.h: Revert last commit, apparently buildbot lied to me. * include/asterisk/compat.h: Change how we handle alloca to conform with how it is suggested in the autoconf manual for AC_FUNC_ALLOCA. FreeBSD 6 now builds again and no other platforms should be broken by this. * configure, configure.ac: Change autoconf logic a bit so it says what it is looking for in two instances where it didn't. * configure, include/asterisk/autoconfig.h.in, configure.ac, include/asterisk/lock.h, include/asterisk/network.h: Use autoconf logic to determine the presence of PTHREAD_RECURSIVE_MUTEX_INITIALIZER_NP and PTHREAD_MUTEX_RECURSIVE_NP. Enclose error message from network.h in " 2007-11-17 21:47 +0000 [r89393] Matthew Fredrickson * channels/chan_zap.c: Add SS7 Generic address support (#11156) 2007-11-17 19:29 +0000 [r89389-89392] Luigi Rizzo * include/asterisk/compat.h: if alloca.h is not present, try malloc.h * agi/Makefile: temporarily disable this target in mingw * Makefile: will i ever get precedences for windows right ? in the meantime, use a variable to ease enabling/disabling print subdirectories. * Makefile: reformulate dependencies in a more correct way 2007-11-17 17:46 +0000 [r89388] Steve Murphy * main/pbx.c, pbx/pbx_dundi.c: a quick fix to pbx_dundi.c to make it so it will compile. Hope I did the right thing. And some additions to removal of extens to take care of hashtab pointers in all cases. 2007-11-17 17:27 +0000 [r89363-89387] Luigi Rizzo * Makefile.moddir_rules, Makefile.rules: as discussed some time ago on the -dev list, create embedde object with a .eo suffix even if they are coming from .cc sources. This simplifies the handling in the build scripts. * include/asterisk/network.h: prefer socket.h over other variants (winsock etc.) * channels/chan_local.c, main/translate.c, channels/chan_features.c, main/http.c, main/config.c: trim more redundant headers * main/acl.c: remove unnecessary includes * main/udptl.c, main/dnsmgr.c, channels/chan_sip.c, main/acl.c, main/dns.c, main/rtp.c, main/netsock.c: fix breakage induced by previous mistake * Makefile: wrong variable, wrong order -> broken build. * include/asterisk/acl.h, include/asterisk/utils.h, include/asterisk/autoconfig.h.in, include/asterisk/rtp.h, configure.ac, main/acl.c, include/asterisk/netsock.h, main/utils.c, include/asterisk/manager.h, main/netsock.c, main/manager.c, res/res_agi.c, pbx/pbx_dundi.c, include/asterisk/udptl.h, include/asterisk/dnsmgr.h, main/asterisk.c: start using asterisk/network.h for network related headers. Also remove some unnecessary includes. * include/asterisk/network.h (added): wrapper for all generic network headers that have different names and locations on the various systems. * main/cygload.c: main is called main not amain! * main/Makefile: conditional targets for building the windows version * Makefile: support cygwin targets * Makefile.moddir_rules: and this is the last one to have asterisk compile (not run yet) natively under cygwin. * apps/app_sms.c: another cygwin compatibility fix. This one must be handled in a better way in configure, also for other architectures * utils/Makefile, main/Makefile, utils/extconf.c: more cygwin/mingw32 compatibility fixes * include/asterisk/channel.h: use autoconf results to conditionally compile timersub * include/asterisk/lock.h: compatibility fixes for cygwin * include/asterisk/compat.h: some version of flex produce code that wants __STDC_VERSION__ defined, but the compiler does not always define it. * Makefile: these linker flags apply to both cygwin and mingw32 * utils/hashtest2.c: add a return NULL to a function that is expected to return a value so compilers that don't understand that this code is NOTREACHED will not complain (the fault is not much on the compiler but on the declaration of pthread_exit on certain platforms) s/certain platform/cygwin/ if you are really curious * main/loader.c: define RTLD_LOCAL for platforms that don't have it. This is only to complete the build, clearly the linker behaviour will be completely different and likely to cause trouble in those cases. * channels/Makefile: filter out modules that do not compile under windows (this should be handled with the dependencies generated by configure and menuselect, but will be fixed later) * main/utils.c: netdb.h is used for gethostbyname, and it was not included in some platforms. * main/cygload.c (added): Loader for cygwin where asterisk is really a big dll (something like this is already in 1.2) * configure, include/asterisk/autoconfig.h.in, configure.ac: timersub is a macro not a function, so write the check in a way that detects both formats. 2007-11-17 06:34 +0000 [r89359-89362] Russell Bryant * pbx/pbx_lua.c: fix the build of pbx_lua * configure, include/asterisk/autoconfig.h.in, include/asterisk/compat.h, configure.ac, include/asterisk/io.h, include/asterisk/channel.h: Update the configure script check for sys/poll.h to also provide the result in include/asterisk/autoconfig.h. Also, move the conditional include of sys/poll.h or asterisk/poll-compat.h into asterisk/config.h instead of the two headers it existed in before. * build_tools/make_buildopts_h: actually let this compile, oops :( * build_tools/make_buildopts_h: Use the fix suggested by Tilghman on the -dev to make cutting up the BUILDSUM friendly to non-bash shells. I think this should work for BSD/mingw as well, but did not yet remove the switch statement. 2007-11-17 04:19 +0000 [r89348-89358] Luigi Rizzo * Makefile: linker flags for mingw32 * configure, include/asterisk/autoconfig.h.in, configure.ac: add detection for timersub() and winsock.h/winsock2.h * include/asterisk/endian.h: provide definitions for __LITTLE_ENDIAN and __BIG_ENDIAN if not present. * main/Makefile, include/asterisk/io.h, include/asterisk/channel.h: use poll as detected by configure * configure, configure.ac, makeopts.in: use autoconf to check for the existence of sys/poll.h * build_tools/make_buildopts_h: this script is run on the build system, not on the host. * Makefile.moddir_rules: compatibility fix for mingw32 * configure, include/asterisk/autoconfig.h.in, configure.ac, acinclude.m4, makeopts.in: acinclude.m4: add a function to help checking sdl-config, gtk-config and the like (this could be used for gtk and gtk2 as well) Other files: add tests for sdl, sdl_image and avcodec and regenerate configure and autoconfig.h.in * include/asterisk/autoconfig.h.in, configure.ac: add check for the presence of glob * channels/chan_jingle.c, channels/chan_unistim.c, funcs/func_enum.c, channels/chan_local.c, channels/chan_misdn.c, channels/chan_skinny.c, funcs/func_odbc.c, channels/chan_h323.c, utils/ael_main.c, cdr/cdr_pgsql.c, channels/chan_gtalk.c, apps/app_db.c, channels/chan_mgcp.c: more removal of duplicate #include lines * main/udptl.c, funcs/func_module.c, res/res_features.c, funcs/func_lock.c, res/res_adsi.c, funcs/func_strings.c, channels/chan_agent.c, pbx/dundi-parser.c, main/rtp.c, pbx/pbx_loopback.c, funcs/func_blacklist.c, channels/chan_features.c, apps/app_dumpchan.c, res/res_agi.c, main/logger.c, pbx/pbx_realtime.c, pbx/pbx_dundi.c, apps/app_rpt.c, main/asterisk.c, apps/app_parkandannounce.c: remove a bunch of duplicate includes Reproduce with grep -r #include . | grep -v .svn | grep -v Binary | sort | uniq -c | sort -nr 2007-11-16 23:44 +0000 [r89347] Terry Wilson * res/res_features.c: Fix broken parking dial-back 2007-11-16 23:33 +0000 [r89346] Steve Murphy * main/pbx.c: My goodness, haven't handled an extension deletion. Add code to ast_context_remove_extension2() to remove an extension from the trie. Done by marking it deleted. The scoreboard won't update for it any more. Also, a couple of calls to insert hashtab had a spurious ->exten, which was removed. 2007-11-16 23:28 +0000 [r89341-89345] Luigi Rizzo * include/asterisk/paths.h, include/asterisk.h: paths are already in include/asterisk/paths.h so don't duplicate them in include/asterisk.h * include/asterisk/utils.h, include/asterisk/lock.h: whitespace only change - adjust indentation and add some comments on the content of these two files. utils.h (which is included in over 150 files) contains a lot of unrelated functions which require the inclusion of a large number of other headers. At some point we should partition its content in a better way. 2007-11-16 21:23 +0000 [r89333-89338] Luigi Rizzo * include/asterisk/logger.h: logger.h does not need options.h * include/asterisk/utils.h, channels/chan_sip.c, include/asterisk/astobj.h, include/asterisk/compat.h, include/asterisk/channel.h, include/asterisk/strings.h, utils/extconf.c, include/asterisk/frame.h, include/asterisk/stringfields.h, include/asterisk/endian.h: remove redundant #include "asterisk/compat.h", but make sure that asterisk/compiler.h is included everywhere * main/acl.c, main/asterisk.c: remove duplicate headers. Properly check for netdb.h (there is actually tens of places to fix) * Makefile.rules: put back default optimization to -O6 (previously changed by mistake) * main/frame.c, main/threadstorage.c, apps/app_alarmreceiver.c, apps/app_ices.c, channels/chan_iax2.c, apps/app_exec.c, channels/chan_skinny.c, main/strcompat.c, pbx/pbx_ael.c, apps/app_zapras.c, formats/format_h263.c, cdr/cdr_odbc.c, include/asterisk/sha1.h, main/db.c, cdr/cdr_manager.c, main/pbx.c, funcs/func_timeout.c, formats/format_wav.c, apps/app_softhangup.c, codecs/codec_g726.c, funcs/func_cut.c, apps/app_talkdetect.c, apps/app_db.c, funcs/func_channel.c, main/privacy.c, funcs/func_iconv.c, pbx/pbx_config.c, main/asterisk.c, res/res_odbc.c, include/asterisk/stringfields.h, apps/app_voicemail.c, formats/format_sln.c, apps/app_authenticate.c, apps/app_readexten.c, apps/app_userevent.c, codecs/codec_gsm.c, Makefile.rules, apps/app_setcallerid.c, include/asterisk/astmm.h, res/res_config_odbc.c, apps/app_osplookup.c, funcs/func_odbc.c, apps/app_mp3.c, formats/format_h264.c, apps/app_directory.c, main/md5.c, res/res_config_pgsql.c, main/dnsmgr.c, funcs/func_version.c, channels/chan_sip.c, funcs/func_lock.c, res/res_crypto.c, include/asterisk/cli.h, channels/chan_jingle.c, apps/app_forkcdr.c, funcs/func_blacklist.c, main/abstract_jb.c, main/file.c, apps/app_sms.c, formats/format_g723.c, main/astmm.c, apps/app_stack.c, apps/app_verbose.c, main/dsp.c, main/udptl.c, main/autoservice.c, funcs/func_module.c, codecs/codec_adpcm.c, cdr/cdr_adaptive_odbc.c, main/devicestate.c, apps/app_image.c, formats/format_wav_gsm.c, main/indications.c, pbx/pbx_loopback.c, funcs/func_shell.c, include/asterisk/compat.h, apps/app_skel.c, main/plc.c, channels/chan_alsa.c, apps/app_externalivr.c, formats/format_gsm.c, apps/app_milliwatt.c, res/res_speech.c, main/sched.c, apps/app_dial.c, apps/app_page.c, apps/app_disa.c, channels/iax2-provision.c, res/res_monitor.c, main/netsock.c, apps/app_waitforring.c, main/fixedjitterbuf.c, include/asterisk/lock.h, apps/app_chanspy.c, apps/app_cdr.c, channels/chan_unistim.c, funcs/func_base64.c, funcs/func_md5.c, apps/app_meetme.c, main/sha1.c, funcs/func_vmcount.c, res/res_musiconhold.c, cdr/cdr_radius.c, apps/app_followme.c, res/res_config_sqlite.c, main/fskmodem.c, channels/misdn_config.c, apps/app_controlplayback.c, cdr/cdr_csv.c, formats/format_ilbc.c, main/cdr.c, channels/chan_phone.c, funcs/func_enum.c, main/dial.c, main/manager.c, funcs/func_groupcount.c, cdr/cdr_sqlite.c, main/logger.c, main/image.c, apps/app_ivrdemo.c, res/res_clioriginate.c, apps/app_nbscat.c, codecs/codec_a_mu.c, channels/chan_zap.c, main/slinfactory.c, res/res_convert.c, pbx/pbx_lua.c, apps/app_queue.c, apps/app_system.c, channels/chan_oss.c, cdr/cdr_tds.c, funcs/func_realtime.c, channels/chan_usbradio.c, main/hashtab.c, apps/app_flash.c, include/asterisk/strings.h, apps/app_senddtmf.c, funcs/func_callerid.c, include/asterisk/time.h, channels/chan_local.c, funcs/func_dialgroup.c, funcs/func_env.c, apps/app_record.c, funcs/func_strings.c, apps/app_chanisavail.c, pbx/pbx_spool.c, apps/app_dumpchan.c, formats/format_pcm.c, main/http.c, main/stdtime/localtime.c, codecs/codec_g722.c, apps/app_morsecode.c, formats/format_ogg_vorbis.c, channels/iax2-parser.c, apps/app_speech_utils.c, include/asterisk/logger.h, main/srv.c, apps/app_sendtext.c, funcs/func_cdr.c, include/asterisk/md5.h, utils/hashtest2.c, utils/ael_main.c, main/audiohook.c, apps/app_mixmonitor.c, formats/format_g726.c, channels/chan_vpb.cc, apps/app_dictate.c, channels/chan_gtalk.c, funcs/func_logic.c, cdr/cdr_pgsql.c, res/res_jabber.c, funcs/func_uri.c, main/io.c, include/asterisk/abstract_jb.h, main/channel.c, apps/app_minivm.c, res/res_realtime.c, main/dns.c, apps/app_rpt.c, channels/chan_mgcp.c, apps/app_adsiprog.c, codecs/codec_lpc10.c, apps/app_read.c, codecs/codec_alaw.c, res/res_adsi.c, include/asterisk/plc.h, apps/app_channelredirect.c, formats/format_vox.c, main/cryptostub.c, main/callerid.c, pbx/pbx_dundi.c, funcs/func_devstate.c, funcs/func_rand.c, apps/app_readfile.c, cdr/cdr_sqlite3_custom.c, main/say.c, res/res_features.c, apps/app_sayunixtime.c, apps/app_test.c, main/config.c, main/loader.c, main/term.c, main/cli.c, res/res_smdi.c, include/asterisk/astobj.h, apps/app_zapscan.c, apps/app_amd.c, pbx/pbx_realtime.c, apps/app_url.c, formats/format_jpeg.c, include/asterisk/utils.h, apps/app_privacy.c, codecs/codec_speex.c, apps/app_echo.c, channels/chan_agent.c, funcs/func_math.c, res/res_ael_share.c, pbx/dundi-parser.c, apps/app_transfer.c, include/asterisk/manager.h, apps/app_playback.c, main/chanvars.c, apps/app_zapbarge.c, channels/chan_misdn.c, funcs/func_curl.c, channels/chan_features.c, apps/app_macro.c, codecs/codec_ilbc.c, res/res_indications.c, apps/app_zapateller.c, main/dlfcn.c, include/asterisk/slinfactory.h, utils/hashtest.c, main/utils.c, funcs/func_sha1.c, codecs/codec_zap.c, main/enum.c, include/asterisk/file.h, main/tdd.c, funcs/func_volume.c, res/res_agi.c, main/app.c, apps/app_parkandannounce.c, cdr/cdr_custom.c, apps/app_while.c, funcs/func_db.c, res/res_limit.c, apps/app_festival.c, apps/app_waitforsilence.c, main/translate.c, include/asterisk/config.h, main/jitterbuf.c, main/acl.c, apps/app_getcpeid.c, funcs/func_global.c, main/rtp.c, funcs/func_extstate.c, apps/app_directed_pickup.c, main/adsistub.c, channels/chan_h323.c, codecs/codec_ulaw.c, main/event.c, channels/chan_nbs.c, pbx/pbx_gtkconsole.c, formats/format_g729.c: Start untangling header inclusion in a way that does not affect build times - tested, there is no measureable difference before and after this commit. In this change: use asterisk/compat.h to include a small set of system headers: inttypes.h, unistd.h, stddef.h, stddint.h, sys/types.h, stdarg.h, stdlib.h, alloca.h, stdio.h Where available, the inclusion is conditional on HAVE_FOO_H as determined by autoconf. Normally, source files should not include any of the above system headers, and instead use either "asterisk.h" or "asterisk/compat.h" which does it better. For the time being I have left alone second-level directories (main/db1-ast, etc.). 2007-11-16 19:51 +0000 [r89331-89332] Mark Michelson * main/manager.c: Fixing a problem pointed out by Qwell * main/manager.c: Added some locks that should have been around astman_send_error, at least according to the comments. (closes issue #11258, reported and patched by eliel) 2007-11-16 19:26 +0000 [r89329-89330] Steve Murphy * main/pbx.c: This corrects a hashtab removal, given a bad argument * main/pbx.c, res/res_features.c: This fixes a problem with pattern ranges; and corrects a situation in res_features, where an extension would be created with the name Zap/51, as an example. THe / is bad because it would tend to mean that the 51 is to be cid matched. 2007-11-16 18:48 +0000 [r89328] Luigi Rizzo * build_tools/make_buildopts_h: both md5sum and variable substitutions such as ${BUILDSUM:0:8} are not available in FreeBSD. For the time being, put in a workaround so we can build the system, and wait for the result of the discussion on whether we can store the md5 as a string rather than 4 ints (if so, we won't need more complex tricks with awk or sed for splitting the md5). 1.4 will be fixed when we decide the issue. 2007-11-16 17:11 +0000 [r89327] Mark Michelson * apps/app_voicemail.c: Adding confirmation playback when forwarding voicemail messages. This will attempt to play the name(s) of the person(s) to whom you are forwarding the message prior to prompting for prepending. If no name is found, the extension is read back verbatim. (closes issue #9046, reported and patched by jaroth) 2007-11-16 16:56 +0000 [r89326] Kevin P. Fleming * /, include/asterisk/module.h, build_tools/make_buildopts_h, main/loader.c: Merged revisions 89325 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r89325 | kpfleming | 2007-11-16 10:47:46 -0600 (Fri, 16 Nov 2007) | 4 lines To help combat problems where people build external modules (asterisk-addons or others) and then change the build options of the Asterisk build in a way that makes the incompatible without warning, this commit introduces an MD5 signature of the important build-time options and includes that signature into modules when they are built. When the loader loads one of these modules and notices the problem, it will emit a warning to console and refuse to initialize the module, as doing so could cause the system to be unstable or even crash. If you upgrade to this version of Asterisk, you must rebuild *all* of your modules that came from other sources before trying to run this version. If you are using Digium's G.729 binary codec module, you will need v33 or newer. ........ 2007-11-16 15:44 +0000 [r89324] Mark Michelson * /, apps/app_queue.c: Merged revisions 89323 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r89323 | mmichelson | 2007-11-16 09:28:22 -0600 (Fri, 16 Nov 2007) | 5 lines Make realtime queues accessible from the QUEUE_MEMBER_COUNT function. (closes issue #11271, reported and patched by atis, with small modifications from me) ........ 2007-11-16 10:07 +0000 [r89322] Luigi Rizzo * include/asterisk/config.h, main/config.c: add a small new function to retrieve variables from a config once we have a pointer to the category. 2007-11-16 10:06 +0000 [r89321] Christian Richter * channels/chan_misdn.c: fixed #10631, about one way audio. thanks IgorG again. 2007-11-16 09:51 +0000 [r89320] Luigi Rizzo * channels/chan_oss.c: move the inner part of config file parsing to a separate function, so it can be reused in the implementation of cli commands when they have a similar syntax. 2007-11-16 08:54 +0000 [r89319] Christian Richter * channels/chan_misdn.c: fixed compilation of chan_misdn, #11269, thanks IgorG. 2007-11-15 23:50 +0000 [r89299-89312] Tilghman Lesher * main/utils.c, include/asterisk/stringfields.h: If we're going to be passing a negative value for the size of a stringfield, in order to indicate something, then using an UNSIGNED parameter is bad, mmmmmkay? * Makefile, /: Merged revisions 89302 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r89302 | tilghman | 2007-11-15 12:37:38 -0600 (Thu, 15 Nov 2007) | 2 lines Start Asterisk in Debian at a more reasonable time (since zaptel is at level 20) ........ * /, channels/misdn/isdn_lib.c: Merged revisions 89301 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r89301 | tilghman | 2007-11-15 12:23:14 -0600 (Thu, 15 Nov 2007) | 2 lines Fix an uninitialized memory read found by valgrind ........ * apps/app_zapscan.c: Fix trunk breakage due to chan->lock being renamed. * /, channels/chan_iax2.c: Merged revisions 89298 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r89298 | tilghman | 2007-11-15 12:05:56 -0600 (Thu, 15 Nov 2007) | 5 lines Yet another memory corruption issue. Reported by: atis Patch by: tilghman Fixes issue #10923 ........ 2007-11-15 17:27 +0000 [r89297] Russell Bryant * /, apps/app_meetme.c: Merged revisions 89296 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r89296 | russell | 2007-11-15 11:19:28 -0600 (Thu, 15 Nov 2007) | 8 lines Update the SLAStation application to account for the case where the SLA thread has a call out to the station, but the user has pressed a line button to answer the call instead of picking up the handset. If they do, the phone sends out a new INVITE. So, the SLAStation app must check to see if it is picking up a ringing trunk, and ensure that the other stations stop ringing. (reported internally, patched by me, tested by mogorman) ........ 2007-11-15 16:50 +0000 [r89294-89295] Steve Murphy * main/pbx.c: Get rid of a previously missed ast_log call for debug, no longer nec. * main/pbx.c: Perhaps I went overboard on initializing things. I can remove unnecc. stuff later. A few bug fixes. Killing small bugs on the way to killing bigger ones. Removed locking on hashtabs; there's plenty of locks already being taken. A small bug in the root_tree hashtab compare func. 2007-11-15 16:20 +0000 [r89293] Luigi Rizzo * main/channel.c, apps/app_channelredirect.c, main/manager.c, res/res_features.c, apps/app_softhangup.c, include/asterisk/channel.h, include/asterisk/lock.h, apps/app_senddtmf.c: access channel locks through ast_channel_lock/unlock/trylock and not through ast_mutex primitives. To detect all occurrences, I have renamed the lock field in struct ast_channel so it is clear that it shouldn't be used directly. There are some uses in res/res_features.c (see details of the diff) that are error prone as they try and lock two channels without caring about the order (or without explaining why it is safe). 2007-11-15 15:39 +0000 [r89290-89291] Joshua Colp * UPGRADE.txt: Fix typo in UPGRADE.txt. 'increase' should have been used, not 'increasing'. * channels/chan_sip.c, channels/chan_h323.c, channels/misdn_config.c: And file said... let trunk build again! Accomplished by some more constification, and marking a function in chan_sip as purposely unused until it is fixed up. 2007-11-15 14:58 +0000 [r89287-89289] Mark Michelson * main/manager.c, /: Merged revisions 89288 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r89288 | mmichelson | 2007-11-15 08:57:28 -0600 (Thu, 15 Nov 2007) | 3 lines Undoing previous commit since I realize it was wrong ........ * main/manager.c, /: Merged revisions 89286 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r89286 | mmichelson | 2007-11-15 08:54:10 -0600 (Thu, 15 Nov 2007) | 4 lines Adding a missing mutex unlock. (closes issue 11256, reported and patched by ys) ........ 2007-11-15 12:21 +0000 [r89278-89285] Olle Johansson * channels/chan_sip.c: Always relying on the responses when crossing NAT's are not a good solution, it breaks communication. Rizzo - you need to implement a configuration option for this code. It's good, but maybe should be off by default. * /, channels/chan_sip.c: Merged revisions 89281 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r89281 | oej | 2007-11-15 12:26:22 +0100 (Tor, 15 Nov 2007) | 6 lines Don't send re-invites during pending INVITE transactions. Patch by one47 - thanks! Closes issue #9305 ........ * /, channels/chan_sip.c: Merged revisions 89280 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r89280 | oej | 2007-11-15 12:15:09 +0100 (Tor, 15 Nov 2007) | 5 lines Improve support for multipart messages. Code by gasparz, changes by me (mostly formatting). Thanks, gasparz! Closes issue #10947 ........ * channels/chan_sip.c: Exit early instead of deciding to exit after processing the message. * channels/chan_sip.c, configs/sip.conf.sample: Add support for application/dtmf SIP INFO dtmf handling. Yep, another way of handling DTMF in SIP. Totally undocumented, but implemented in enough devices so we have to support it. Code by sergee, small changes by oej. Closes issue #11049 2007-11-15 01:42 +0000 [r89277] Steve Murphy * main/pbx.c: Had trouble playing with parking; spent a long time trying to reason out MATCHMORE mode. made these updates and xfers on zaptel lines seem to work ok now 2007-11-15 00:01 +0000 [r89273-89276] Tilghman Lesher * /, main/app.c: Merged revisions 89275 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r89275 | tilghman | 2007-11-14 17:23:58 -0600 (Wed, 14 Nov 2007) | 5 lines When a recording ends with '#', we are improperly trimming an extra 200ms from the recording. Reported by: sim Patch by: tilghman Closes issue #11247 ........ * main/channel.c: Typo * main/channel.c: Add callerid to the Hangup manager event. Reported by: outtolunc Patch by: outtolunc Closes issue #11248 2007-11-14 18:05 +0000 [r89271-89272] Steve Murphy * main/pbx.c: Rescaled the weights of the patterns to give something more independent of pattern length; and make . less likely to win. Question: which should win for 14102241145-- _1xxxxxxx. or _XXXXXXXXXXX -- right now, the pure X pattern will win. * main/pbx.c: A further problem highlighted by 11233 has been resolved; a certain combination of patterns in a certain order, led to a malformed trie, due to a ptr not being initialized in the loop. Also, some tree printing prettifications. 2007-11-14 15:13 +0000 [r89269-89270] Tilghman Lesher * channels/chan_phone.c, channels/chan_zap.c, res/res_jabber.c, res/res_config_sqlite.c, main/config.c, res/res_odbc.c: One more typo in config.c; and missed conversions due to the constifying of ast_variable_new parameters * main/config.c: Typo 2007-11-14 13:18 +0000 [r89268] Luigi Rizzo * include/asterisk/acl.h, channels/chan_sip.c, include/asterisk/config.h, channels/chan_agent.c, res/res_adsi.c, main/acl.c, pbx/dundi-parser.c, apps/app_queue.c, channels/chan_iax2.c, main/enum.c, channels/chan_oss.c, apps/app_playback.c, main/config.c, pbx/dundi-parser.h, include/asterisk/abstract_jb.h, main/manager.c, channels/chan_skinny.c, apps/app_minivm.c, main/abstract_jb.c, main/logger.c, pbx/pbx_dundi.c, apps/app_directory.c, apps/app_voicemail.c: make the 'name' and 'value' fields in ast_variable const char * This prevents modifying the strings in the stored variables, and catched a few instances where this was actually done. Given the differences between trunk and 1.4 (and the fact that this is effectively an API change) it is better to fix 1.4 independently. These are chan_sip.c::sip_register() chan_skinny.c:: near line 2847 config.c:: near line 1774 logger.c::make_components() res_adsi.c:: near line 1049 I may have missed some instances for modules that do not build here. 2007-11-14 03:22 +0000 [r89263-89266] Russell Bryant * main/hashtab.c, include/asterisk/hashtab.h: Fix up various coding guidelines issues ... - handle memory allocation failures - add an ast_ prefix to a publicly exported function - put curly braces in the right places - add a bunch of spaces where they should be be used * res/res_clioriginate.c: - Use the ARRAY_LEN macro in a couple places - return errors from load_module / unload_module * apps/app_dial.c: Use BEGIN_OPTIONS / END_OPTIONS to make the syntax highlighting in my editor happy * apps/app_queue.c: Instead of reserving 800 bytes for periodic announcements, use an array of ast_str pointers and only alloate space for the strings as needed. 2007-11-14 01:16 +0000 [r89262] Joshua Colp * main/srv.c, /: Merged revisions 89260 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r89260 | file | 2007-11-13 21:15:12 -0400 (Tue, 13 Nov 2007) | 4 lines Return the proper value when the srv_callback function executes properly. (closes issue #11240) Reported by: jtodd ........ 2007-11-14 01:15 +0000 [r89261] Russell Bryant * apps/app_queue.c: Convert most of the strings in the call_queue struct to use stringfields. 2007-11-14 00:54 +0000 [r89259] Kevin P. Fleming * main/channel.c, main/pbx.c: use simpler technique for removing known entries from lists 2007-11-14 00:33 +0000 [r89258] Russell Bryant * main/image.c: - Simplify removing an item from a list - move a verbose message to after the item is added to the list - make use of the ARRAY_LEN macro in one spot 2007-11-13 23:43 +0000 [r89256-89257] Steve Murphy * main/pbx.c: This hopefully will fix the re-opened 11233. Hadn't covered the case of a context with no patterns. (blush) * main/pbx.c: closes issue #11233 -- where some fine points in the algorithm to build the tree needed to be corrected. Many thanks for the test case, jtodd 2007-11-13 21:01 +0000 [r89250-89253] Russell Bryant * include/asterisk/lock.h: This fixes a build error on my mac. It also works on my linux box. Let me know if it breaks any other platform ... * res/res_features.c: Fix a typo pointed out by outtolunc, thanks :) * channels/chan_sip.c: - Convert initialization of a struct to C99 style instead of GNU style - Fix a minor spelling error in a comment * res/res_features.c, CHANGES: Update the ParkedCall application to grab the first available parked call if no parked extension is provided as an argument. (closes issue #10803) Reported by: outtolunc Patches: res_features-parkedcall-any.diff4 uploaded by outtolunc (license 237) - modified by me to work a bit differently ... 2007-11-13 19:48 +0000 [r89249] Jason Parker * /, res/res_features.c: Merged revisions 89248 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 (closes issue #11237) ........ r89248 | qwell | 2007-11-13 13:47:45 -0600 (Tue, 13 Nov 2007) | 7 lines Revert change from revision 67064. It is documented behavior that if a parking extension already exists while using PARKINGEXTEN, dialplan execution will continue. If blind transferring to a Park with PARKINGEXTEN, you must keep this in mind, and handle the failure yourself. Issue 11237, reported by jon. ........ 2007-11-13 17:41 +0000 [r89247] Tilghman Lesher * /, channels/chan_sip.c: Merged revisions 89246 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r89246 | tilghman | 2007-11-13 11:34:11 -0600 (Tue, 13 Nov 2007) | 2 lines If we set a value for qualify, we should actually pay attention to it, instead of overriding the value ........ 2007-11-13 16:03 +0000 [r89242] Mark Michelson * /, apps/app_mixmonitor.c: Merged revisions 89241 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r89241 | mmichelson | 2007-11-13 10:02:02 -0600 (Tue, 13 Nov 2007) | 5 lines Reverting commit made in revision 89205 since it is unnecessary. Thanks to Kevin for pointing this out ........ 2007-11-13 14:03 +0000 [r89240] Tilghman Lesher * /, main/utils.c: Merged revisions 89239 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r89239 | tilghman | 2007-11-13 07:51:53 -0600 (Tue, 13 Nov 2007) | 4 lines Debugging is running into the 16-lock limit. Increase to avoid. (This define is only effective when debugging is turned on, so there's no effect for most installations.) ........ 2007-11-13 01:19 +0000 [r89206-89207] Mark Michelson * apps/app_mixmonitor.c: There is the potential to copy uninitialized memory into the mixmonitor->post_process string. This fix prevents that. * /, apps/app_mixmonitor.c: Merged revisions 89205 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r89205 | mmichelson | 2007-11-12 18:56:46 -0600 (Mon, 12 Nov 2007) | 5 lines Some sanity checking for MixMonitor. If only 1 argument is given, then the args.options and args.post_process strings are uninitialized and could contain garbage. This change handles this situation properly by only using arguments that we have parsed. ........ 2007-11-13 00:19 +0000 [r89202-89203] Jason Parker * Makefile: oops, somebody left out the directory here... * channels/chan_unistim.c, res/res_features.c, main/ast_expr2f.c, include/asterisk/config.h, res/res_convert.c, res/res_crypto.c, pbx/pbx_lua.c, include/asterisk/cli.h, include/asterisk/pbx.h, res/res_config_sqlite.c, res/res_monitor.c, include/asterisk/stringfields.h, res/res_clioriginate.c: Doxygen fixes. Also fix a common typo I kept seeing (arguement) in various files. Closes issue #11222, patch by snuffy (with arguement > argument by me). 2007-11-12 23:33 +0000 [r89196-89201] Steve Murphy * utils/hashtest.c: Don't forget the ASTERISK_VERSION for the sake of the mtx_prof stuff. * include/asterisk/hashtab.h: Thanks to snuffy for this doxygen update to hashtab.h; closes issue #11223 * main/hashtab.c, include/asterisk/hashtab.h: Thanks to snuff-work, who brought up that these fixes might need to be made. 2007-11-12 20:48 +0000 [r89195] Jason Parker * main/pbx.c, /: Merged revisions 89194 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r89194 | qwell | 2007-11-12 14:46:52 -0600 (Mon, 12 Nov 2007) | 1 line Fix a typo pointed out by De_Mon on #asterisk-dev ........ 2007-11-12 20:16 +0000 [r89190] Kevin P. Fleming * utils/Makefile, utils/hashtest.c: (closes issue #11221) Reported by: eliel Patches: utils.Makefile.patch uploaded by eliel (modified by me) (license 64) 2007-11-12 18:44 +0000 [r89186] Steve Murphy * main/pbx.c, pbx/pbx_realtime.c, pbx/pbx_dundi.c, funcs/func_logic.c, apps/app_exec.c, apps/app_queue.c, apps/app_mixmonitor.c, cdr/cdr_manager.c: Based on a note in asterisk-dev by Brian Capouch, I determined I too agressive in not initializing arrays passed to pbx_substitute_variables_xxxx; I reviewed the code (again) and hopefully found every possible spot where substitute_variables is called conditionally, and made sure the char array involved was set to a null string. 2007-11-12 17:44 +0000 [r89185] Tilghman Lesher * main/channel.c, /, channels/chan_sip.c: Merged revisions 89184 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r89184 | tilghman | 2007-11-12 11:29:17 -0600 (Mon, 12 Nov 2007) | 5 lines Fix two cases of memory corruption caused by background threads. Reported by: atis Patch by: tilghman Fixes issue #10923 ........ 2007-11-12 13:36 +0000 [r89178-89179] Christian Richter * channels/chan_misdn.c, /, configs/misdn.conf.sample: Merged revisions 89173 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r89173 | crichter | 2007-11-12 12:26:48 +0100 (Mo, 12 Nov 2007) | 1 line if we're NT and no number was dialed and overlapdial is set, we wait for the ISDN timeout instead of starting our own timer. added a comment for the misdn.conf.sample for the overlapdial config option. ........ * channels/misdn/isdn_lib_intern.h, channels/chan_misdn.c, /, channels/misdn/isdn_msg_parser.c, channels/misdn/isdn_lib.c: Merged revisions 89172 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r89172 | crichter | 2007-11-12 12:23:57 +0100 (Mo, 12 Nov 2007) | 1 line added restart all interfaces Restart_Indicator, to automatically send a RESTART after the L2 of a PTP Port comes up. Also fixed some places where we have send a RELEASE without need for it. ........ 2007-11-12 13:26 +0000 [r89177] Joshua Colp * channels/chan_unistim.c, utils/hashtest.c: Fix building on FreeBSD by including/not including some headers. (closes issue #11218) Reported by: ys Patches: trunk89169.diff uploaded by ys (license 281) 2007-11-12 13:22 +0000 [r89174-89176] Christian Richter * channels/chan_misdn.c, /, channels/misdn/isdn_lib.c: Merged revisions 89171 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r89171 | crichter | 2007-11-12 12:13:13 +0100 (Mo, 12 Nov 2007) | 1 line fixed a state/event issue with overlapdial=yes when no extension matched. removed the general sending of a RELEASE_COMPLETE when we receive a RELEASE, this is done by mISDNuser/mISDN. This makes it possible to use asterisk-1.4 with mISDN trunk, but requires users of mISDN/mISDNuser-1.1.X to upgrade to at least mISDNuser-1.1.6 (when using the NT mode at all) ........ * /, channels/misdn/isdn_lib.c: Merged revisions 89170 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r89170 | crichter | 2007-11-12 10:57:23 +0100 (Mo, 12 Nov 2007) | 1 line fixed the support for CW and therefore for the reject_cause option. ........ * channels/misdn/isdn_lib.h, channels/chan_misdn.c, /, channels/misdn/chan_misdn_config.h, configs/misdn.conf.sample, channels/misdn/isdn_lib.c, channels/misdn_config.c: Merged revisions 89169 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r89169 | crichter | 2007-11-12 10:45:36 +0100 (Mo, 12 Nov 2007) | 1 line aded ntkeepcalls option, to avoid droÃpping calls when the L2 goes down on a PTP link. There are some pbx which do turn off the L1 for a very short while and restart it immediately. normally T310 should be started and after 10 seconds or so the calls should be dropped, this is a simple fix wihtout this timer. ........ 2007-11-09 18:57 +0000 [r89130-89132] Jason Parker * configs/usbradio.conf.sample (added): Add usbradio.conf.sample from branches/1.4/configs - r84162. It was mistakenly deleted in 1.4 without ever being merged to trunk. Reported by eliel on #asterisk-dev. * cdr/cdr_sqlite3_custom.c, configs/cdr_sqlite3_custom.conf (removed), configs/cdr_sqlite3_custom.conf.sample (added): Fix a few potential deadlocks in cdr_sqlite3_custom. (also rename sample config to .sample) Closes issue #11208, patch by Laureano. 2007-11-09 16:00 +0000 [r89129] Steve Murphy * res/ael/pval.c, utils/Makefile, main/pbx.c, main/hashtab.c (added), main/Makefile, utils/hashtest.c (added), pbx/pbx_ael.c, include/asterisk/hashtab.h (added), main/config.c: This is the perhaps the biggest, boldest, most daring change I've ever committed to trunk. Forgive me in advance any disruption this may cause, and please, report any problems via the bugtracker. The upside is that this can speed up large dialplans by 20 times (or more). Context, extension, and priority matching are all fairly constant-time searches. I introduce here my hashtables (hashtabs), and a regression for them. I would have used the ast_obj2 tables, but mine are resizeable, and don't need the object destruction capability. The hashtab stuff is well tested and stable. I introduce a data structure, a trie, for extension pattern matching, in which knowledge of all patterns is accumulated, and all matches can be found via a single traversal of the tree. This is per-context. The trie is formed on the first lookup attempt, and stored in the context for future lookups. Destruction routines are in place for hashtabs and the pattern match trie. You can see the contents of the pattern match trie by using the 'dialplan show' cli command when 'core set debug' has been done to put it in debug mode. The pattern tree traversal only traverses those parts of the tree that are interesting. It uses a scoreboard sort of approach to find the best match. The speed of the traversal is more a function of the length of the pattern than the number of patterns in the tree. The tree also contains the CID matching patterns. See the source code comments for details on how everything works. I believe the approach general enough that any issues that might come up involving fine points in the pattern matching algorithm, can be solved by just tweaking things. We shall see. The current pattern matcher is fairly involved, and replicating every nuance of it is difficult. If you find and report problems, I will try to resolve than as quickly as I can. The trie and hashtabs are added to the existing context and exten structs, and none of the old machinery has been removed for the sake of the multitude of functions that use them. In the future, we can (maybe) weed out the linked lists and save some space. 2007-11-08 23:53 +0000 [r89124-89126] Jason Parker * /, main/say.c: Merged revisions 89125 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 (closes issue #11203) ........ r89125 | qwell | 2007-11-08 17:52:35 -0600 (Thu, 08 Nov 2007) | 4 lines Properly say the seconds here.. Issue 11203, fix described by vma. ........ * pbx/pbx_lua.c: Add check_hangup() method to pbx_lua, which can be used to check whether it is time to hangup a channel. Closes issue #11202, patch by mnicholson 2007-11-08 22:33 +0000 [r89122-89123] Mark Michelson * apps/app_voicemail.c: app_voicemail failed to build when compiling with IMAP_STORAGE Now it does not. * main/threadstorage.c: AST_LIST_REMOVE_CURRENT takes only one argument. Thanks to snuffy for pointing this out on IRC 2007-11-08 21:27 +0000 [r89121] Joshua Colp * funcs/func_env.c: Make func_env build again. 2007-11-08 21:01 +0000 [r89120] Mark Michelson * /, channels/chan_sip.c: Merged revisions 89119 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r89119 | mmichelson | 2007-11-08 15:00:08 -0600 (Thu, 08 Nov 2007) | 7 lines Rework of the commit I made yesterday to use the already built-in ast_uri_decode function as opposed to my home-rolled one. Also added comments. Thanks to oej for pointing me in the right direction ........ 2007-11-08 20:39 +0000 [r89118] Kevin P. Fleming * channels/chan_features.c: convert this code to a more efficient idiom 2007-11-08 18:49 +0000 [r89116-89117] Jason Parker * res/res_smdi.c: Change a warning to a notice. Issue #11195, patch by eliel * /, configs/cdr_adaptive_odbc.conf.sample, configs/res_odbc.conf.sample: Merged revisions 89115 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 (closes issue #11195) ........ r89115 | qwell | 2007-11-08 12:45:15 -0600 (Thu, 08 Nov 2007) | 4 lines Avoid warnings on load when using sample configuration files. Issue 11195, patch by eliel. ........ 2007-11-08 17:32 +0000 [r89113-89114] Tilghman Lesher * apps/app_readfile.c, funcs/func_env.c: Add the FILE() dialplan function and deprecate ReadFile. * channels/chan_features.c: Fix missed conversion to linkedlists macro change 2007-11-08 16:51 +0000 [r89112] Mark Michelson * /: Blocking changes from previous 1.4 commit 2007-11-08 09:21 +0000 [r89108-89110] Luigi Rizzo * apps/app_voicemail.c: use %f instead of %lf (the 'l' is ignored anyways). * main/audiohook.c: use %d and cast to int instead of %zd for size_t object, this helps portability. * channels/chan_unistim.c: initialize a variable to silence compiler. The type of warnings emitted depends on the optimization level, at the lower levels the compiler doesn't always understand what the programmer has in mind. In this case I could not understand it either. 2007-11-08 05:36 +0000 [r89106-89107] Kevin P. Fleming * main/srv.c, /: Merged revisions 89105 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r89105 | kpfleming | 2007-11-08 00:26:47 -0500 (Thu, 08 Nov 2007) | 2 lines fix a glaring bug in the new SRV record handling that would cause incorrect weight sorting ........ * main/autoservice.c, main/frame.c, apps/app_meetme.c, res/res_features.c, funcs/func_strings.c, main/devicestate.c, res/res_musiconhold.c, channels/chan_iax2.c, apps/app_followme.c, codecs/codec_zap.c, res/res_jabber.c, main/indications.c, main/astobj2.c, main/config.c, main/loader.c, main/cli.c, main/cdr.c, main/channel.c, main/manager.c, res/res_agi.c, main/logger.c, main/app.c, main/image.c, res/res_speech.c, main/sched.c, main/pbx.c, main/translate.c, res/res_crypto.c, channels/chan_agent.c, utils/astman.c, apps/app_queue.c, channels/iax2-parser.c, main/srv.c, include/asterisk/linkedlists.h, main/file.c, pbx/pbx_dundi.c, main/event.c, main/audiohook.c, res/res_odbc.c, main/asterisk.c, apps/app_voicemail.c: improve linked-list macros in two ways: - the *_CURRENT macros no longer need the list head pointer argument - add AST_LIST_MOVE_CURRENT to encapsulate the remove/add operation when moving entries between lists 2007-11-08 05:00 +0000 [r89104] Tilghman Lesher * /, doc/valgrind.txt: Merged revisions 89103 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r89103 | tilghman | 2007-11-07 22:55:19 -0600 (Wed, 07 Nov 2007) | 2 lines Typo ........ 2007-11-08 02:28 +0000 [r89096-89102] Joshua Colp * /, channels/chan_sip.c: Merged revisions 89101 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r89101 | file | 2007-11-07 22:26:48 -0400 (Wed, 07 Nov 2007) | 4 lines Do not add a sip: to the beginning of the To URI unless needed. (closes issue #10756) Reported by: goestelecom ........ * /, channels/chan_sip.c: Merged revisions 89099 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r89099 | file | 2007-11-07 21:28:56 -0400 (Wed, 07 Nov 2007) | 6 lines Improve the devicestate logic for multiple devices. If any are available then the extension is considered available. (closes issue #10164) Reported by: nic_bellamy Patches: sip-hinting-svn-branch-1.4.patch uploaded by nic (license 299) ........ * /, channels/chan_sip.c: Merged revisions 89097 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r89097 | file | 2007-11-07 21:11:25 -0400 (Wed, 07 Nov 2007) | 8 lines Add support for allowing one outgoing transaction. This means if a response comes back out of order chan_sip will still handle it. I dream of a chan_sip with real transaction support. (closes issue #10946) Reported by: flefoll (closes issue #10915) Reported by: ramonpeek (closes issue #9567) Reported by: atca_pres ........ * /, channels/chan_sip.c: Merged revisions 89095 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r89095 | file | 2007-11-07 19:53:25 -0400 (Wed, 07 Nov 2007) | 4 lines If callerid is configured in sip.conf use that for checking the presence of an extension in the dialplan. (closes issue #11185) Reported by: spditner ........ 2007-11-07 23:47 +0000 [r89094] Tilghman Lesher * /, apps/app_queue.c: Merged revisions 89093 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r89093 | tilghman | 2007-11-07 17:39:37 -0600 (Wed, 07 Nov 2007) | 7 lines The member refcount must be incremented, to avoid using it after deallocation. A huge thanks go to lvl- for patiently providing the necessary valgrind output that was necessary to finding this problem of memory corruption. Reported by: lvl- Patch by: tilghman Closes issue #11174 ........ 2007-11-07 23:18 +0000 [r89091-89092] Mark Michelson * apps/app_voicemail.c: If imapfolder has been specified in voicemail.conf, we should not connect to INBOX... ever. It may not exist. (closes issue #11151, reported by selsky, patched by me) * /, channels/chan_sip.c: Merged revisions 89090 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r89090 | mmichelson | 2007-11-07 16:40:35 -0600 (Wed, 07 Nov 2007) | 6 lines This patch makes it possible for SIP phones to dial extensions defined with '#' characters in extensions.conf AND maintain their escaped characters when forming URI's (closes issue #10681, reported by cahen, patched by me, code review by file) ........ 2007-11-07 22:09 +0000 [r89089] Steve Murphy * /, res/res_jabber.c, cdr/cdr_tds.c: Merged revisions 89088 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r89088 | murf | 2007-11-07 14:40:28 -0700 (Wed, 07 Nov 2007) | 1 line In response to 10578, I just ran 1.4 thru valgrind; some of the config leakage I've already fixed, but it doesn't hurt to double check. I found and fixed leaks in res_jabber, cdr_tds, pbx_ael. Nothing major, tho. ........ 2007-11-07 17:45 +0000 [r89086] Joshua Colp * channels/h323/ast_h323.cxx: Minor change so chan_h323 builds again. 2007-11-07 13:12 +0000 [r89082-89084] Luigi Rizzo * Makefile: remove enter/exit comments when handling subdirectory. If we really want them we can remove the --no-print-directory * main/loader.c: remove a debugging message which i forgot in. * Makefile: match changes in menuselect's Makefile 2007-11-07 04:21 +0000 [r89077-89081] Tilghman Lesher * apps/app_playback.c: Suppress erroneous warnings on load. Reported by: eliel Patch by: eliel Closes issue #11177 * /, configs/extensions.ael.sample: Merged revisions 89079 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r89079 | tilghman | 2007-11-06 22:07:49 -0600 (Tue, 06 Nov 2007) | 5 lines Suppress AEL warnings on load. Reported by: eliel Patch by: eliel Closes issue #11178 ........ * channels/chan_zap.c, configs/zapata.conf.sample: Provide the ability to directly manipulate the TON/NPI bits in the dialstring. Reported by: thetatag Patch by: thetatag/stevens/tilghman Closes issue #5331 * contrib/utils/eagi_proxy.c (added): Add contributed EAGI proxy, which provides FastAGI functionality for EAGI, while also buffering the audio stream. Reported by: devil_slayer Patch by: devil_slayer Closes issue #8921 2007-11-07 00:16 +0000 [r89076] Russell Bryant * main/astmm.c: Fix another CLI command so it doesn't run the real code when called for initialization. 2007-11-07 00:04 +0000 [r89075] Mark Michelson * doc/tex/imapstorage.tex: Adding documentation regarding imapfolder, imapgreetings, and greetingsfolder options in voicemail.conf (closes issue #11133, reported by selsky, patched by blitzrage) 2007-11-07 00:00 +0000 [r89073-89074] Russell Bryant * include/asterisk/agi.h, res/res_agi.c, CHANGES: Print out the channel name as a prefix to the "agi debug" output. This makes AGI debugging on busy systems much easier. (closes issue #10730) Reported by: junky Patches: agi_debug_chan.diff uploaded by junky (license 177) 20070923_10730.diff uploaded by mvanbaak (license 7) * apps/app_meetme.c, CHANGES: Added the ability to do "meetme concise" with the "meetme" CLI command. This extends the concise capabilities of this CLI command to include listing all conferences, instead of an addition to the other sub commands for the "meetme" command. (closes issue #11078) Reported by: jthomas Patches: meetme-concise.patch uploaded by jthomas (license 293) 2007-11-06 23:08 +0000 [r89072] Joshua Colp * main/pbx.c: Fix up some PBX logic that became broken. The code would exit prematurely when it should have been collecting more digits. (closes issue #11175) Reported by: pj 2007-11-06 22:51 +0000 [r89071] Tilghman Lesher * channels/chan_jingle.c, channels/chan_phone.c, codecs/codec_g722.c, main/frame.c, channels/chan_sip.c, channels/chan_skinny.c, main/translate.c, channels/chan_h323.c, main/file.c, channels/chan_gtalk.c, include/asterisk/frame.h, main/rtp.c, channels/chan_mgcp.c, include/asterisk/translate.h: Commit some cleanups to the format type code. - Remove the AST_FORMAT_MAX_* types, as these are consuming 3 out of our available 32 bits. - Add a native slin16 type, so that 16kHz codecs can translate without losing resolution. (This doesn't affect anything immediately, until another codec has wb support.) 2007-11-06 22:36 +0000 [r89070] Mark Michelson * configs/queues.conf.sample, CHANGES, apps/app_queue.c: Adding the queue strategy wrandom (closes issue #10942, reported and patched by julianjm, documentation changes by me) 2007-11-06 22:15 +0000 [r89069] Russell Bryant * apps/app_meetme.c, doc/tex/channelvariables.tex, CHANGES: Added the S() and L() options to the MeetMe application. These are pretty much identical to the S() and L() options to Dial(). They let you set timeouts for the conference, as well as have warning sounds played to let the caller know how much time is left, and when it is running out. (closes issue #8030) Reported by: areski Patches: meetme_timeout_timelimit_v2.patch uploaded by areski (license 29) 2007-11-06 22:05 +0000 [r89068] Mark Michelson * apps/app_queue.c: Added CLI and manager commands for changing a queue member's penalty (closes issue #9374, reported and initially patched by wuwu, intermediate patch by eliel, and final patch by me) 2007-11-06 22:01 +0000 [r89067] Matthew Fredrickson * channels/chan_zap.c: Add some more locking as well as API update for libss7 for new transport types 2007-11-06 21:08 +0000 [r89062] Steve Murphy * /, main/config.c: Merged revisions 89036 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r89036 | murf | 2007-11-06 10:52:50 -0700 (Tue, 06 Nov 2007) | 1 line closes issue #8786 - where the [catname](!) and [catname](othercat1,othercat2,...) notation gets dropped across a ConfigUpdate (or any other thing that would cause a config file to be written). While I was at it, I also cleaned up some of the destroy routines to free up comments, which was not being done. Made sure the new struct I introduced is also cleaned up properly at destruction time. My code handles multiple template inclusions. Many thanks to ssokol for his patch, which, while not literally used in the final merge, served as a foundation for the fix. ........ 2007-11-06 20:55 +0000 [r89057] Joshua Colp * main/channel.c: Remove native bridging check for DTMF based transfers. Thanks to the last batch of RTP changes it is no longer required for the media stream to go through Asterisk if DTMF is going over signalling. It will simply reinvite back as needed. (closes issue #11172) Reported by: ibc 2007-11-06 20:32 +0000 [r89055] Mark Michelson * res/res_features.c: Instead of trying to callback a local channel on a failed attended transfer, call the device that made the transfer instead. This makes for much smoother calling back when queues are involved. (closes issue #11155, reported by IPetrov) Tremendous thanks to Russell for pulling me out of my block I was having on this one 2007-11-06 20:22 +0000 [r89052-89054] Russell Bryant * /, res/res_musiconhold.c: Merged revisions 89053 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r89053 | russell | 2007-11-06 14:18:49 -0600 (Tue, 06 Nov 2007) | 3 lines Fix init_classes() so that classes that actually do have files loaded aren't treated as empty, and immediately destroyed ... ........ * main/astmm.c: Fix the memory show allocations CLI command so that it doesn't spew out all of the current memory allocations when you start Asterisk, when the command's handler gets called for initialization. 2007-11-06 19:40 +0000 [r89051] Steve Murphy * main/ast_expr2f.c, main/ast_expr2.fl: Hoping to avoid a crash in OSX for a problem blitzrage found 2007-11-06 19:23 +0000 [r89050] Olle Johansson * main/fskmodem.c: Formatting. Illegaly using some spare spaces from Russell's space-bucket. 2007-11-06 19:16 +0000 [r89049] Tilghman Lesher * /, include/asterisk/lock.h: Merged revisions 89045 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r89045 | tilghman | 2007-11-06 13:09:06 -0600 (Tue, 06 Nov 2007) | 2 lines We went to the trouble of creating a method of tracking failed trylocks, then never turned it on (oops). ........ 2007-11-06 19:10 +0000 [r89048] Olle Johansson * main/tdd.c, include/asterisk/tdd.h: Additional TDD changes (preparing for SIP changes - adding TDD support to SIP) 2007-11-06 19:10 +0000 [r89047] Jason Parker * /, codecs/codec_zap.c: Merged revisions 89046 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r89046 | qwell | 2007-11-06 13:09:30 -0600 (Tue, 06 Nov 2007) | 4 lines Correctly set the total number of channels from a zaptel transcoder board. SPD-49, patch by Matthew Nicholson. ........ 2007-11-06 19:04 +0000 [r89044] Mark Michelson * apps/app_readfile.c, res/res_features.c, apps/app_sayunixtime.c, apps/app_test.c, apps/app_chanisavail.c, res/res_musiconhold.c, apps/app_exec.c, apps/app_followme.c, apps/app_minivm.c, apps/app_mp3.c, apps/app_amd.c, apps/app_while.c, main/pbx.c, apps/app_nbscat.c, channels/chan_sip.c, apps/app_festival.c, apps/app_softhangup.c, apps/app_waitforsilence.c, channels/chan_agent.c, apps/app_morsecode.c, apps/app_getcpeid.c, apps/app_playback.c, res/res_monitor.c, apps/app_speech_utils.c, apps/app_forkcdr.c, apps/app_waitforring.c, apps/app_directed_pickup.c, apps/app_macro.c, apps/app_sms.c, res/res_indications.c, apps/app_chanspy.c, apps/app_mixmonitor.c, apps/app_stack.c: "show application " changes for clarity. (closes issue #11171, reported and patched by blitzrage) Many thanks! 2007-11-06 19:04 +0000 [r89043] Olle Johansson * /, main/tdd.c: Merged revisions 89042 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r89042 | oej | 2007-11-06 19:53:37 +0100 (Tis, 06 Nov 2007) | 2 lines Bug fixes to tdd support in zaptel. ........ (Small changes for trunk) 2007-11-06 18:44 +0000 [r89041] Jason Parker * channels/chan_jingle.c, include/asterisk/jabber.h, channels/chan_gtalk.c, res/res_jabber.c: Allow gtalk and jingle to use TLS connections again. Closes issue #9972 2007-11-06 18:23 +0000 [r89038] Russell Bryant * /, res/res_musiconhold.c: Merged revisions 89037 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r89037 | russell | 2007-11-06 12:20:07 -0600 (Tue, 06 Nov 2007) | 11 lines If someone were to delete the files used by an existing MOH class, and then issue a reload, further use of that class could result in a crash due to dividing by zero. This set of changes fixes up some places to prevent this from happening. (closes issue #10948) Reported by: jcomellas Patches: res_musiconhold_division_by_zero.patch uploaded by jcomellas (license 282) Additional changes added by me. ........ 2007-11-06 17:10 +0000 [r89034] Joshua Colp * /, channels/chan_sip.c: Merged revisions 89032 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r89032 | file | 2007-11-06 13:08:05 -0400 (Tue, 06 Nov 2007) | 4 lines Make it so that if a peer is determined to be unreachable using qualify their devicestate will report back unavailable. (closes issue #11006) Reported by: pj ........ 2007-11-06 17:05 +0000 [r89031] Luigi Rizzo * main/loader.c: Fix embedding of modules on FreeBSD: the constructor for the list of modules was run after the constructors for the embedded modules (which appended entries to the list). As a result, the list appeared empty when it was time to use it. On linux the order of execution of constructor was evidently different (it may depend on the ordering of modules in the ELF file). This is only a workaround - there may be other situations where the execution of constructors causes problems, so if we manage to find a more general solution this workaround can go away. 2007-11-06 16:29 +0000 [r88974-88995] Joshua Colp * channels/chan_zap.c, /, configs/zapata.conf.sample: Merged revisions 88994 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r88994 | file | 2007-11-06 12:24:56 -0400 (Tue, 06 Nov 2007) | 6 lines Fix improbable but possible memory leaks in chan_zap. (closes issue #11166) Reported by: eliel Patches: chan_zap.c.patch uploaded by eliel (license 64) ........ * channels/chan_agent.c: Update chan_agent documentation. Change a | to , as that is now the required way. (closes issue #11167) Reported by: eliel Patches: chan_agent.c.patch uploaded by eliel (license 64) 2007-11-06 15:01 +0000 [r88973] Tilghman Lesher * channels/chan_unistim.c, configure, include/asterisk/autoconfig.h.in, configure.ac: Set up detection of IP_PKTINFO in autoconf for chan_unistim 2007-11-06 14:17 +0000 [r88932-88937] Russell Bryant * channels/chan_unistim.c: convert uses of LOG_DEBUG to use ast_debug() * channels/chan_unistim.c, configs/unistim.conf.sample: Add jitterbuffer support to chan_unistim. (closes issue #11168) Reported by: IgorG Patches: unistimjb-88863-1.patch uploaded by IgorG (license 20) * main/pbx.c, /, channels/busy.h, channels/ringtone.h, include/asterisk/pbx.h: Merged revisions 88805 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r88805 | russell | 2007-11-05 16:07:54 -0600 (Mon, 05 Nov 2007) | 12 lines After seeing crashes related to channel variables, I went looking around at the ways that channel variables are handled. In general, they were not handled in a thread-safe way. The channel _must_ be locked when reading or writing from/to the channel variable list. What I have done to improve this situation is to make pbx_builtin_setvar_helper() and friends lock the channel when doing their thing. Asterisk API calls almost all lock the channel for you as necessary, but this family of functions did not. (closes issue #10923, reported by atis) (closes issue #11159, reported by 850t) ........ * /, include/asterisk/lock.h: Merged revisions 88931 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r88931 | russell | 2007-11-06 07:50:15 -0600 (Tue, 06 Nov 2007) | 8 lines Remove some checks to see if locks are initialized from the non-DEBUG_THREADS versions of the lock routines. These are incorrect for a number of reasons: - It breaks the build on mac. - If there is a problem with locks not getting initialized, then the proper fix is to find that place and fix the code so that it does get initialized. - If additional debug code is needed to help find the problem areas, then this type of things should _only_ be put in the DEBUG_THREADS wrappers. ........ 2007-11-06 08:17 +0000 [r88898-88913] Luigi Rizzo * channels/Makefile: explain that the host environment must be used to build gentone; Remove unset variables, they would be misleading. * Makefile: don't export variables that can be retrieved from makeopts in child subdirs 2007-11-06 02:53 +0000 [r88863] Kevin P. Fleming * /, include/asterisk/srv.h: Merged revisions 88862 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r88862 | kpfleming | 2007-11-05 20:52:05 -0600 (Mon, 05 Nov 2007) | 2 lines update comment to match the state of the code ........ 2007-11-05 23:31 +0000 [r88827] Mark Michelson * main/channel.c, /: Merged revisions 88826 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r88826 | mmichelson | 2007-11-05 17:29:29 -0600 (Mon, 05 Nov 2007) | 6 lines Reworked deadlock avoidance in __ast_read. Restored audio to callback agents. (closes issue #11071, reported by callguy, patched by me, tested by callguy and Ted Brown) ........ 2007-11-05 21:36 +0000 [r88770] Luigi Rizzo * Makefile, utils/Makefile: Move AUDIO_LIBS outside the top level Makefile. This too is used only in one place. 2007-11-05 21:35 +0000 [r88769] Russell Bryant * /, channels/chan_sip.c: Merged revisions 88768 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r88768 | russell | 2007-11-05 15:33:56 -0600 (Mon, 05 Nov 2007) | 8 lines When traversing the list of channel variables here in transmit_invite(), the asterisk channel must be locked, as this data may change at any time. (I have seen numerous reports of crashes related to the handling of channel variables. There are a couple of issues on the bug tracker related to it, but it has also been noted on IRC and mailing lists. So, I am finding and fixing some places where channel variables are handled improperly.) ........ 2007-11-05 21:27 +0000 [r88767] Luigi Rizzo * Makefile, main/Makefile: Move the last instance of AST_LIBS to the only place it is used, namely main/Makefile . I am unclear where decisions on the build environment (CFLAGS, LDFLAGS, LIBS and so on) should be made - right now they are split here and there. As a first step in cleaning up this situation, i am trying to at least collect all instances of each variable in one place. 2007-11-05 21:23 +0000 [r88766] Russell Bryant * /, channels/chan_sip.c: Merged revisions 88765 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r88765 | russell | 2007-11-05 15:21:39 -0600 (Mon, 05 Nov 2007) | 2 lines Fix up some indentation. ........ 2007-11-05 20:50 +0000 [r88764] Luigi Rizzo * Makefile.moddir_rules: comment out an unused variable. Remove it in a few days if no problems arise. 2007-11-05 20:44 +0000 [r88710-88740] Russell Bryant * main/srv.c, /, include/asterisk/srv.h: Merged revisions 88719 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r88719 | russell | 2007-11-05 14:40:01 -0600 (Mon, 05 Nov 2007) | 7 lines Merge changes from asterisk/team/kpfleming/SRV-priority-handling Previously, the SRV record support in Asterisk was broken. There was no guarantee on what record Asterisk would choose to actually use. This set of changes improves the situation by ensuring that Asterisk will choose the highest priority record. ........ * main/channel.c, /: Merged revisions 88709 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r88709 | russell | 2007-11-05 14:11:04 -0600 (Mon, 05 Nov 2007) | 20 lines Merge the last bit of changes from asterisk/team/russell/readq-1.4 The issue here is that the channel frame readq handling got broken when the code was converted to use the linked list macros. It caused corruption of the list head and tail pointers. So, I fixed up the usage of the linked list macros and in passing, simplified the code. I also documented what the code is doing, as it was a bit difficult to figure out at first. This bug showed itself with crashes showing messed up head/tail pointers for the readq. However, there are a couple of crashes that aren't quite as obvious, but I think may be related. So, if your bug gets closed by this commit, but you still have a problem, please reopen or create a new bug report. (closes issue #10936) (closes issue #10595) (closes issue #10368) (closes issue #11084) (closes issue #10040) (closes issue #10840) ........ 2007-11-05 19:22 +0000 [r88675] Luigi Rizzo * Makefile: Cleanup the installation of samples, avoiding repetitions. I am preserving the behaviour on *.adsi files, i.e. overwrite anything there without making a backup. However I am not sure that this is the intended behaviour. 2007-11-05 18:52 +0000 [r88673] Joshua Colp * /, channels/chan_sip.c: Merged revisions 88671 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r88671 | file | 2007-11-05 14:47:13 -0400 (Mon, 05 Nov 2007) | 7 lines If a SIP channel is put on hold multiple times do not keep incrementing the onHold value. (closes issue #11085) Reported by: francesco_r Tested by: blitzrage (closes issue #10474) Reported by: acennami ........ 2007-11-05 18:22 +0000 [r88653] Tilghman Lesher * CHANGES: Change wording to that suggested by MasterYoda 2007-11-05 18:00 +0000 [r88652] Luigi Rizzo * Makefile: simplify (hopefully) the printing of $(MAKE) in aligned output. 2007-11-05 17:52 +0000 [r88651] Russell Bryant * main/channel.c, /: Merged revisions 88624 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r88624 | russell | 2007-11-05 11:46:02 -0600 (Mon, 05 Nov 2007) | 5 lines Fix up datastore handling in ast_do_masquerade(). The code is intended to move any channel datastores from the old channel to the new one. However, it did not use the linked list macros properly to accomplish the task. The existing code would only work if there was only a single datastore on the old channel. ........ 2007-11-05 17:44 +0000 [r88587-88615] Luigi Rizzo * Makefile: print messages when entering/leaving a directory so we know where we are (sometimes it is obvious, sometimes it is not). * Makefile.moddir_rules: merge two rules with the same right hand; document a bit what is done here. 2007-11-05 17:21 +0000 [r88586] Jason Parker * /, channels/chan_sip.c: Merged revisions 88585 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 (closes issue #11163) ........ r88585 | qwell | 2007-11-05 11:19:41 -0600 (Mon, 05 Nov 2007) | 4 lines Make sure we destroy the config structure on configuration failure. Issue 11163, patch by eliel. ........ 2007-11-05 17:00 +0000 [r88584] Kevin P. Fleming * Makefile.rules: use a variable name that actually indicates what it is for 2007-11-05 16:41 +0000 [r88553] Luigi Rizzo * Makefile.rules: Put extra compiler flags into a variable so they are not repeated too many times. On passing, add some comments and fix indentation a bit. On passing, i suspect that the following pattern is wrong %.eoo: %.o but in case it will be fixed in a later commit. 2007-11-05 16:30 +0000 [r88540] Tilghman Lesher * /, res/res_odbc.c: Merged revisions 88539 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r88539 | tilghman | 2007-11-05 10:20:13 -0600 (Mon, 05 Nov 2007) | 4 lines Don't check used pooled connections for connection status, as it will cause issues for prepared queries. Reported by: Nick Gorham (via -dev list) Patch by: tilghman ........ 2007-11-05 15:15 +0000 [r88525] Luigi Rizzo * main/db.c: remove a cygwin-specific function remap that does not work. 2007-11-05 13:11 +0000 [r88510] Joshua Colp * channels/chan_unistim.c: Fix memory leaks and deadlocks in chan_unistim. (closes issue #11158) Reported by: eliel Patches: chan_unistim.c.patch uploaded by eliel (license 64) 2007-11-04 22:42 +0000 [r88454-88490] Luigi Rizzo * /: block merging of not-applicable patch * main/channel.c, main/pbx.c, apps/app_meetme.c, channels/chan_sip.c, res/res_features.c, main/utils.c, channels/chan_iax2.c, include/asterisk/stringfields.h: Simplify the implementation and the API for stringfields; details and examples are in include/asterisk/stringfields.h. Not applicable to older branches except for 1.4 which will receive a fix for the routines that free memory pools. 2007-11-03 14:19 +0000 [r88437] Tilghman Lesher * main/term.c: Revert commit #86119. Some users intentionally do not want colorized terminals, so this was a misfeature. 2007-11-03 04:55 +0000 [r88422] James Golovich * main/db.c: Set CLI command to the correct name. Rev 85460 introduced two 'database show' commands when this one should have been 'database showkey' 2007-11-02 22:36 +0000 [r88368-88409] Russell Bryant * channels/chan_unistim.c: fix some issues with crashing on unload, when it didn't completely load cleanly * channels/chan_unistim.c: Convert the CLI commands to the new format * pbx/pbx_lua.c: propagate the DECLINE return value back to the loader * pbx/pbx_lua.c: Don't kill asterisk if extensions.lua is not present. * main/cli.c: Show the channel unique ID in the "show channel concise" output (closes issue #11148, requested by falves11, patched by me) * channels/chan_unistim.c (added), CREDITS, configs/unistim.conf.sample (added), CHANGES, doc/unistim.txt (added): Merge the code from asterisk/team/group/chan_unistim: This introduces a new channel driver, chan_unistim, that supports the Unistim VoIP protocol for Nortel phones. The following models have been confirmed to work: i2002, i2004 and i2050. (closes issue #8864) Reported by: c_hans Patches: chan_unistim.patch uploaded by c (license 304) ustm_no_conf.diff uploaded by junky (license 177) Tested by: c_hans, dbowerman, math, junky, loloski 2007-11-02 20:51 +0000 [r88329-88367] Joshua Colp * /, channels/chan_sip.c: Merged revisions 88366 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r88366 | file | 2007-11-02 17:49:45 -0300 (Fri, 02 Nov 2007) | 4 lines Make subscribecontext behave as advertised. It will now look for the presence of a hint in the given context (be it subscribecontext or context). (closes issue #10702) Reported by: slavon ........ * /, channels/chan_sip.c: Merged revisions 88328 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r88328 | file | 2007-11-02 17:20:21 -0300 (Fri, 02 Nov 2007) | 6 lines If an INFO request within a dialog is received with a content length of 0 simply send back a 200 OK. It is valid to do this and the remote side is probably using it to make sure the signalling is still alive. (closes issue #5747) Reported by: chandi Patches: infofix-81430-1.patch uploaded by IgorG (license 20) ........ 2007-11-02 20:13 +0000 [r88327] Russell Bryant * doc/tex/Makefile: Fix replacing the version number when it has a '/' in it, like SVN-group-chan_unistim-r88326M-/trunk 2007-11-02 17:34 +0000 [r88287] Tilghman Lesher * pbx/pbx_lua.c: Oops, some dev-mode changes for ISO C90 2007-11-02 16:54 +0000 [r88284] Jason Parker * /, main/say.c: Merged revisions 88283 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 (closes issue #11147) ........ r88283 | qwell | 2007-11-02 11:51:08 -0500 (Fri, 02 Nov 2007) | 4 lines We need to make sure to specify a language to ast_fileexists, otherwise it may fail for anything besides en Issue 11147, fix discovered by both citats and myself (independently), with input from Corydon76 ........ 2007-11-02 16:26 +0000 [r88209-88267] Tilghman Lesher * CHANGES: Add a few bytes on LUA * main/pbx.c, utils/build-extensions-conf.lua (added), build_tools/menuselect-deps.in, configure, include/asterisk/autoconfig.h.in, configure.ac, pbx/pbx_lua.c (added), configs/extensions.lua.sample (added), include/asterisk/pbx.h, makeopts.in: Add pbx_lua as a method of doing extensions Reported by: mnicholson Patch by: mnicholson Closes issue #11140 * main/config.c: Don't re-cache the filename, but check to see if it already exists Reported by: jamesgolovich Patch by: jamesgolovich Closes issue #11144 * /, include/asterisk/lock.h: Merged revisions 88210 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r88210 | tilghman | 2007-11-02 08:03:03 -0500 (Fri, 02 Nov 2007) | 5 lines Fix build on Solaris Reported by: snuffy Patch by: ys Closes issue #11143 ........ * main/pbx.c: 'h' extension doesn't execute past first priority Reported by: dimas Patch by: dimas Closes bug #11146 2007-11-02 03:09 +0000 [r88197] Joshua Colp * cdr/cdr_odbc.c: Restore building under 64-bit platforms. 2007-11-01 23:26 +0000 [r88184] Jason Parker * channels/chan_jingle.c, configure, include/asterisk/autoconfig.h.in, configure.ac, include/asterisk/jabber.h, channels/chan_gtalk.c, makeopts.in: Remove traces of gnutls, since we no longer use/need it. 2007-11-01 23:26 +0000 [r88182-88183] Tilghman Lesher * main/pbx.c: Modify WaitExten to include an optional dialtone Closes issue #10783 * UPGRADE.txt, cdr/cdr_odbc.c: Convert cdr_odbc to use res_odbc managed connections Closes issue #10614 2007-11-01 22:26 +0000 [r88166] Steve Murphy * apps/app_dial.c, main/pbx.c, cdr/cdr_sqlite3_custom.c, funcs/func_strings.c, funcs/func_cut.c, funcs/func_logic.c, apps/app_exec.c, apps/app_queue.c, apps/app_playback.c, res/ael/pval.c, pbx/pbx_loopback.c, funcs/func_odbc.c, apps/app_minivm.c, res/res_agi.c, main/logger.c, pbx/pbx_realtime.c, apps/app_macro.c, pbx/pbx_dundi.c, utils/extconf.c, include/asterisk/pbx.h, pbx/pbx_config.c, apps/app_mixmonitor.c, apps/app_rpt.c, cdr/cdr_custom.c, cdr/cdr_manager.c: This commits the performance mods that give the priority processing engine in the pbx, a 25-30% speed boost. The two updates used, are, first, to merge the ast_exists_extension() and the ast_spawn_extension() where they are called sequentially in a loop in the code, into a slightly upgraded version of ast_spawn_extension(), with a few extra args; and, second, I modified the substitute_variables_helper_full, so it zeroes out the byte after the evaluated string instead of demanding you pre-zero the buffer; I also went thru the code and removed the code that zeroed this buffer before every call to the substitute_variables_helper_full. The first fix provides about a 9% speedup, and the second the rest. These figures come from the 'PIPS' benchmark I describe in blogs, conf. reports, etc. 2007-11-01 22:19 +0000 [r88164-88165] Jason Parker * /: Crap, accidentally copied the props. Thanks for pointing this out mvanbaak. The odds are quite high that this will break automerge on every team branch. * /, include/asterisk/jabber.h, res/res_jabber.c: Switch res_jabber to use openssl rather than gnutls. Closes issue #9972, patch by phsultan. Copied from branch at http://svn.digium.com/svn/asterisk/team/phsultan/res_jabber-openssl/ 2007-11-01 17:25 +0000 [r88117] Tilghman Lesher * /, doc/valgrind.txt (added): Merged revisions 88116 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r88116 | tilghman | 2007-11-01 12:17:56 -0500 (Thu, 01 Nov 2007) | 2 lines Add some notes on using valgrind ........ 2007-11-01 16:22 +0000 [r88079] Jason Parker * channels/chan_zap.c, /: Merged revisions 88078 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r88078 | qwell | 2007-11-01 11:21:22 -0500 (Thu, 01 Nov 2007) | 4 lines Make sure we set the poll fds to NULL after free()ing it. Part of issue 11017, patch by tzafrir. ........ 2007-11-01 15:56 +0000 [r88062-88077] Russell Bryant * channels/chan_sip.c, pbx/pbx_dundi.c: Change some uses of free() to ast_free(). (No functional differences.) (closes issue #11138) Reported by: eliel Patches: pbx_dundi.c.patch uploaded by eliel (license 64) chan_sip.c.patch uploaded by eliel (license 64) * utils/Makefile: Remove another copied source file on "make clean". (closes issue #11137) Reported by: IgorG Patches: addonclean-87971-1.patch uploaded by IgorG (license 20) 2007-11-01 13:30 +0000 [r88027] Joshua Colp * /, apps/app_meetme.c: Merged revisions 88026 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r88026 | file | 2007-11-01 10:27:37 -0300 (Thu, 01 Nov 2007) | 2 lines Fix up commit for my Zap channel with spies in Meetme fix. (thanks Tony Mountifield!) ........ 2007-11-01 06:12 +0000 [r88007-88010] Tilghman Lesher * main/utils.c: Conditionally free lock_info->thread_name to avoid a useless warning Reported by: snuffy Patch by: snuffy Closes issue #11125 * apps/app_meetme.c, channels/chan_iax2.c: Janitor: use ast_free to pair calls of ast_malloc and ast_calloc Reported by: eliel Patch by: eliel Closes issue #11135 * cdr/cdr_adaptive_odbc.c: Fix memory leak Reported by: eliel Fixed by: tilghman Closes issue #11136 2007-11-01 01:55 +0000 [r87953-87971] Joshua Colp * /, apps/app_meetme.c: Merged revisions 87970 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r87970 | file | 2007-10-31 22:53:55 -0300 (Wed, 31 Oct 2007) | 4 lines If a Zap channel contains a spy or a spy is added take it out of the conference in kernel space and make it go through Asterisk so the spy gets audio from both sides. (closes issue #10060) Reported by: mparker ........ * main/pbx.c: Drop any more references to type in the Exception dialplan function. (closes issue #11134) Reported by: blitzrage Patches: exception_patch.txt uploaded by blitzrage (license 10) 2007-10-31 21:23 +0000 [r87889-87909] Jason Parker * /, res/res_jabber.c: Merged revisions 87908 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 (closes issue #11131) ........ r87908 | qwell | 2007-10-31 16:23:11 -0500 (Wed, 31 Oct 2007) | 4 lines Make sure we free some allocated memory before returning. Issue 11131, patch by eliel. ........ * channels/chan_jingle.c, /, channels/chan_gtalk.c: Merged revisions 87906 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 (closes issue #11130) (closes issue #11132) ........ r87906 | qwell | 2007-10-31 16:16:20 -0500 (Wed, 31 Oct 2007) | 4 lines Don't try to allocate memory that we're just going to re-allocate later anyways. Issues 11130 and 11132, patch by eliel. ........ * formats/format_sln.c, codecs/codec_adpcm.c, codecs/codec_gsm.c, formats/format_wav_gsm.c, res/res_musiconhold.c, codecs/codec_zap.c, formats/format_ilbc.c, res/res_smdi.c, formats/format_pcm.c, formats/format_h263.c, formats/format_h264.c, formats/format_jpeg.c, formats/format_gsm.c, res/res_speech.c, res/res_clioriginate.c, codecs/codec_lpc10.c, codecs/codec_g722.c, codecs/codec_a_mu.c, formats/format_wav.c, codecs/codec_speex.c, codecs/codec_alaw.c, res/res_adsi.c, res/res_convert.c, codecs/codec_g726.c, formats/format_ogg_vorbis.c, res/res_ael_share.c, formats/format_vox.c, codecs/codec_ulaw.c, formats/format_g723.c, res/res_indications.c, codecs/codec_ilbc.c, formats/format_g726.c, formats/format_g729.c: More changes to change return values from load_module functions. (issue #11096) Patches: codec_adpcm.c.patch uploaded by moy (license 222) codec_alaw.c.patch uploaded by moy (license 222) codec_a_mu.c.patch uploaded by moy (license 222) codec_g722.c.patch uploaded by moy (license 222) codec_g726.c.diff uploaded by moy (license 222) codec_gsm.c.patch uploaded by moy (license 222) codec_ilbc.c.patch uploaded by moy (license 222) codec_lpc10.c.patch uploaded by moy (license 222) codec_speex.c.patch uploaded by moy (license 222) codec_ulaw.c.patch uploaded by moy (license 222) codec_zap.c.patch uploaded by moy (license 222) format_g723.c.patch uploaded by moy (license 222) format_g726.c.patch uploaded by moy (license 222) format_g729.c.patch uploaded by moy (license 222) format_gsm.c.patch uploaded by moy (license 222) format_h263.c.patch uploaded by moy (license 222) format_h264.c.patch uploaded by moy (license 222) format_ilbc.c.patch uploaded by moy (license 222) format_jpeg.c.patch uploaded by moy (license 222) format_ogg_vorbis.c.patch uploaded by moy (license 222) format_pcm.c.patch uploaded by moy (license 222) format_sln.c.patch uploaded by moy (license 222) format_vox.c.patch uploaded by moy (license 222) format_wav.c.patch uploaded by moy (license 222) format_wav_gsm.c.patch uploaded by moy (license 222) res_adsi.c.patch uploaded by eliel (license 64) res_ael_share.c.patch uploaded by eliel (license 64) res_clioriginate.c.patch uploaded by eliel (license 64) res_convert.c.patch uploaded by eliel (license 64) res_indications.c.patch uploaded by eliel (license 64) res_musiconhold.c.patch uploaded by eliel (license 64) res_smdi.c.patch uploaded by eliel (license 64) res_speech.c.patch uploaded by eliel (license 64) 2007-10-31 18:53 +0000 [r87888] Steve Murphy * /: Merged revisions 87849 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r87849 | murf | 2007-10-31 11:49:39 -0600 (Wed, 31 Oct 2007) | 1 line closes issue #11108 -- where the 'dialplan save' cli command saves a file where the semicolon is not escaped. Fixed this; User also wanted comments to be preserved across dialplan save, but this is impossible at this point in time, because comments are not stored in the dialplan. They are 'compiled' out of extensions.conf. The only way to preserve those comments is to use the config file reader/writer that the GUI uses to allow online user edits. extensions.conf is first and foremost, a config file, and is read in by the normal config-file reading routines. Then, it is processed into a dialplan (context/exten structs). (in the case of trunk, tho, no mods needed to be made -- works OK there -- just make sure you use ',' to sep app args!) ........ 2007-10-31 18:09 +0000 [r87854] Tilghman Lesher * Makefile, /: Merged revisions 87852 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r87852 | tilghman | 2007-10-31 13:03:53 -0500 (Wed, 31 Oct 2007) | 2 lines Create samples for ALL of the available options in asterisk.conf ........ 2007-10-31 18:03 +0000 [r87833-87851] Joshua Colp * apps/app_mixmonitor.c: Add volume adjustment in. * apps/app_mixmonitor.c: Restore operation of the option that only writes when the channel is bridged. * apps/app_chanspy.c: Add volume adjustment to spy audiohook in app_chanspy. 2007-10-31 16:13 +0000 [r87817] Tilghman Lesher * CREDITS: Formatting cleanups, remove obsolete contributions (modules no longer in Asterisk), and obfuscate email addresses enough to stop most spam harvesters. 2007-10-31 16:07 +0000 [r87815] Joshua Colp * include/asterisk/channel.h: Remove old whisper remnants from channel.h 2007-10-31 15:46 +0000 [r87811] Tilghman Lesher * main/pbx.c: Optimize pbx_substitute_variables 2007-10-31 04:20 +0000 [r87776] Steve Murphy * res/ael/pval.c, /: Merged revisions 87775 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r87775 | murf | 2007-10-30 21:51:52 -0600 (Tue, 30 Oct 2007) | 1 line Included some verbage in the check_includes func, to inform the user that included contexts that have no match in the AEL, might be OK, as AEL cannot check in the extensions.conf or the in-memory contexts, as they may not be there at the time of the check. ........ 2007-10-30 23:08 +0000 [r87724-87740] Tilghman Lesher * /, include/asterisk/lock.h: Merged revisions 87739 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r87739 | tilghman | 2007-10-30 18:02:22 -0500 (Tue, 30 Oct 2007) | 5 lines Fix for uninitialized mutexes on *BSD Reported by: ys Fixed by: ys Closes issue #11116 ........ * apps/app_exec.c: If no '?' is found in the arguments, don't attempt to continue. Reported by: blitzrage Fixed by: tilghman Closes issue #11111 2007-10-30 21:22 +0000 [r87687] Russell Bryant * /, channels/chan_iax2.c: Merged revisions 87686 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r87686 | russell | 2007-10-30 16:19:09 -0500 (Tue, 30 Oct 2007) | 11 lines Merge the changes from team/russell/iax2_poke_fix and iax2-poke-fix-trunk There was a race condition related to the handling of POKEing peers. Essentially, a reference to a peer is held by the scheduler when there are pending callbacks, but the reference count didn't reflect it. So, it was possible for a peer to hit a reference count of zero and have its destructor begin to be called at the same time that the scheduler thread ran a POKE related callback. If that happened, a crash would likely occur. (closes issue #11082, closes issue #11094) ........ 2007-10-30 20:30 +0000 [r87626-87651] Jason Parker * /, channels/Makefile: Merged revisions 87650 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r87650 | qwell | 2007-10-30 15:29:41 -0500 (Tue, 30 Oct 2007) | 1 line Only try to clean out h323/ if the h323/Makefile exists. ........ * main/pbx.c: Update documentation to give an example of how to use the return status of RaiseException Closes issue #11117, patch by blitzrage (yay blitzrage) 2007-10-30 17:07 +0000 [r87573-87608] Mark Michelson * main/pbx.c: The priority gets incremented after raising an exception, so the priority should be set to 0 * main/pbx.c: Jumped the gun a bit in the RaiseException app. It would always return -1 since it checked for the existence of something that will never exist. 2007-10-30 16:15 +0000 [r87572] Joshua Colp * /, res/res_features.c: Merged revisions 87571 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r87571 | file | 2007-10-30 13:13:39 -0300 (Tue, 30 Oct 2007) | 4 lines Add two more checks before printing out a warning message about bridging. If either channel has hungup of course the bridge will have failed. (closes issue #10009) Reported by: dimas ........ 2007-10-30 15:47 +0000 [r87568] Jason Parker * /, main/editline/np/vis.c: Merged revisions 87567 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 (closes issue #11113) ........ r87567 | qwell | 2007-10-30 10:45:35 -0500 (Tue, 30 Oct 2007) | 4 lines Fix build of editline on Solaris. Issue 11113, patch by snuffy. ........ 2007-10-29 22:44 +0000 [r87462-87498] Kevin P. Fleming * utils/Makefile, utils, utils/hashtest2.c: UGH... while trying to fix #10995, I found all kinds of cruft in this Makefile. It should all be gone now, and as a side effect hashtest2 now builds with --enable-dev-mode enabled without a host of errors * agi/Makefile, utils/Makefile, codecs/g722/Makefile, main/editline/Makefile.in, Makefile.moddir_rules, codecs/ilbc/Makefile, codecs/lpc10/Makefile, main/db1-ast/Makefile: clean up assembler and preprocessor files if they are here too * utils, agi, codecs, apps, cdr, codecs/ilbc, formats, funcs, codecs/lpc10, main/db1-ast, codecs/g722, main/editline, main, codecs/gsm, main/minimime, pbx, res, channels: ignore preprocessor and assembler files if they are present * Makefile, /: Merged revisions 87460 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r87460 | kpfleming | 2007-10-29 17:04:29 -0500 (Mon, 29 Oct 2007) | 2 lines don't put '-pipe' into ASTCFLAGS if '-save-temps' is already there (used when debugging preprocessor issues) because the compiler will whine about each compile command ........ 2007-10-29 21:34 +0000 [r87397-87428] Russell Bryant * apps/app_meetme.c: If a caller is listen-only, then don't bother with doing talker detection. (closes issue #10911, reported by junky, patched by me) * /, main/utils.c, include/asterisk/lock.h: Merged revisions 87396 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r87396 | russell | 2007-10-29 15:22:07 -0500 (Mon, 29 Oct 2007) | 5 lines Add some more details to the output of "core show locks". When a thread is waiting for a lock, this will now show the details about who currently has it locked. (inspired by issue #11100) ........ 2007-10-29 20:13 +0000 [r87395] Mark Michelson * UPGRADE.txt, apps/app_queue.c: Adding the more flexible QUEUE_MEMBER function to replace the QUEUE_MEMBER_COUNT function. A deprecation notice will be issued the first time QUEUE_MEMBER_COUNT is used. 2007-10-29 20:02 +0000 [r87394] Joshua Colp * main/rtp.c: Drop the RTCP Read too short message to debug. There are some phones out there that send a sort of keep alive packet in the RTCP that trigger this every 5 seconds. 2007-10-29 19:56 +0000 [r87393] Jason Parker * apps/app_record.c: Make sure we set flags to a 0 value before trying to use it. Pointed out by seanbright while I was debugging issue 11109. 2007-10-29 19:47 +0000 [r87392] Russell Bryant * /, main/astmm.c: Merged revisions 87373 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r87373 | russell | 2007-10-29 14:21:06 -0500 (Mon, 29 Oct 2007) | 5 lines Remove a lock that doesn't make any sense. The regions lock needs to be held when traversing the list of allocated chunks so that they can be printed out to the CLI. (Thanks to eliel on #asterisk-dev for pointing this out!) ........ 2007-10-29 17:22 +0000 [r87343] Joshua Colp * /, channels/chan_sip.c: Merged revisions 87342 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r87342 | file | 2007-10-29 14:20:28 -0300 (Mon, 29 Oct 2007) | 6 lines Fix issue where if both sides of the dialog cancelled the dialog at the same time chan_sip could kepe retransmitting a response for no reason. (closes issue #9566) Reported by: atca_pres Patches: bug9566.patch uploaded by oej ........ 2007-10-29 16:38 +0000 [r87295-87327] Joshua Colp * apps/app_voicemail.c: Remove duplicate stdlib.h include. (closes issue #11105) Reported by: eliel Patches: app_voicemail.c.patch uploaded by eliel (license 64) * channels/chan_misdn.c, configure, include/asterisk/autoconfig.h.in, configure.ac: Add autoconf checks for extra suppserv definitions that are not present in releases yet. chan_misdn should now build against the latest release. (closes issue #11103) Reported by: IgorG * /, main/utils.c: Merged revisions 87294 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r87294 | file | 2007-10-29 11:23:49 -0300 (Mon, 29 Oct 2007) | 6 lines Fix issue with ast_unescape_semicolon going into an endless loop. (closes issue #10550) Reported by: ramonpeek Patches: unescape-85177-1.patch uploaded by IgorG (license 20) ........ 2007-10-28 14:16 +0000 [r87263-87264] Tilghman Lesher * funcs/func_dialgroup.c (added): Add a simple dialgroup function. By taking one of the simpler uses of Queue away from Queue, we simplify the lives of people who do not need all the bells and whistles. Also, this is part of the functions that people need to reimplement Queue in the dialplan, as a set of logic, rather than as a single app with hundreds of options. * /, funcs/func_odbc.c, funcs/func_strings.c, funcs/func_cut.c, funcs/func_realtime.c: Merged revisions 87262 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r87262 | tilghman | 2007-10-28 08:46:55 -0500 (Sun, 28 Oct 2007) | 7 lines Add autoservice to several more functions which might delay in their responses. Also, make sure that func_odbc functions have a channel on which to set variables. Reported by russell Fixed by tilghman Closes issue #11099 ........ 2007-10-27 15:41 +0000 [r87233-87247] Russell Bryant * configure, configure.ac: Update the configure script for the last libss7 API change * funcs/func_shell.c, funcs/func_lock.c: Make sure a channel exists before attempting to start or stop channel autoservice in func_lock and func_shell. 2007-10-27 00:48 +0000 [r87231-87232] Matthew Fredrickson * channels/chan_zap.c: Add Circuit Group Queury message code * channels/chan_zap.c: Make sure we turn on the DSP when we answer the call 2007-10-26 22:21 +0000 [r87217] Mark Michelson * CHANGES: Forgot to update CHANGES when I committed the linear queue strategy. Thank you Russell, for pointing this out! 2007-10-26 21:37 +0000 [r87202] Jason Parker * channels/chan_local.c, channels/chan_zap.c, channels/chan_agent.c, channels/chan_features.c, res/res_crypto.c, res/res_realtime.c, res/res_monitor.c: Correctly use defined return values in (some) load_module functions. (issue #11096) Patches: chan_agent.c.patch uploaded by eliel (license 64) chan_local.c.patch uploaded by eliel (license 64) chan_features.c.patch uploaded by eliel (license 64) chan_zap.c.patch uploaded by eliel (license 64) res_monitor.c.patch uploaded by eliel (license 64) res_realtime.c.patch uploaded by eliel (license 64) res_crypto.c.patch uploaded by eliel (license 64) 2007-10-26 17:39 +0000 [r87187] Steve Murphy * res/ael/pval.c, /, include/asterisk/pval.h, res/ael/ael.tab.c, res/ael/ael.y, pbx/pbx_ael.c, res/ael/ael_lex.c, res/ael/ael.tab.h, utils/ael_main.c, pbx/ael/ael-test/ref.ael-test16, res/ael/ael.flex, utils/conf2ael.c, pbx/ael/ael-test/ref.ael-test19: Merged revisions 87168 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r87168 | murf | 2007-10-26 10:34:02 -0600 (Fri, 26 Oct 2007) | 1 line closes issue #11086 where a user complains that references to following contexts report a problem; The problem was REALLy that he was referring to empty contexts, which were being ignored. Reporter stated that empty contexts should be OK. I checked it out against extensions.conf, and sure enough, empty contexts ARE ok. So, I removed the restriction from AEL. This, though, highlighted a problem with multiple contexts of the same name. This should be OK, also. So, I added the extend keyword to AEL, and it can preceed the 'context' keyword (mixed with 'abstract', if nec.). This will turn off the warnings in AEL if the same context name is used 2 or more times. Also, I now call ast_context_find_or_create for contexts now, instead of just ast_context_create; I did this because pbx_config does this. The 'extend' keyword thus becomes a statement of intent. AEL can now duplicate the behavior of pbx_config, ........ 2007-10-26 15:19 +0000 [r87153-87154] Mark Michelson * configs/queues.conf.sample, apps/app_queue.c: Added queue strategy "linear". This strategy is useful for those who always wish for their phones to be rung in a specific order. (closes issue #7279, reported and initially patched by diLLec, patch reworked by me) * configs/queues.conf.sample: Remove information about the roundrobin strategy from trunk's queues.conf.sample since it no longer exists 2007-10-26 14:00 +0000 [r87103-87121] Tilghman Lesher * funcs/func_curl.c, /: Merged revisions 87120 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r87120 | tilghman | 2007-10-26 08:54:30 -0500 (Fri, 26 Oct 2007) | 7 lines The addition of autoservice to func_curl additionally made func_curl dependent on the existence of a channel, with no real reason. This should make func_curl once again work without a channel. Reported by jmls. Fixed by tilghman. Closes issue #11090 ........ * include/asterisk/app.h, funcs/func_strings.c, funcs/func_cut.c, main/app.c: Use the same delimited character as the FILTER function in FIELDQTY and CUT. 2007-10-25 23:11 +0000 [r87070] Kevin P. Fleming * main/channel.c, /, include/asterisk/linkedlists.h: Merged revisions 87069 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r87069 | kpfleming | 2007-10-25 18:03:11 -0500 (Thu, 25 Oct 2007) | 2 lines appending one list to another should leave the first list empty, and not require the user to do that ........ 2007-10-25 18:59 +0000 [r87040] Russell Bryant * apps/app_meetme.c: Add support for a muted user to request to talk. The '2' option in the user menu will adjust this status if a user is muted. The talk request status will be reflected in the CLI commands as well as the manager interface. (closes issue #9418) Reported by: imesper Patches: app_meetme_v2.patch uploaded by imesper (license 275) 2007-10-25 16:21 +0000 [r87024] Steve Murphy * main/ast_expr2.y, res/res_config_sqlite.c, main/ast_expr2.c: closes issue #11045 - each file needs to define ASTERISK_FILE_VERSION, if you are going to set MTX_PROFILE in the compiler flags; the problem was that the fixes were getting made to the generated .c file, and erased the next time someone regenerated that file from the corresponding .y or .flex file. Moral of story: keep your eyes open and make mods to the .y (or flex input file) and re-run bison (or flex) as the Makefile directs for that file, and then check in both. Also, res_config_sqlite was kinda missed, and has the same issue. 2007-10-24 21:26 +0000 [r86985] Mark Michelson * configs/queues.conf.sample, apps/app_queue.c: Adding the general option "shared_lastcall" to queues so that a member's wrapuptime may be used across multiple queues. (closes issue #9777, reported and patched by eliel) 2007-10-24 20:59 +0000 [r86983] Jason Parker * channels/chan_zap.c, /: Merged revisions 86982 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 (closes issue #11079) ........ r86982 | qwell | 2007-10-24 15:56:47 -0500 (Wed, 24 Oct 2007) | 5 lines Correctly respect hidecalleridname configuration option. Simplify code slightly in the process. Issue 11079, reported by ddv2005 ........ 2007-10-24 13:21 +0000 [r86900-86967] Steve Murphy * pbx/ael/ael-test/ref.ael-ntest22, pbx/ael/ael-test/ref.ael-test2, pbx/ael/ael-test/ref.ael-test3, res/ael/ael_lex.c, pbx/ael/ael-test/ref.ael-test4, res/ael/ael.flex: closes issue #11005, where #include uses the current dir instead of the config dir (/etc/asterisk) for relative path includes for AEL * /, res/ael/ael.tab.c, res/ael/ael.y: Merged revisions 86936 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r86936 | murf | 2007-10-23 22:14:28 -0600 (Tue, 23 Oct 2007) | 1 line closes issue #11037 -- unable to specify app:spec in hint arguments ........ * /, funcs/func_logic.c: Merged revisions 86902 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r86902 | murf | 2007-10-23 15:18:08 -0600 (Tue, 23 Oct 2007) | 1 line closes issue #11052 -- where nothing after the ? will allow un-initialized variable values to corrupt and crash asterisk on 64-bit platforms ........ * /, main/ast_expr2f.c: Merged revisions 86880 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r86880 | murf | 2007-10-23 14:20:54 -0600 (Tue, 23 Oct 2007) | 1 line This should get rid of a really, really irritating warning generated by some 64-bit platforms from libc, where free(0) is frowned upon ........ * /, main/Makefile: Merged revisions 86881 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r86881 | murf | 2007-10-23 14:22:25 -0600 (Tue, 23 Oct 2007) | 1 line this update to Makefile corrects how ast_expr2f.c should be generated ........ 2007-10-22 21:37 +0000 [r86835-86839] Russell Bryant * /, include/asterisk/lock.h: Merged revisions 86836 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r86836 | russell | 2007-10-22 16:36:12 -0500 (Mon, 22 Oct 2007) | 9 lines If lock tracking is not enabled, then we can not attempt to log any mutex failures. If so, we could end up in infinite recursion. The only lock that is affected by this is a mutex in astmm.c used when MALLOC_DEBUG is enabled. (closes issue #11044) Reported by: ys Patches: lock.h.diff uploaded by ys (license 281) ........ * apps/app_playback.c: Convert some spaces to tabs and make it so the CLI command is only registered once instead of 3 times. (closes issue #11053) Reported by: seanbright Patches: app_playback.patch uploaded by seanbright (license 71) 2007-10-22 20:05 +0000 [r86820] Jason Parker * main/udptl.c, channels/chan_local.c, main/frame.c, res/res_features.c, main/threadstorage.c, channels/chan_iax2.c, main/astobj2.c, main/config.c, main/cli.c, channels/chan_skinny.c, main/http.c, pbx/pbx_ael.c, channels/chan_alsa.c, main/db.c, main/pbx.c, channels/chan_agent.c, channels/iax2-provision.c, apps/app_playback.c, channels/chan_misdn.c, channels/chan_features.c, res/res_indications.c, pbx/pbx_config.c, apps/app_mixmonitor.c, main/asterisk.c, res/res_odbc.c, apps/app_voicemail.c, apps/app_meetme.c, main/utils.c, channels/chan_gtalk.c, res/res_musiconhold.c, res/res_jabber.c, codecs/codec_zap.c, res/res_config_sqlite.c, main/channel.c, main/cdr.c, apps/app_osplookup.c, main/manager.c, res/res_agi.c, apps/app_minivm.c, main/logger.c, res/res_realtime.c, main/image.c, apps/app_rpt.c, channels/chan_mgcp.c, res/res_clioriginate.c, res/res_config_pgsql.c, main/dnsmgr.c, channels/chan_zap.c, channels/chan_sip.c, res/res_limit.c, main/translate.c, res/res_convert.c, res/res_crypto.c, include/asterisk/cli.h, apps/app_queue.c, channels/chan_oss.c, main/rtp.c, channels/chan_jingle.c, channels/chan_usbradio.c, main/file.c, channels/chan_h323.c, pbx/pbx_dundi.c, main/astmm.c, funcs/func_devstate.c: Switch from AST_CLI (formerly NEW_CLI) to AST_CLI_DEFINE, since the former didn't make much sense 2007-10-22 17:40 +0000 [r86790] Tilghman Lesher * /, main/astmm.c: Merged revisions 86787 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r86787 | tilghman | 2007-10-22 12:38:13 -0500 (Mon, 22 Oct 2007) | 2 lines Minor FreeBSD build fix ........ 2007-10-22 16:36 +0000 [r86755-86757] Joshua Colp * /, channels/chan_sip.c: Merged revisions 86756 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r86756 | file | 2007-10-22 13:35:22 -0300 (Mon, 22 Oct 2007) | 4 lines After reading online I have confirmed that Record-Route headers should be copied to 1xx responses as well. (closes issue #10113) Reported by: makoto ........ * /, apps/app_controlplayback.c: Merged revisions 86754 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r86754 | file | 2007-10-22 13:15:18 -0300 (Mon, 22 Oct 2007) | 4 lines Make sure res is a positive value before performing the check to determine whether the user stopped it or not. (closes issue #11023) Reported by: cfc ........ 2007-10-22 15:57 +0000 [r86734-86751] Russell Bryant * main/channel.c, /: Merged revisions 86750 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r86750 | russell | 2007-10-22 10:52:48 -0500 (Mon, 22 Oct 2007) | 8 lines Don't leak a frame in the case that an END frame is received and the time since the BEGIN is less than that of the defined minimum DTMF duration. (closes issue #11051) Reported by: casper Patches: channel.c.86664.diff uploaded by casper (license 55) ........ * channels/chan_zap.c: There is a really fun game that you can play before committing code, and it's called "make". :) * /, include/asterisk/lock.h: Merged revisions 86726 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r86726 | russell | 2007-10-22 10:43:30 -0500 (Mon, 22 Oct 2007) | 4 lines Update the static mutex initializer to include the initialization of the internal mutex used to protect the lock debugging data. (closes issue #11044, patch suggested by Ivan) ........ 2007-10-22 14:59 +0000 [r86697] Kevin P. Fleming * channels/chan_zap.c, configs/zapata.conf.sample: resetinterval defaulting to something other than 'never' doesn't seem to accomplish any good and causes problems for plenty of people... 2007-10-22 14:58 +0000 [r86696] Mark Michelson * /, apps/app_voicemail.c: Merged revisions 86694 via svnmerge from https://origsvn.digium.com/svn/aste