271 Commits (71ebf2e7ca7ed51a88d46a1ac58aed1945b97cd8)

Author	SHA1	Message	Date
Russell Bryant	a29672060a	Merged revisions 168562 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ................ r168562 | russell | 2009-01-13 13:22:13 -0600 (Tue, 13 Jan 2009) | 10 lines Merged revisions 168561 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r168561 | russell | 2009-01-13 13:13:05 -0600 (Tue, 13 Jan 2009) | 2 lines Revert unnecessary indications API change from rev 122314 ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@168565 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Steve Murphy	a7aeaf341b	Merged revisions 166665 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk This merged from trunk with no conflicts. I tested mostly the 'tired' cases, and for the most part ignored the tests for reconnecting and dialing in to fetch a parked call, after the first case. ................ r166665 | murf | 2008-12-23 11:13:49 -0700 (Tue, 23 Dec 2008) | 153 lines Merged revisions 166093 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 In order to merge this 1.4 patch into trunk, I had to resolve some conflicts and wait for Russell to make some changes to res_agi. I re-ran all the tests; 39 calls in all, and made fairly careful notes and comparisons: I don't want this to blow up some aspect of asterisk; I completely removed the KEEPALIVE from the pbx.h decls. The first 3 scenarios involving feature park; feature xfer to 700; hookflash park to Park() app call all behave the same, don't appear to leave hung channels, and no crashes. ........ r166093 | murf | 2008-12-19 15:30:32 -0700 (Fri, 19 Dec 2008) | 131 lines This merges the masqpark branch into 1.4 These changes eliminate the need for (and use of) the KEEPALIVE return code in res_features.c; There are other places that use this result code for similar purposes at a higher level, these appear to be left alone in 1.4, but attacked in trunk. The reason these changes are being made in 1.4, is that parking ends a channel's life, in some situations, and the code in the bridge (and some other places), was not checking the result code properly, and dereferencing the channel pointer, which could lead to memory corruption and crashes. Calling the masq_park function eliminates this danger in higher levels. A series of previous commits have replaced some parking calls with masq_park, but this patch puts them ALL to rest, (except one, purposely left alone because a masquerade is done anyway), and gets rid of the code that tests the KEEPALIVE result, and the NOHANGUP_PEER result codes. While bug 13820 inspired this work, this patch does not solve all the problems mentioned there. I have tested this patch (again) to make sure I have not introduced regressions. Crashes that occurred when a parked party hung up while the parking party was listening to the numbers of the parking stall being assigned, is eliminated. These are the cases where parking code may be activated: 1. Feature one touch (eg. *3) 2. Feature blind xfer to parking lot (eg ##700) 3. Run Park() app from dialplan (eg sip xfer to 700) (eg. dahdi hookflash xfer to 700) 4. Run Park via manager. The interesting testing cases for parking are: I. A calls B, A parks B a. B hangs up while A is getting the numbers announced. b. B hangs up after A gets the announcement, but before the parking time expires c. B waits, time expires, A is redialed, A answers, B and A are connected, after which, B hangs up. d. C picks up B while still in parking lot. II. A calls B, B parks A a. A hangs up while B is getting the numbers announced. b. A hangs up after B gets the announcement, but before the parking time expires c. A waits, time expires, B is redialed, B answers, A and B are connected, after which, A hangs up. d. C picks up A while still in parking lot. Testing this throroughly involves acting all the permutations of I and II, in situations 1,2,3, and 4. Since I added a few more changes (ALL references to KEEPALIVE in the bridge code eliimated (I missed one earlier), I retested most of the above cases, and no crashes. H-extension weirdness. Current h-extension execution is not completely correct for several of the cases. For the case where A calls B, and A parks B, the 'h' exten is run on A's channel as soon as the park is accomplished. This is expected behavior. But when A calls B, and B parks A, this will be current behavior: After B parks A, B is hung up by the system, and the 'h' (hangup) exten gets run, but the channel mentioned will be a derivative of A's... Thus, if A is DAHDI/1, and B is DAHDI/2, the h-extension will be run on channel Parked/DAHDI/1-1<ZOMBIE>, and the start/answer/end info will be those relating to Channel A. And, in the case where A is reconnected to B after the park time expires, when both parties hang up after the joyful reunion, no h-exten will be run at all. In the case where C picks up A from the parking lot, when either A or C hang up, the h-exten will be run for the C channel. CDR's are a separate issue, and not addressed here. As to WHY this strange behavior occurs, the answer lies in the procedure followed to accomplish handing over the channel to the parking manager thread. This procedure is called masquerading. In the process, a duplicate copy of the channel is created, and most of the active data is given to the new copy. The original channel gets its name changed to XXX<ZOMBIE> and keeps the PBX information for the sake of the original thread (preserving its role as a call originator, if it had this role to begin with), while the new channel is without this info and becomes a call target (a "peer"). In this case, the parking lot manager thread is handed the new (masqueraded) channel. It will not run an h-exten on the channel if it hangs up while in the parking lot. The h exten will be run on the original channel instead, in the original thread, after the bridge completes. See bug 13820 for our intentions as to how to clean up the h exten behavior. Review: http://reviewboard.digium.com/r/29/ ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@166730 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	52f5e8fcfb	Merged revisions 165723 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ........ r165723 | russell | 2008-12-18 13:33:42 -0600 (Thu, 18 Dec 2008) | 14 lines Remove the need for AST_PBX_KEEPALIVE with the GoSub option from Dial. This is part of an effort to completely remove AST_PBX_KEEPALIVE and other similar return codes from the source. While this usage was perfectly safe, there are others that are problematic. Since we know ahead of time that we do not want to PBX to destroy the channel, the PBX API has been changed so that information can be provided as an argument, instead, thus removing the need for the KEEPALIVE return value. Further changes to get rid of KEEPALIVE and related code is being done by murf. There is a patch up for that on review 29. Review: http://reviewboard.digium.com/r/98/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@165728 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Steve Murphy	e3e5457dc4	Merged revisions 164801 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ........ r164801 | murf | 2008-12-16 13:04:46 -0700 (Tue, 16 Dec 2008) | 36 lines (closes issue #14076) Reported by: toc Tested by: murf OK, Well this issue has had its share of flip-flopping. I found the following: 1. the code in question, in ext_cmp1 in pbx.c, would not allow two extensions that vary only by any dashes contained within them, to be defined in the same context. 2. for input dialstrings, dashes are NOT ignored. So, skipping them when sorting patterns seemed a bit silly. Thus, you might declare ext 891 in a context, but if you try dialing 8-9-1, it will NOT match 891. So, I proposed to remove the code from ext_cmp1 to skip the spaces and dashes. Just kept us from declaring 891 and 8-9-1 in the same context, forcing users to generate otherwise uselessly obfuscated dialplan code to get the same effect. Then, I tried out 1.4, and found that: 1. you can declare 891 and 8-9-1 in the same context! 2. You can't define 891, and have 8-9-1 match it! Nor can you define 8-9-1, and have 891 match it! So, it appears that my proposal simply restores the pbx to behaving as it did in 1.4. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@164805 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Steve Murphy	a13fe6807a	Merged revisions 164648 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ................ r164648 | murf | 2008-12-16 08:31:54 -0700 (Tue, 16 Dec 2008) | 13 lines Merged revisions 164634 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r164634 | murf | 2008-12-16 08:15:58 -0700 (Tue, 16 Dec 2008) | 5 lines I added a sentence to clarify why - and ' ' are ignored in patterns as per bug 14076. Leif says he'll put some stuff about it in the extensions.conf sample, etc. ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@164658 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	704f4a7d5c	Merged revisions 162922,162930 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ........ r162922 | tilghman | 2008-12-10 16:48:09 -0600 (Wed, 10 Dec 2008) | 7 lines Checking global variables here actually overwrote the previous substitution by channel variables, and in any case, was redundant; pbx_substitute_variables_helper ALREADY does substitution for global variables. (closes issue #13327) Reported by: pj ........ r162930 | tilghman | 2008-12-10 17:01:14 -0600 (Wed, 10 Dec 2008) | 2 lines Previously missing line, now the substitution works correctly ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@162949 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	b6b83bbe9c	Merged revisions 162266 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ................ r162266 | mmichelson | 2008-12-09 14:30:07 -0600 (Tue, 09 Dec 2008) | 14 lines Merged revisions 162265 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r162265 | mmichelson | 2008-12-09 14:28:44 -0600 (Tue, 09 Dec 2008) | 6 lines If we fail to start a thread for the pbx to run in, we need to be sure to decrease the number of active calls on the system. This fix may relate to ABE-1713, but it is not certain yet. ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@162269 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Joshua Colp	f1a71be36b	Merged revisions 161787 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ........ r161787 | file | 2008-12-08 14:47:32 -0400 (Mon, 08 Dec 2008) | 6 lines Fix a regression introduced when the PBX timeouts were converted to milliseconds. collect_digits now gets milliseconds fed to it, not seconds. (closes issue #14012) Reported by: dveiga Patches: 14012.patch uploaded by bkruse (license 132) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@161789 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	96e176f2a0	Merged revisions 161288 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ................ r161288 | russell | 2008-12-05 08:16:24 -0600 (Fri, 05 Dec 2008) | 10 lines Merged revisions 161287 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r161287 | russell | 2008-12-05 08:12:14 -0600 (Fri, 05 Dec 2008) | 2 lines Fix a NULL format string warning found by buildbot. ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@161290 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	6674ef2c22	Merged revisions 160208 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ................ r160208 | tilghman | 2008-12-01 18:37:21 -0600 (Mon, 01 Dec 2008) | 10 lines Merged revisions 160207 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r160207 | tilghman | 2008-12-01 18:25:16 -0600 (Mon, 01 Dec 2008) | 3 lines Ensure that Asterisk builds with --enable-dev-mode, even on the latest gcc and glibc. ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@160234 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	e05fd35dc6	Merged revisions 158602 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ................ r158602 | tilghman | 2008-11-21 17:14:11 -0600 (Fri, 21 Nov 2008) | 12 lines Merged revisions 158600 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r158600 | tilghman | 2008-11-21 17:07:46 -0600 (Fri, 21 Nov 2008) | 5 lines The passed extension may not be the same in the list as the current entry, because we strip spaces when copying the extension into the structure. Therefore, use the copied item to place the item into the list. (found by lmadsen on -dev, fixed by me) ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@158604 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Jeff Peeler	fe6767f2e1	Merged revisions 156649 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ........ r156649 | jpeeler | 2008-11-13 13:17:50 -0600 (Thu, 13 Nov 2008) | 6 lines (closes issue #13891) Reported by: smurfix This reverts a change I made in 116297. At the time it seemed the change was required to solve an issue with attempting a transfer but then letting it timeout without dialing any digits. However, I didn't realize that having an empty extension was possible. I'm removing the immediate return that was added in pbx_find_extension if the extension is null. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@156650 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Steve Murphy	475be50f76	Merged revisions 156299 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ................ r156299 | murf | 2008-11-12 12:47:29 -0700 (Wed, 12 Nov 2008) | 26 lines Merged revisions 156297 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r156297 | murf | 2008-11-12 12:36:16 -0700 (Wed, 12 Nov 2008) | 18 lines It turns out that the 0x0XX00 codes being returned for N, X, and Z are off by one, as per conversation with jsmith on #asterisk-dev; he was teaching a class and disconcerted that this published rule was not being followed, with patterns _NXX, _[1-8]22 and _[2-9]22... and NXX was winning, but [1-8] should have been. This change, tested on these 3 patterns now picks the proper one. However, this change may surprise users who set up dialplans based on previous behavior, which has been there for what, 2 and half years or so now. ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@156354 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	f34b1869bd	Merged revisions 152689 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ........ r152689 | tilghman | 2008-10-29 19:45:47 -0500 (Wed, 29 Oct 2008) | 2 lines Track down and fix annoying lock errors ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@152690 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Steve Murphy	c6fc68ab1a	Merged revisions 148519 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ........ r148519 | murf | 2008-10-13 11:14:38 -0600 (Mon, 13 Oct 2008) | 15 lines Hmmm. Nobody (but me) is interested in seeing the trie info when they do 'dialplan show ...' (even with debug set to non-zero); so I set up a 'dialplan debug [context]' cli command instead, to explicitly show just the trie info. I even added an extension_exists() call to make sure the trie info is built. I moved the explanatory header to above the extension loop to ensure it only prints once. And it will do this now, whether debug is set or not. I removed the trie printing from the 'dialplan show' command entirely. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@148562 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Steve Murphy	2c1bfe7643	Merged revisions 147807 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ........ r147807 | murf | 2008-10-09 08:17:33 -0600 (Thu, 09 Oct 2008) | 15 lines (closes issue #13557) Reported by: nickpeirson Patches: pbx.c.patch uploaded by nickpeirson (license 579) replace_bzero+bcopy.patch uploaded by nickpeirson (license 579) Tested by: nickpeirson, murf 1. replaced all refs to bzero and bcopy to memset and memmove instead. 2. added a note to the CODING-GUIDELINES 3. add two macros to asterisk.h to prevent bzero, bcopy from creeping back into the source 4. removed bzero from configure, configure.ac, autoconfig.h.in ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@147811 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Steve Murphy	59d9a004e9	Merged revisions 144678 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ................ r144678 | murf | 2008-09-26 11:50:35 -0600 (Fri, 26 Sep 2008) | 20 lines Merged revisions 144677 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r144677 | murf | 2008-09-26 11:47:13 -0600 (Fri, 26 Sep 2008) | 12 lines (closes issue #13563) Reported by: mnicholson Patches: found1.diff uploaded by mnicholson (license 96) This patch was mainly meant to apply to trunk and 1.6.x, but I'm applying it to 1.4 also, which should be a perfectly harmless fix to the vast majority of users who are not using external switches, but the few who might be affected will not have to go to the pain of filing a bug report. ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@144680 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Steve Murphy	6f1745168e	Merged revisions 144569 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ........ r144569 | murf | 2008-09-25 16:21:28 -0600 (Thu, 25 Sep 2008) | 14 lines (closes issue #13557) Reported by: nickpeirson The user attached a patch, but the license is not yet recorded. I took the liberty of finding and replacing ALL index() calls with strchr() calls, and that involves more than just main/pbx.c; chan_oss, app_playback, func_cut also had calls to index(), and I changed them out. 1.4 had no references to index() at all. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@144578 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Steve Murphy	bc73329607	Merged revisions 142676 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ................ r142676 | murf | 2008-09-11 22:50:48 -0600 (Thu, 11 Sep 2008) | 40 lines Merged revisions 142675 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r142675 | murf | 2008-09-11 22:29:34 -0600 (Thu, 11 Sep 2008) | 29 lines Tested by: sergee, murf, chris-mac, andrew, KNK This is a "second attempt" to restore the previous "endbeforeh" behavior in 1.4 and up. In order to capture information concerning all the legs of transfers in all their infinite combinations, I was forced to this particular solution by a chain of logical necessities, the first being that I was not allowed to rewrite the CDR mechanism from the ground up! This change basically leaves the original machinery alone, which allows IVR and local channel type situations to generate CDR's as normal, but a channel flag can be set to suppress the normal running of the h exten. That flag would be set by the code that runs the h exten from the ast_bridge_call routine, to prevent the h exten from being run twice. Also, a flag in the ast_bridge_config struct passed into ast_bridge_call can be used to suppress the running of the h exten in that routine. This would happen, for instance, if you use the 'g' option in the Dial app. Running this routine 'early' allows not only the CDR() func to be used in the h extension for reading CDR variables, but also allows them to be modified before the CDR is posted to the backends. While I dearly hope that this patch overcomes all problems, and introduces no new problems, reality suggests that surely someone will have problems. In this case, please re-open 13251 (or 13289), and we'll see if we can't fix any remaining issues. ** trunk note: some code to suppress the h exten being run from app_queue was added; for the 'continue' option available only in trunk/1.6.x. ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@142678 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	b55ecb6aea	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 ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@141952 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Steve Murphy	09fe3752a5	Merged revisions 140824 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ........ r140824 | murf | 2008-09-03 08:01:27 -0600 (Wed, 03 Sep 2008) | 21 lines In these changes, I have added some explanation of changes to the Set and MSet apps, so people aren't so shocked and surprised when they upgrade from 1.4 to 1.6. Also, for the sake of those upgrading from 1.4 to 1.6 with AEL, I provide automatic support for the "old" way of using Set(), that still does the exact same old thing with quotes and backslashes and so on as 1.4 did, by having AEL compile in the use of MSet() instead of Set(), everywhere it inserts this code. But, if the app_set var is set to 1.6 or higher, it uses the "new", non-evaluative Set(). This only usually happens if the user manually inserts this into the asterisk.conf file, or runs the "make samples" command. (closes issue #13249) Reported by: dimas Patches: ael-MSet.diff uploaded by murf (license 17) Tested by: dimas, murf ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@140886 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Steve Murphy	0c8172a97c	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. ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@140693 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Steve Murphy	dfe7bf451a	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! ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@139778 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Steve Murphy	41138ad333	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. ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@139629 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Steve Murphy	7af40ac1a1	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. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@138853 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	88d23d712e	Merged revisions 138409 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ........ r138409 | tilghman | 2008-08-16 07:52:06 -0500 (Sat, 16 Aug 2008) | 3 lines Also make sure hinting won't crash on reload. (Closes issue #13312) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@138411 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	f42b756512	Merged revisions 138148 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ........ r138148 | tilghman | 2008-08-15 14:36:11 -0500 (Fri, 15 Aug 2008) | 2 lines Change free to ast_free_ptr, too ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@138150 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	0ffc03dffc	Merged revisions 138124 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/trunk ........ r138124 | tilghman | 2008-08-15 14:22:48 -0500 (Fri, 15 Aug 2008) | 4 lines e->data can be NULL, so use the safe version of ast_strdup() (closes issue #13312) Reported by: pj ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.1@138133 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Sean Bright	790fde68d9	Another batch of files from RSW. The remaining apps and a few more ... files from main/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@137089 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	316fb598d2	Don't allow Answer() to accept a negative argument. ... Negative argument means an infinite delay and we don't want that. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@136635 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Kevin P. Fleming	7df8b8b848	make datastore creation and destruction a generic API since it is not really channel related, and add the ability to add/find/remove datastores to manager sessions ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@135680 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Steve Murphy	9051edfa4a	(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. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@135265 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Steve Murphy	1adecc56eb	(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 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@133299 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	1517710d7e	Change several 'core' commands to be 'dialplan' commands (with appropriate ... deprecation, of course) (closes issue #13016) Reported by: caio1982 Patches: dialplan_globals6.diff uploaded by caio1982 (license 22) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@131606 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Steve Murphy	a1fe3d917f	(closes issue #12960) ...	17 years ago
Tilghman Lesher	49715c05f1	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) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@131044 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Steve Murphy	42942b790d	(closes issue #13041) ... Reported by: eliel OK, now the context registrar slot is strdup'd. It is freed on destruction. I don't see the need to do this with all the structs' registrar fields, but if some wild case proves they should also be handled this way, then we can put in the extra work at that time. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@130297 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Steve Murphy	2ca242b2ba	(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. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@130145 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Matthew Fredrickson	0b185a2276	Add Proceeding() application (#13025) ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@129399 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	4ff527903e	Code wasn't ready to be merged - see -dev list discussion ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@129307 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Brett Bryant	d185405755	Janitor project to convert sizeof to ARRAY_LEN macro. ... (closes issue #13002) Reported by: caio1982 Patches: janitor_arraylen5.diff uploaded by caio1982 (license 22) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@129045 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Olle Johansson	6f400edeab	Changing name of global api call to ast_* ... My mistake, pointed out by Russell. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@128378 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Olle Johansson	45e79490ba	Implement flags for AGI in the channel structure so taht "show channels" and ... AMI commands can display that a channel is under control of an AGI. Work inspired by work at customer site, but paid for by Edvina AB git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@128240 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	12e5c68622	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 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@128027 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Steve Murphy	bc2cfb3e81	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 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@127793 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	6d5b1d76ab	If we don't match registrar when destroying a context, it can cause a crash. ... (closes issue #12835) Reported by: ys Patches: pbx.c.diff uploaded by ys (license 281) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@123358 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Steve Murphy	f4c85ebd22	(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. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@123165 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	e9d72e0cb2	Merge another big set of changes from team/russell/events ... This commit merges in the rest of the code needed to support distributed device state. There are two main parts to this commit. Core changes: - The device state handling in the core has been updated to understand device state across a cluster of Asterisk servers. Every time the state of a device changes, it looks at all of the device states on each node, and determines the aggregate device state. That resulting device state is what is provided to modules in Asterisk that take actions based on the state of a device. New module, res_ais: - A module has been written to facilitate the communication of events between nodes in a cluster of Asterisk servers. This module uses the SAForum AIS (Service Availability Forum Application Interface Specification) CLM and EVT services (Cluster Management and Event) to handle this task. This module currently supports sharing Voicemail MWI (Message Waiting Indication) and device state events between servers. It has been tested with openais, though other implementations of the spec do exist. For more information on testing distributed device state, see the following doc: - doc/distributed_devstate.txt git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@121559 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	42c1e3601e	Merge another change from team/russell/events ... This commit breaks out some logic from pbx.c into a simple API. The hint processing code had logic for taking the state from multiple devices and turning that into the state for a single extension. So, I broke this out and made an API that lets you take multiple device states and determine the aggregate device state. I needed this for some core device state changes to support distributed device state. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@121501 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	f4a8062e93	Merge another change from team/russell/events ... ... DUNDi uses a concept called the Entity ID for unique server identifiers. I have pulled out the handling of EIDs and made it something available to all of Asterisk. There is now a global Entity ID that can be used for other purposes as well, such as code providing distributed device state, which is why I did this. The global Entity ID is set automatically, just like it was done in DUNDi, but it can also be set in asterisk.conf. DUNDi will now use this global EID unless one is specified in dundi.conf. The current EID for the system can be seen in the "core show settings" CLI command. It is also available in the dialplan via the ENTITYID variable. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@121439 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago