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2009-01-23Merged revisions 170569 via svnmerge from file1-0/+3
https://origsvn.digium.com/svn/asterisk/trunk ................ r170569 | file | 2009-01-23 15:09:18 -0400 (Fri, 23 Jan 2009) | 11 lines Merged revisions 170568 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r170568 | file | 2009-01-23 15:06:54 -0400 (Fri, 23 Jan 2009) | 4 lines When a call is forwarded stop any active indications. The new channel will provide an indication, if need be, itself. (closes issue #14310) Reported by: RadicAlish ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@170571 f38db490-d61c-443f-a65b-d21fe96a405b
2009-01-22Merged revisions 170148 via svnmerge from file1-4/+4
https://origsvn.digium.com/svn/asterisk/trunk ................ r170148 | file | 2009-01-22 12:52:21 -0400 (Thu, 22 Jan 2009) | 11 lines Merged revisions 170147 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r170147 | file | 2009-01-22 12:50:54 -0400 (Thu, 22 Jan 2009) | 4 lines If we are unable to request a DAHDI pseudo channel and we are using the user introduction without review option make sure it gets unset so other code does not blindly assume a DAHDI pseudo channel exists. (closes issue #14282) Reported by: cheesegrits ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@170150 f38db490-d61c-443f-a65b-d21fe96a405b
2009-01-22Merged revisions 170047 via svnmerge from file1-1/+3
https://origsvn.digium.com/svn/asterisk/trunk ........ r170047 | file | 2009-01-22 11:01:54 -0400 (Thu, 22 Jan 2009) | 4 lines Clear the autoloop flag when parsing and setting the context/extension/priority to go back to. When the channel executes a PBX again we want it to start out at the point we explicitly say and at that point it will not yet be doing autoloop. (closes issue #14304) Reported by: jcovert ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@170049 f38db490-d61c-443f-a65b-d21fe96a405b
2009-01-21Merged revisions 169611 via svnmerge from mmichelson1-2/+2
https://origsvn.digium.com/svn/asterisk/trunk ........ r169611 | mmichelson | 2009-01-20 18:33:32 -0600 (Tue, 20 Jan 2009) | 22 lines Fix device state parsing issues for channel names with multiple slashes The fix being applied is a bit different for trunk and the 1.6.X branches. For trunk, we only wish to strip off the characters beyond the second slash if the channel is a Local channel (i.e. we are removing the /n from the device name). Other channel technologies with multiple slashes (e.g. DAHDI) need the information after the second slash in order to get the proper device state information. In addition to this fix, the 1.6.X branches are receiving a much more important fix as well. The problem in 1.6.X is that the member's device name was being directly changed instead of having a copy changed. This meant that we would strip off the second slash and trailing characters and then leave the member's device name like that permanently thereafter. (closes issue #14014) Reported by: kebl0155 Patches: 14014_number2.patch uploaded by putnopvut (license 60) Tested by: kebl0155 ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@169613 f38db490-d61c-443f-a65b-d21fe96a405b
2009-01-20Merged revisions 169574 via svnmerge from mmichelson1-1/+1
https://origsvn.digium.com/svn/asterisk/trunk ........ r169574 | mmichelson | 2009-01-20 15:57:24 -0600 (Tue, 20 Jan 2009) | 6 lines Use the default timeout for a queue instead of -1 (closes issue #14272) Reported by: timking ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@169575 f38db490-d61c-443f-a65b-d21fe96a405b
2009-01-19Merged revisions 169365 via svnmerge from tilghman1-7/+12
https://origsvn.digium.com/svn/asterisk/trunk ................ r169365 | tilghman | 2009-01-19 14:05:52 -0600 (Mon, 19 Jan 2009) | 11 lines Merged revisions 169364 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r169364 | tilghman | 2009-01-19 13:49:25 -0600 (Mon, 19 Jan 2009) | 4 lines Truncate userevents at the end of a line, when the command exceeds the buffer. (closes issue #14278) Reported by: fnordian ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@169368 f38db490-d61c-443f-a65b-d21fe96a405b
2009-01-16Merged revisions 168832 via svnmerge from tilghman1-146/+98
https://origsvn.digium.com/svn/asterisk/trunk ................ r168832 | tilghman | 2009-01-16 12:49:09 -0600 (Fri, 16 Jan 2009) | 13 lines Merged revisions 168828 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r168828 | tilghman | 2009-01-16 12:41:35 -0600 (Fri, 16 Jan 2009) | 6 lines Fix the conjugation of Russian and Ukrainian languages. (related to issue #12475) Reported by: chappell Patches: vm_multilang.patch uploaded by chappell (license 8) ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@168836 f38db490-d61c-443f-a65b-d21fe96a405b
2009-01-15Merged revisions 168705 via svnmerge from seanbright1-1/+3
https://origsvn.digium.com/svn/asterisk/trunk ........ r168705 | seanbright | 2009-01-15 10:33:18 -0500 (Thu, 15 Jan 2009) | 11 lines Add a missing unlock and properly handle the 'maxusers' setting on MeetMe conferences. We were using the 'user number' field to compare against the maximum allowed users, which works assuming users with lower user numbers didn't leave the conference. (closes issue #14117) Reported by: sergedevorop Patches: 20090114__bug14117-2.diff.txt uploaded by seanbright (license 71) Tested by: sergedevorop ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@168707 f38db490-d61c-443f-a65b-d21fe96a405b
2009-01-15Merged revisions 168629 via svnmerge from mmichelson1-5/+10
https://origsvn.digium.com/svn/asterisk/trunk ................ r168629 | mmichelson | 2009-01-14 18:14:17 -0600 (Wed, 14 Jan 2009) | 24 lines Merged revisions 168628 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r168628 | mmichelson | 2009-01-14 18:11:01 -0600 (Wed, 14 Jan 2009) | 16 lines Fix some crashes from bad datastore handling in app_queue.c * The queue_transfer_fixup function was searching for and removing the datastore from the incorrect channel, so this was fixed. * Most datastore operations regarding the queue_transfer datastore were being done without the channel locked, so proper channel locking was added, too. (closes issue #14086) Reported by: ZX81 Patches: 14086v2.patch uploaded by putnopvut (license 60) Tested by: ZX81, festr ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@168631 f38db490-d61c-443f-a65b-d21fe96a405b
2009-01-14Merged revisions 168613 via svnmerge from murf1-1/+1
https://origsvn.digium.com/svn/asterisk/trunk ................ r168613 | murf | 2009-01-14 13:51:26 -0700 (Wed, 14 Jan 2009) | 9 lines Merged revisions 168608 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r168608 | murf | 2009-01-14 12:34:35 -0700 (Wed, 14 Jan 2009) | 1 line app_page was failing to compile in dev-mode on my gcc-4.2.4 system. This change gets rid of the warning. ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@168621 f38db490-d61c-443f-a65b-d21fe96a405b
2009-01-14Merged revisions 168594 via svnmerge from twilson1-5/+17
https://origsvn.digium.com/svn/asterisk/trunk ................ r168594 | twilson | 2009-01-13 20:00:40 -0600 (Tue, 13 Jan 2009) | 27 lines Merged revisions 168593 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r168593 | twilson | 2009-01-13 19:27:18 -0600 (Tue, 13 Jan 2009) | 20 lines Don't overflow when paging more than 128 extensions The number of available slots for calls in app_page was hardcoded to 128. Proper bounds checking was not in place to enforce this limit, so if more than 128 extensions were passed to the Page() app, Asterisk would crash. This patch instead dynamically allocates memory for the ast_dial structures and removes the (non-functional) arbitrary limit. This issue would have special importance to anyone who is dynamically creating the argument passed to the Page application and allowing more than 128 extensions to be added by an outside user via some external interface. The patch posted by a_villacis was slightly modified for some coding guidelines and other cleanups. Thanks, a_villacis! (closes issue #14217) Reported by: a_villacis Patches: 20080912-asterisk-app_page-fix-buffer-overflow.patch uploaded by a (license 660) Tested by: otherwiseguy ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@168596 f38db490-d61c-443f-a65b-d21fe96a405b
2009-01-13Merged revisions 168562 via svnmerge from russell3-3/+3
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: http://svn.digium.com/svn/asterisk/branches/1.6.1@168565 f38db490-d61c-443f-a65b-d21fe96a405b
2009-01-12Merged revisions 168497 via svnmerge from oej1-2/+2
https://origsvn.digium.com/svn/asterisk/trunk ........ r168497 | oej | 2009-01-12 17:31:27 +0100 (MÃ¥n, 12 Jan 2009) | 2 lines Better to use the proper app name ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@168500 f38db490-d61c-443f-a65b-d21fe96a405b
2009-01-09Merged revisions 167935 via svnmerge from twilson1-0/+4
https://origsvn.digium.com/svn/asterisk/trunk ........ r167935 | twilson | 2009-01-08 18:13:12 -0600 (Thu, 08 Jan 2009) | 2 lines Set peer context and exten values so MACRO_EXTEN and MACRO_CONTEXT will be set ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@167972 f38db490-d61c-443f-a65b-d21fe96a405b
2009-01-08Merged revisions 167835 via svnmerge from tilghman1-62/+67
https://origsvn.digium.com/svn/asterisk/trunk ........ r167835 | tilghman | 2009-01-08 15:32:45 -0600 (Thu, 08 Jan 2009) | 6 lines Textual changes, consistency in status variable naming, and other minor bugs. (closes issue #13943) Reported by: Marquis Patches: minivm_trunk_fixes3.patch uploaded by Marquis (license 32) ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@167836 f38db490-d61c-443f-a65b-d21fe96a405b
2009-01-07Merged revisions 167478 via svnmerge from bweschke1-0/+5
https://origsvn.digium.com/svn/asterisk/trunk ........ r167478 | bweschke | 2009-01-07 13:20:31 -0500 (Wed, 07 Jan 2009) | 7 lines Answer the channel if it has not already been answered and we've already found a valid profile for followme. (closes issue #14140) Reported by: dimas Patches: 14140.patch uploaded by dimas ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@167502 f38db490-d61c-443f-a65b-d21fe96a405b
2008-12-29Merged revisions 166861 via svnmerge from mmichelson2-5/+7
https://origsvn.digium.com/svn/asterisk/trunk ........ r166861 | mmichelson | 2008-12-29 12:04:52 -0600 (Mon, 29 Dec 2008) | 14 lines Update app_queue to deal with the removal of AST_PBX_KEEPALIVE When placing a call to a queue which ran a gosub on the member's channel, Asterisk would crash every time, stemming from the fact that the member's channel was being hung up unexpectedly when the Gosub completed. The necessary change was pretty much copied and pasted from app_dial's similar changes made last week. I also took the opportunity to change a LOG_DEBUG message in app_dial to use ast_debug. I am guessing this was due to a direct merge from 1.4 that was not corrected to use trunk's preferred syntax. ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@166863 f38db490-d61c-443f-a65b-d21fe96a405b
2008-12-24Merged revisions 166665 via svnmerge from murf3-53/+33
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: http://svn.digium.com/svn/asterisk/branches/1.6.1@166730 f38db490-d61c-443f-a65b-d21fe96a405b
2008-12-19Merged revisions 165890 via svnmerge from russell1-1/+1
https://origsvn.digium.com/svn/asterisk/trunk ................ r165890 | russell | 2008-12-19 09:05:09 -0600 (Fri, 19 Dec 2008) | 17 lines Merged revisions 165889 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r165889 | russell | 2008-12-19 09:03:02 -0600 (Fri, 19 Dec 2008) | 9 lines Ensure that the chanspy datastore is fully initialized. This patch resolved some random crash issues observed by a user on a BSD system (closes issue #14111) Reported by: ys Patches: app_chanspy.c.diff uploaded by ys (license 281) ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@165892 f38db490-d61c-443f-a65b-d21fe96a405b
2008-12-18Merged revisions 165797 via svnmerge from tilghman1-10/+52
https://origsvn.digium.com/svn/asterisk/trunk ................ r165797 | tilghman | 2008-12-18 15:41:02 -0600 (Thu, 18 Dec 2008) | 15 lines Merged revisions 165767 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r165767 | tilghman | 2008-12-18 15:14:47 -0600 (Thu, 18 Dec 2008) | 8 lines Add mutexes around accesses to the IMAP library interface. This prevents certain crashes, especially when shared mailboxes are used. (closes issue #13653) Reported by: howardwilkinson Patches: asterisk-1.4.21.2-appvoicemail-sharedimap-lock.patch uploaded by howardwilkinson (license 590) Tested by: jpeeler ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@165808 f38db490-d61c-443f-a65b-d21fe96a405b
2008-12-18Merged revisions 165792 via svnmerge from file1-8/+8
https://origsvn.digium.com/svn/asterisk/trunk ........ r165792 | file | 2008-12-18 17:21:44 -0400 (Thu, 18 Dec 2008) | 6 lines Numerous documentation updates. (closes issue #13970) Reported by: pkempgen Patches: __20081217_cli_usage_fixes.patch.txt uploaded by blitzrage (license 10) ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@165794 f38db490-d61c-443f-a65b-d21fe96a405b
2008-12-18Merged revisions 165723 via svnmerge from russell1-2/+6
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: http://svn.digium.com/svn/asterisk/branches/1.6.1@165728 f38db490-d61c-443f-a65b-d21fe96a405b
2008-12-18Merged revisions 165658 via svnmerge from tilghman1-5/+10
https://origsvn.digium.com/svn/asterisk/trunk ........ r165658 | tilghman | 2008-12-18 12:36:48 -0600 (Thu, 18 Dec 2008) | 2 lines Fix 2 resource leaks and fix another pipe-to-comma conversion ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@165660 f38db490-d61c-443f-a65b-d21fe96a405b
2008-12-17Merged revisions 165325 via svnmerge from tilghman1-1/+1
https://origsvn.digium.com/svn/asterisk/trunk ........ r165325 | tilghman | 2008-12-17 15:28:51 -0600 (Wed, 17 Dec 2008) | 2 lines Oops, broke trunk ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@165329 f38db490-d61c-443f-a65b-d21fe96a405b
2008-12-17Merged revisions 165318 via svnmerge from mmichelson3-1/+10
https://origsvn.digium.com/svn/asterisk/trunk ................ r165318 | mmichelson | 2008-12-17 15:17:20 -0600 (Wed, 17 Dec 2008) | 15 lines Merged revisions 165255 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r165255 | mmichelson | 2008-12-17 14:51:38 -0600 (Wed, 17 Dec 2008) | 7 lines Fix some memory leaks found while looking at how realtime configs are handled. Also cleaned up some coding guidelines violations in app_realtime.c, mostly related to spacing ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@165324 f38db490-d61c-443f-a65b-d21fe96a405b
2008-12-17Merged revisions 165319 via svnmerge from tilghman1-21/+54
https://origsvn.digium.com/svn/asterisk/trunk ................ r165319 | tilghman | 2008-12-17 15:18:57 -0600 (Wed, 17 Dec 2008) | 11 lines Merged revisions 165317 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r165317 | tilghman | 2008-12-17 15:14:37 -0600 (Wed, 17 Dec 2008) | 4 lines Reverse the fix from issue #6176 and add proper handling for that issue. (Closes issue #13962, closes issue #13363) Fixed by myself (license 14) ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@165323 f38db490-d61c-443f-a65b-d21fe96a405b
2008-12-17Merged revisions 165142-165143 via svnmerge from mmichelson1-18/+7
https://origsvn.digium.com/svn/asterisk/trunk ........ r165142 | mmichelson | 2008-12-17 11:52:50 -0600 (Wed, 17 Dec 2008) | 10 lines Use the create_vm_state_from_user function in a place where it was not being used before. Also, I've moved the urgent folder check in messagecount() up a bit so that the flow is a bit better. This was something I noticed while taking a look at issue #13973, although I don't think this is the underlying cause of the issue. ........ r165143 | mmichelson | 2008-12-17 11:53:37 -0600 (Wed, 17 Dec 2008) | 3 lines And actually assign the function to a pointer... ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@165146 f38db490-d61c-443f-a65b-d21fe96a405b
2008-12-16Merged revisions 164942 via svnmerge from jpeeler1-0/+1
https://origsvn.digium.com/svn/asterisk/trunk ........ r164942 | jpeeler | 2008-12-16 16:45:39 -0600 (Tue, 16 Dec 2008) | 6 lines (closes issue #13669) Reported by: pj Delete file recording if recording terminated from a hangup. ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@164960 f38db490-d61c-443f-a65b-d21fe96a405b
2008-12-16Merged revisions 164877 via svnmerge from russell1-1/+1
https://origsvn.digium.com/svn/asterisk/trunk ................ r164877 | russell | 2008-12-16 15:12:49 -0600 (Tue, 16 Dec 2008) | 14 lines Merged revisions 164876 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r164876 | russell | 2008-12-16 15:10:44 -0600 (Tue, 16 Dec 2008) | 6 lines Do not dereference the channel if AST_PBX_KEEPALIVE has been returned. This is a bug I noticed while looking at the code for app_macro. This return code means that another thread has assumed ownership of the channel and it can no longer be touched. (I hate this return code with a passion, by the way.) ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@164879 f38db490-d61c-443f-a65b-d21fe96a405b
2008-12-16Merged revisions 164623 via svnmerge from russell1-4/+6
https://origsvn.digium.com/svn/asterisk/trunk ........ r164623 | russell | 2008-12-16 09:00:27 -0600 (Tue, 16 Dec 2008) | 5 lines Set MINIVM_ACCMESS_STATUS in all cases. Also, remove a variable that was not needed. (closes issue #14081) Reported by: pkempgen ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@164625 f38db490-d61c-443f-a65b-d21fe96a405b
2008-12-15Merged revisions 164270 via svnmerge from mmichelson1-2/+2
https://origsvn.digium.com/svn/asterisk/trunk ........ r164270 | mmichelson | 2008-12-15 10:16:47 -0600 (Mon, 15 Dec 2008) | 4 lines Fix a compile warning and a logic error that could have been bad for non-realtime queues ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@164274 f38db490-d61c-443f-a65b-d21fe96a405b
2008-12-15Merged revisions 164268 via svnmerge from mmichelson1-12/+23
https://origsvn.digium.com/svn/asterisk/trunk ........ r164268 | mmichelson | 2008-12-15 10:10:43 -0600 (Mon, 15 Dec 2008) | 17 lines Fix up a few issues with regards to queues * Fix reference counting used in the __queues_show function * Add code to be sure that the "queue show" command does not print information for a realtime queue which has been deleted from the backend * Add a missing unref to the realtime queue loading function for the case where a queue is in the module's container but has been deleted from the realtime backend (closes issue #14033) Reported by: cristiandimache Patches: 14033.patch uploaded by putnopvut (license 60) Tested by: cristiandimache ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@164273 f38db490-d61c-443f-a65b-d21fe96a405b
2008-12-15Merged revisions 164257 via svnmerge from file1-0/+3
https://origsvn.digium.com/svn/asterisk/trunk ........ r164257 | file | 2008-12-15 11:41:22 -0400 (Mon, 15 Dec 2008) | 4 lines Make app_fax compatible with newer versions of spandsp. This remains backwards compatible with earlier versions though so do not fret. (closes issue #14073) Reported by: seandarcy ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@164266 f38db490-d61c-443f-a65b-d21fe96a405b
2008-12-13Merged revisions 163912 via svnmerge from file1-8/+13
https://origsvn.digium.com/svn/asterisk/trunk ........ r163912 | file | 2008-12-12 20:59:24 -0400 (Fri, 12 Dec 2008) | 2 lines Only detach and destroy the whisper audiohooks if they are actually in use. ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@163914 f38db490-d61c-443f-a65b-d21fe96a405b
2008-12-13Merged revisions 163873 via svnmerge from twilson1-2/+20
https://origsvn.digium.com/svn/asterisk/trunk ........ r163873 | twilson | 2008-12-12 17:48:26 -0600 (Fri, 12 Dec 2008) | 6 lines When using realtime queues, app_queue wasn't updating the strategy if it was changed in the realtime backend. This patch resolves the issue for almost all situations. It is currently not supported to switch to the linear strategy via realtime since the ao2_container for members will have been set to have multiple buckets and therefore the members would be unordered. (closes issue #14034) Reported by: cristiandimache Tested by: otherwiseguy, cristiandimache ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@163875 f38db490-d61c-443f-a65b-d21fe96a405b
2008-12-11Merged revisions 163213 via svnmerge from mmichelson1-1/+10
https://origsvn.digium.com/svn/asterisk/trunk ........ r163213 | mmichelson | 2008-12-11 14:57:44 -0600 (Thu, 11 Dec 2008) | 9 lines Add an option to voicemail.conf to allow urgent messages to be forwarded as not urgent. (closes issue #14063) Reported by: jaroth Patches: urgfwd_v2.patch uploaded by jaroth (license 50) ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@163214 f38db490-d61c-443f-a65b-d21fe96a405b
2008-12-11Merged revisions 163085 via svnmerge from mmichelson1-2/+2
https://origsvn.digium.com/svn/asterisk/trunk ................ r163085 | mmichelson | 2008-12-11 10:47:34 -0600 (Thu, 11 Dec 2008) | 12 lines Merged revisions 163084 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r163084 | mmichelson | 2008-12-11 10:46:22 -0600 (Thu, 11 Dec 2008) | 4 lines Revert this cast to long. Using time_t here causes build failures on a FreeBSD 32-bit build. ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@163087 f38db490-d61c-443f-a65b-d21fe96a405b
2008-12-11Merged revisions 163081 via svnmerge from mmichelson1-21/+21
https://origsvn.digium.com/svn/asterisk/trunk ................ r163081 | mmichelson | 2008-12-11 10:33:16 -0600 (Thu, 11 Dec 2008) | 22 lines Merged revisions 163080 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r163080 | mmichelson | 2008-12-11 10:24:43 -0600 (Thu, 11 Dec 2008) | 14 lines Fix a potential crash due to unsafe datastore handling. This patch also contains a conversion from using long to time_t for representing times for a queue, as well as some whitespace fixes. (closes issue #14060) Reported by: nivek Patches: datastore_fixup.patch.corrected uploaded by nivek (license 636) with slight modification from me Tested by: nivek ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@163083 f38db490-d61c-443f-a65b-d21fe96a405b
2008-12-09Merged revisions 162466 via svnmerge from tilghman1-1/+1
https://origsvn.digium.com/svn/asterisk/trunk ................ r162466 | tilghman | 2008-12-09 17:10:34 -0600 (Tue, 09 Dec 2008) | 9 lines Merged revisions 162463 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r162463 | tilghman | 2008-12-09 17:08:53 -0600 (Tue, 09 Dec 2008) | 2 lines Oops, should be "tz", not "zonetag". ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@162472 f38db490-d61c-443f-a65b-d21fe96a405b
2008-12-09Merged revisions 162355 via svnmerge from tilghman1-0/+5
https://origsvn.digium.com/svn/asterisk/trunk ................ r162355 | tilghman | 2008-12-09 15:57:09 -0600 (Tue, 09 Dec 2008) | 11 lines Merged revisions 162348 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r162348 | tilghman | 2008-12-09 15:53:25 -0600 (Tue, 09 Dec 2008) | 4 lines We appear to have documented tz= in the [general] section of voicemail.conf, without actually having implemented it. Oops. (Reported by Olivier on the -users list) ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@162372 f38db490-d61c-443f-a65b-d21fe96a405b
2008-12-09Merged revisions 162342 via svnmerge from file1-1/+1
https://origsvn.digium.com/svn/asterisk/trunk ................ r162342 | file | 2008-12-09 17:16:37 -0400 (Tue, 09 Dec 2008) | 11 lines Merged revisions 162341 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r162341 | file | 2008-12-09 17:14:29 -0400 (Tue, 09 Dec 2008) | 4 lines Add 'down' as a valid state for directed call pickup. This creeps up when we receive session progress when dialing a device and not ringing. (closes issue #14005) Reported by: ddl ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@162344 f38db490-d61c-443f-a65b-d21fe96a405b
2008-12-09Merged revisions 162291 via svnmerge from russell1-7/+27
https://origsvn.digium.com/svn/asterisk/trunk ................ r162291 | russell | 2008-12-09 14:59:54 -0600 (Tue, 09 Dec 2008) | 17 lines Merged revisions 162286 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r162286 | russell | 2008-12-09 14:57:35 -0600 (Tue, 09 Dec 2008) | 9 lines Fix an issue where callers on an incoming call on an SLA trunk would not hear ringback. We need to make sure that we don't start writing audio to the trunk channel until we're actually ready to answer it. Otherwise, the channel driver will treat it as inband progress, even though all they are getting is silence. (closes issue #12471) Reported by: mthomasslo ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@162302 f38db490-d61c-443f-a65b-d21fe96a405b
2008-12-09Merged revisions 162275 via svnmerge from file1-1/+0
https://origsvn.digium.com/svn/asterisk/trunk ................ r162275 | file | 2008-12-09 16:46:11 -0400 (Tue, 09 Dec 2008) | 11 lines Merged revisions 162273 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r162273 | file | 2008-12-09 16:44:32 -0400 (Tue, 09 Dec 2008) | 4 lines Fix double declaration of 'x' on the PPC platform. (closes issue #14038) Reported by: ffloimair ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@162278 f38db490-d61c-443f-a65b-d21fe96a405b
2008-12-09Merged revisions 162016 via svnmerge from russell1-2/+8
https://origsvn.digium.com/svn/asterisk/trunk ................ r162016 | russell | 2008-12-09 10:47:39 -0600 (Tue, 09 Dec 2008) | 13 lines Merged revisions 162014 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r162014 | russell | 2008-12-09 10:46:53 -0600 (Tue, 09 Dec 2008) | 5 lines Allow DISA to handle extensions that start with #. (closes issue #13330) Reported by: jcovert ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@162018 f38db490-d61c-443f-a65b-d21fe96a405b
2008-12-05Merged revisions 161493 via svnmerge from mmichelson1-1/+1
https://origsvn.digium.com/svn/asterisk/trunk ........ r161493 | mmichelson | 2008-12-05 17:24:38 -0600 (Fri, 05 Dec 2008) | 8 lines If the autoloop flag is set on a channel, then we need to add 1 to the priority when checking if the extension exists. Otherwise, gosubs will fail. This was discovered when investigating an asterisk-users mailing list post made by Gary Hawkins. ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@161496 f38db490-d61c-443f-a65b-d21fe96a405b
2008-12-05Merged revisions 161349-161350 via svnmerge from seanbright1-50/+51
https://origsvn.digium.com/svn/asterisk/trunk ........ r161349 | seanbright | 2008-12-05 10:56:15 -0500 (Fri, 05 Dec 2008) | 5 lines When using IMAP_STORAGE, it's important to convert bare newlines (\n) in emailbody and pagerbody to CR-LF so that the IMAP server doesn't spit out an error. This was informally reported on #asterisk-dev a few weeks ago. Reviewed by Mark M. on IRC. ........ r161350 | seanbright | 2008-12-05 11:04:36 -0500 (Fri, 05 Dec 2008) | 2 lines Use ast_free() instead of free(), pointed out by eliel on IRC. ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@161352 f38db490-d61c-443f-a65b-d21fe96a405b
2008-12-05Merged revisions 161252 via svnmerge from russell1-1/+1
https://origsvn.digium.com/svn/asterisk/trunk ........ r161252 | russell | 2008-12-05 07:46:01 -0600 (Fri, 05 Dec 2008) | 2 lines Resolve a compiler warning from buildbot about a NULL format string. ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@161285 f38db490-d61c-443f-a65b-d21fe96a405b
2008-12-05Merged revisions 161147 via svnmerge from seanbright1-2/+11
https://origsvn.digium.com/svn/asterisk/trunk ........ r161147 | seanbright | 2008-12-04 21:47:54 -0500 (Thu, 04 Dec 2008) | 3 lines Check the return value of fread/fwrite so the compiler doesn't complain. Only a problem when IMAP_STORAGE is enabled. ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@161149 f38db490-d61c-443f-a65b-d21fe96a405b
2008-12-03Merged revisions 160791 via svnmerge from tilghman1-6/+6
https://origsvn.digium.com/svn/asterisk/trunk ................ r160791 | tilghman | 2008-12-03 15:58:21 -0600 (Wed, 03 Dec 2008) | 9 lines Merged revisions 160770 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r160770 | tilghman | 2008-12-03 15:54:07 -0600 (Wed, 03 Dec 2008) | 2 lines Some compilers warn on null format strings; some don't (caught by buildbot) ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@160811 f38db490-d61c-443f-a65b-d21fe96a405b
2008-12-03Merged revisions 160626 via svnmerge from mmichelson3-9/+23
https://origsvn.digium.com/svn/asterisk/trunk ........ r160626 | mmichelson | 2008-12-03 12:37:46 -0600 (Wed, 03 Dec 2008) | 16 lines Add some safety measures when using gosub, especially when using the options for app_dial and app_queue to run a gosub when the call is answered. * Check for the existence of the gosub target in gosub_exec. If it is nonexistent, then this will cause errors when we attempt to actually run the gosub, including a definite memory leak and potential crashes. Return an error in this situation * Check the return value of pbx_exec in app_dial and app_queue before attempting to actually run the gosub routine. If there was an error, we should not attempt to run the gosub. * Change a '|' to a ',' in app_queue. * Add some extra curly braces where they had been missing previously. (closes issue #13548) Reported by: fiddur ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.1@160628 f38db490-d61c-443f-a65b-d21fe96a405b
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