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2009-01-27Fix queue crashes that would occur after the calling channel was masqueraded.Mark Michelson
The data passed to the end_bridge_callback was assumed to be data which was still stack'd. The problem was that with some call features, attended transfers in particular, a new bridge thread is started once the feature completes, meaning that when the end_bridge_callback is called, the end_bridge_callback_data was invalid. To fix this problem, there are two measures taken 1. Instead of pointing to stacked data, we now used heap-allocated data for passing to the end_bridge_callback in app_queue 2. Since bridges can end multiple times on a single logical call, we wait until the final bridge is broken to actually set any queue variables. This is accomplished through reference-counting and the use of an end_bridge_callback_data_fixup function in app_queue.c (closes issue #14260) Reported by: ccesario Patches: 14260.patch uploaded by putnopvut (license 60) Tested by: ccesario git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@171618 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-01-25Merged revisions 170979 via svnmerge from Sean Bright
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r170979 | seanbright | 2009-01-25 08:33:20 -0500 (Sun, 25 Jan 2009) | 9 lines Resolve a logic error that was causing Page() to crash when more than one channel was specified. (closes issue #14308) Reported by: bluefox Patches: 20090124__bug14308.diff.txt uploaded by seanbright (license 71) Tested by: kc0bvu ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@170980 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-01-23Merged revisions 170568 via svnmerge from Joshua Colp
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: https://origsvn.digium.com/svn/asterisk/trunk@170569 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-01-23Reset the ast_str used for escape substitution. We need to do this since it ↵Joshua Colp
is a thread local variable that may contain the value of a previous substitution. (closes issue #14312) Reported by: pj git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@170498 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-01-22Merged revisions 170147 via svnmerge from Joshua Colp
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: https://origsvn.digium.com/svn/asterisk/trunk@170148 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-01-22Clear the autoloop flag when parsing and setting the ↵Joshua Colp
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: https://origsvn.digium.com/svn/asterisk/trunk@170047 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-01-21Fix device state parsing issues for channel names with multiple slashesMark Michelson
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: https://origsvn.digium.com/svn/asterisk/trunk@169611 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-01-20Use the default timeout for a queue instead of -1Mark Michelson
(closes issue #14272) Reported by: timking git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@169574 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-01-19Merged revisions 169364 via svnmerge from Tilghman Lesher
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: https://origsvn.digium.com/svn/asterisk/trunk@169365 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-01-16Merged revisions 168828 via svnmerge from Tilghman Lesher
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: https://origsvn.digium.com/svn/asterisk/trunk@168832 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-01-15remove the PBX_ODBC logic from the configure script, and add GENERIC_ODCB ↵Kevin P. Fleming
logic that includes copying the relevant LIB and INCLUDE data from either UnixODBC or iODBC, based on which was found; if both were found, prefer UnixODBC this stops modules from being linked against both sets of libraries on systems that have both installed git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@168734 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-01-15Add a missing unlock and properly handle the 'maxusers' setting on MeetMeSean Bright
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: https://origsvn.digium.com/svn/asterisk/trunk@168705 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-01-15Add support for setting the Reason header when cancelling a call in the queue Olle Johansson
because someone else answered. Previously, only dial() was supported. EDV-102 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@168636 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-01-15Merged revisions 168628 via svnmerge from Mark Michelson
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: https://origsvn.digium.com/svn/asterisk/trunk@168629 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-01-14Merged revisions 168608 via svnmerge from Steve Murphy
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: https://origsvn.digium.com/svn/asterisk/trunk@168613 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-01-14Merged revisions 168593 via svnmerge from Terry Wilson
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: https://origsvn.digium.com/svn/asterisk/trunk@168594 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-01-13Clarify a message that app_queue prints and change to a debug-level messageMark Michelson
The "No one is answering..." verbose message contained 3 numbers that were not explained in any way to whoever was viewing the message. It is more helpful now since the message explains what the numbers mean. Also, the message has been downgraded to "DEBUG" level. (closes issue #14172) Reported by: caio1982 Patches: queue_answering_debug.diff uploaded by caio1982 (license 22) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@168579 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-01-13Merged revisions 168561 via svnmerge from Russell Bryant
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/trunk@168562 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-01-12Better to use the proper app nameOlle Johansson
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@168497 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-01-09Set ORIGINATE_STATUS instead of OUTGOING_STATUS to match the documentationTerry Wilson
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@167973 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-01-09Set peer context and exten values so MACRO_EXTEN and MACRO_CONTEXT will be setTerry Wilson
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@167935 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-01-08Fix variables to comply with documentation changesTilghman Lesher
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@167837 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-01-08Textual changes, consistency in status variable naming, and other minor bugs.Tilghman Lesher
(closes issue #13943) Reported by: Marquis Patches: minivm_trunk_fixes3.patch uploaded by Marquis (license 32) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@167835 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-01-08Add the average talk time for a queueMark Michelson
This patch adds the functionality to app_queue of calculating the average amount of time that channels are bridged for a queue. The algorithm used to calculate the average is the same exponential average currently used to calculate the average holdtime. See the CHANGES file to see the methods you may use to view this information. (closes issue #13960) Reported by: coolmig Patches: app_queue.c.diff.trunk-r158840 uploaded by coolmig (license 621) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@167792 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-01-07 Answer the channel if it has not already been answered and we've already ↵BJ Weschke
found a valid profile for followme. (closes issue #14140) Reported by: dimas Patches: 14140.patch uploaded by dimas git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@167478 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-12-29Update app_queue to deal with the removal of AST_PBX_KEEPALIVEMark Michelson
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: https://origsvn.digium.com/svn/asterisk/trunk@166861 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-12-23Merged revisions 166093 via svnmerge from Steve Murphy
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/trunk@166665 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-12-19Fix the XML documentation for Record().Eliel C. Sardanons
<value> tags inside <variable> elements must have CDATA and no another XML node. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@165954 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-12-19Merged revisions 165889 via svnmerge from Russell Bryant
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: https://origsvn.digium.com/svn/asterisk/trunk@165890 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-12-18Merged revisions 165767 via svnmerge from Tilghman Lesher
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: https://origsvn.digium.com/svn/asterisk/trunk@165797 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-12-18Numerous documentation updates.Joshua Colp
(closes issue #13970) Reported by: pkempgen Patches: __20081217_cli_usage_fixes.patch.txt uploaded by blitzrage (license 10) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@165792 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-12-18Remove the need for AST_PBX_KEEPALIVE with the GoSub option from Dial.Russell Bryant
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/trunk@165723 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-12-18Fix 2 resource leaks and fix another pipe-to-comma conversionTilghman Lesher
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@165658 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-12-18Add a \todo note for app_originate.Russell Bryant
Jared Smith suggested that we add a way to be able to set variables and functions on the outbound channel. I think that it's a great idea, so I have added it as a todo so that it gets done at some point. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@165469 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-12-18Add a new application, Originate.Russell Bryant
(closes issue #14075) Reported by: rcasas Patches: app_originate.c uploaded by rcasas (license 641), heavily modified by me Tested by: russell Review: http://reviewboard.digium.com/r/95/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@165433 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-12-17Add RECORD_STATUS variable, as requested on the -users list.Tilghman Lesher
Patch by me (license 14) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@165397 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-12-17Fix the buildMark Michelson
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@165326 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-12-17Oops, broke trunkTilghman Lesher
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@165325 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-12-17Merged revisions 165317 via svnmerge from Tilghman Lesher
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: https://origsvn.digium.com/svn/asterisk/trunk@165319 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-12-17Merged revisions 165255 via svnmerge from Mark Michelson
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: https://origsvn.digium.com/svn/asterisk/trunk@165318 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-12-17And actually assign the function to a pointer...Mark Michelson
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@165143 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-12-17Use the create_vm_state_from_user function in a place whereMark Michelson
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. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@165142 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-12-16(closes issue #13669)Jeff Peeler
Reported by: pj Delete file recording if recording terminated from a hangup. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@164942 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-12-16Merged revisions 164876 via svnmerge from Russell Bryant
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: https://origsvn.digium.com/svn/asterisk/trunk@164877 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-12-16Set MINIVM_ACCMESS_STATUS in all cases. Also, remove a variable that was ↵Russell Bryant
not needed. (closes issue #14081) Reported by: pkempgen git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@164623 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-12-15Add an 'i' option to app_page. This option works the same asMark Michelson
the 'i' options for app_dial and app_queue, in that they will ignore any attempts by phones to forward the call. (closes issue #13977) Reported by: putnopvut Patches: page_ignore_forwards.patch uploaded by putnopvut (license 60) Tested by: putnopvut, acunningham git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@164428 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-12-15Fix a compile warning and a logic error that could have been badMark Michelson
for non-realtime queues git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@164270 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-12-15Fix up a few issues with regards to queuesMark Michelson
* 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: https://origsvn.digium.com/svn/asterisk/trunk@164268 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-12-15Make app_fax compatible with newer versions of spandsp. This remains ↵Joshua Colp
backwards compatible with earlier versions though so do not fret. (closes issue #14073) Reported by: seandarcy git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@164257 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-12-15Fix build WRT ast_str_opaqueRussell Bryant
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@164202 65c4cc65-6c06-0410-ace0-fbb531ad65f3
generated by cgit v1.2.3 (git 2.39.1) at 2024-07-06 02:05:12 +0000