Commit Graph

3345 Commits

Author SHA1 Message Date
Mark Michelson 04e56bde03 Fix queue crashes that would occur after the calling channel was masqueraded.
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
2009-01-27 19:30:54 +00:00
Sean Bright ec5aa59105 Merged revisions 170979 via svnmerge from
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
2009-01-25 13:35:48 +00:00
Joshua Colp 665bba38f1 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: https://origsvn.digium.com/svn/asterisk/trunk@170569 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-01-23 19:09:18 +00:00
Joshua Colp fcf4d42cde Reset the ast_str used for escape substitution. We need to do this since it 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
2009-01-23 17:32:26 +00:00
Joshua Colp f6f5197e63 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: https://origsvn.digium.com/svn/asterisk/trunk@170148 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-01-22 16:52:21 +00:00
Joshua Colp 9ee6a741dc 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: https://origsvn.digium.com/svn/asterisk/trunk@170047 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-01-22 15:01:54 +00:00
Mark Michelson 4263503bd5 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: https://origsvn.digium.com/svn/asterisk/trunk@169611 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-01-21 00:33:32 +00:00
Mark Michelson b52253d590 Use the default timeout for a queue instead of -1
(closes issue #14272)
Reported by: timking



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@169574 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-01-20 21:57:24 +00:00
Tilghman Lesher 071301029c 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: https://origsvn.digium.com/svn/asterisk/trunk@169365 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-01-19 20:05:52 +00:00
Tilghman Lesher 3728c3aa92 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: https://origsvn.digium.com/svn/asterisk/trunk@168832 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-01-16 18:49:09 +00:00
Kevin P. Fleming 9a7efae8fd remove the PBX_ODBC logic from the configure script, and add GENERIC_ODCB 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
2009-01-15 20:18:53 +00:00
Sean Bright fcb69e6f9d 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: https://origsvn.digium.com/svn/asterisk/trunk@168705 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-01-15 15:33:18 +00:00
Olle Johansson 526cc089a9 Add support for setting the Reason header when cancelling a call in the queue
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
2009-01-15 13:01:52 +00:00
Mark Michelson b9060d4435 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: https://origsvn.digium.com/svn/asterisk/trunk@168629 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-01-15 00:14:17 +00:00
Steve Murphy a474341105 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: https://origsvn.digium.com/svn/asterisk/trunk@168613 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-01-14 20:51:26 +00:00
Terry Wilson 2015f9c980 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: https://origsvn.digium.com/svn/asterisk/trunk@168594 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-01-14 02:00:40 +00:00
Mark Michelson ec0f18405e Clarify a message that app_queue prints and change to a debug-level message
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
2009-01-13 22:30:59 +00:00
Russell Bryant ef6ad2b53c 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/trunk@168562 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-01-13 19:22:13 +00:00
Olle Johansson 64e564aa55 Better to use the proper app name
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@168497 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-01-12 16:31:27 +00:00
Terry Wilson 1253c8aa0d Set ORIGINATE_STATUS instead of OUTGOING_STATUS to match the documentation
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@167973 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-01-09 01:15:43 +00:00
Terry Wilson c37aa68d77 Set peer context and exten values so MACRO_EXTEN and MACRO_CONTEXT will be set
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@167935 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-01-09 00:13:12 +00:00
Tilghman Lesher 6dbe101045 Fix variables to comply with documentation changes
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@167837 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-01-08 21:40:28 +00:00
Tilghman Lesher 31a3307245 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: https://origsvn.digium.com/svn/asterisk/trunk@167835 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-01-08 21:32:45 +00:00
Mark Michelson 454241dd58 Add the average talk time for a queue
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
2009-01-08 19:48:42 +00:00
BJ Weschke 3a4e3df193 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: https://origsvn.digium.com/svn/asterisk/trunk@167478 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-01-07 18:20:31 +00:00
Mark Michelson ff20b9116a 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: https://origsvn.digium.com/svn/asterisk/trunk@166861 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-12-29 18:04:52 +00:00
Steve Murphy aa905e347e 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/trunk@166665 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-12-23 18:13:49 +00:00
Eliel C. Sardanons 340d22ab39 Fix the XML documentation for Record().
<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
2008-12-19 18:20:46 +00:00
Russell Bryant 37d8f255e4 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: https://origsvn.digium.com/svn/asterisk/trunk@165890 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-12-19 15:05:09 +00:00
Tilghman Lesher a117714b88 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: https://origsvn.digium.com/svn/asterisk/trunk@165797 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-12-18 21:41:02 +00:00
Joshua Colp 654ea55a65 Numerous documentation updates.
(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
2008-12-18 21:21:44 +00:00
Russell Bryant 50a25ac847 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/trunk@165723 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-12-18 19:33:42 +00:00
Tilghman Lesher 665b55e6f8 Fix 2 resource leaks and fix another pipe-to-comma conversion
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@165658 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-12-18 18:36:48 +00:00
Russell Bryant 9e3ecac5ab Add a \todo note for app_originate.
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
2008-12-18 14:23:22 +00:00
Russell Bryant aecde42abb Add a new application, Originate.
(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
2008-12-18 13:33:34 +00:00
Tilghman Lesher 08ae164b58 Add RECORD_STATUS variable, as requested on the -users list.
Patch by me (license 14)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@165397 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-12-17 23:39:48 +00:00
Mark Michelson 7c1bd94231 Fix the build
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@165326 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-12-17 21:29:30 +00:00
Tilghman Lesher 6c521ba21f Oops, broke trunk
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@165325 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-12-17 21:28:51 +00:00
Tilghman Lesher f09b0b3a83 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: https://origsvn.digium.com/svn/asterisk/trunk@165319 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-12-17 21:18:57 +00:00
Mark Michelson a7829044ec 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: https://origsvn.digium.com/svn/asterisk/trunk@165318 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-12-17 21:17:20 +00:00
Mark Michelson 93a51114cb And actually assign the function to a pointer...
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@165143 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-12-17 17:53:37 +00:00
Mark Michelson d0671d8d30 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.



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@165142 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-12-17 17:52:50 +00:00
Jeff Peeler 91b4a30be8 (closes issue #13669)
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
2008-12-16 22:45:39 +00:00
Russell Bryant cf502aa246 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: https://origsvn.digium.com/svn/asterisk/trunk@164877 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-12-16 21:12:49 +00:00
Russell Bryant c76bd59354 Set MINIVM_ACCMESS_STATUS in all cases. Also, remove a variable that was not needed.
(closes issue #14081)
Reported by: pkempgen


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@164623 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-12-16 15:00:27 +00:00
Mark Michelson 763d4dcabb Add an 'i' option to app_page. This option works the same as
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
2008-12-15 20:07:03 +00:00
Mark Michelson 00c40264b7 Fix a compile warning and a logic error that could have been bad
for non-realtime queues



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@164270 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-12-15 16:16:47 +00:00
Mark Michelson 8a2cf30830 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: https://origsvn.digium.com/svn/asterisk/trunk@164268 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-12-15 16:10:43 +00:00
Joshua Colp 8be6bc5f67 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: https://origsvn.digium.com/svn/asterisk/trunk@164257 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-12-15 15:41:22 +00:00
Russell Bryant bca058070e Fix build WRT ast_str_opaque
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@164202 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-12-15 14:40:08 +00:00