Commit graph

632 commits

Author SHA1 Message Date
Olle Johansson
097822966b Add final part of previously committed work for answered elsewhere in queue - the missing piece that started with app_dial() earlier on.
This is to avoid having the list and counter of missed calls being touched by queue calls. Add the C option to queue() and nothing 
will be logged on phones that support the Reason: header on SIP cancel, like the SNOM phones.



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@171924 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-01-28 14:37:16 +00:00
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
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
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
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
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
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
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
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
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
Tilghman Lesher
c8223fc957 Merge ast_str_opaque branch (discontinue usage of ast_str internals)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@163991 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-12-13 08:36:35 +00:00
Terry Wilson
74de8fdaa7 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: https://origsvn.digium.com/svn/asterisk/trunk@163873 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-12-12 23:48:26 +00:00
Mark Michelson
1772fc56f0 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: https://origsvn.digium.com/svn/asterisk/trunk@163085 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-12-11 16:47:34 +00:00
Mark Michelson
cda010c3b7 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: https://origsvn.digium.com/svn/asterisk/trunk@163081 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-12-11 16:33:16 +00:00
Eliel C. Sardanons
1e8e12efcf Janitor, use ARRAY_LEN() when possible.
(closes issue #13990)
Reported by: eliel
Patches:
      array_len.diff uploaded by eliel (license 64)



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@161218 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-12-05 10:31:25 +00:00
Mark Michelson
a53877b469 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: https://origsvn.digium.com/svn/asterisk/trunk@160626 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-12-03 18:37:46 +00:00
Mark Michelson
ac1b520de6 When investigating issue #13548, I found that gosub
handling in app_queue was just completely wrong, mostly
because the channel operations being performed were being
done on the incorrect channel.

With this set of changes, a gosub will correctly run on
the answering queue member's channel. There are still crash
issues which occur if there are dialplan syntax errors, so
I cannot yet close the referenced issue.



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@160555 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-12-03 17:07:09 +00:00
Sean Bright
fd8caa1778 This is basically a complete rollback of r155401, as it was determined that
it would be best to maintain API compatibility.  Instead, this commit introduces
ao2_callback_data() which is functionally identical to ao2_callback() except
that it allows you to pass arbitrary data to the callback.

Reviewed by Mark Michelson via ReviewBoard:
	http://reviewboard.digium.com/r/64


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@158959 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-11-25 01:01:49 +00:00
Matthew Nicholson
17ed84ff07 Make the Join event from app_queue use CallerIDNum insead of CallerID for
indicating the callerid number just like the rest of asterisk.

(closes issue #13883)
Reported by: davidw


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@158924 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-11-25 00:05:41 +00:00
Mark Michelson
9e1283e160 Add a space to the output
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@157940 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-11-19 23:30:42 +00:00
Mark Michelson
de5ba432da Add a RES_NOT_DYNAMIC case for the CLI command
'queue remove member'



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@157939 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-11-19 23:29:14 +00:00
Mark Michelson
cf6c66de65 Fix some refcounting in app_queue.c and change the
hashing used by app_queue.c to be case-insensitive.
This is accomplished by adding a new case-insensitive
hashing function.

This was necessary to prevent bad refcount errors
(and potential crashes) which would occur due to the
fact that queues were initially read from the config
file in a case-sensitive manner. Then, when a user
issued a CLI command or manager action, we allowed
for case-insensitive input and used that input to 
directly try to find the queue in the hash table. The result
was either that we could not find a queue that was input or
worse, we would end up hashing to a completely bogus value
based on the input.

This commit resolves the problem presented in
issue #13703. However, that issue was reported against
1.6.0. Since this fix introduces a behavior change, I am
electing to not place this same fix in to the 1.6.0 or 1.6.1
branches, and instead will opt for a change which does not
change behavior.



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@156883 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-11-14 16:53:38 +00:00
Sean Bright
9ef09ad1d4 Merged revisions 155553 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
r155553 | seanbright | 2008-11-08 20:08:07 -0500 (Sat, 08 Nov 2008) | 6 lines

Use static functions here instead of nested ones.  This requires a small
change to the ast_bridge_config struct as well.  To understand the reason
for this change, see the following post:

    http://gcc.gnu.org/ml/gcc-help/2008-11/msg00049.html

........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@155554 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-11-09 01:27:00 +00:00
Sean Bright
30d1744ffc Add ability to pass arbitrary data to the ao2_callback_fn (called from
ao2_callback and ao2_find).  Currently, passing OBJ_POINTER to either
of these mandates that the passed 'arg' is a hashable object, making
searching for an ao2 object based on outside criteria difficult.

Reviewed by Russell and Mark M. via ReviewBoard:
    http://reviewboard.digium.com/r/36/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@155401 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-11-07 22:39:30 +00:00
Eliel C. Sardanons
d23dff9ca8 - Add some see-also references based on TFOT.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@154542 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-11-05 12:13:57 +00:00
Tilghman Lesher
2cc8e25222 Slightly optimize ast_devstate_str and rename global functions devstate2str and config_text_file_save to have an ast_ prefix
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@154260 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-11-04 18:47:20 +00:00
Kevin P. Fleming
bd4eb070f3 bring over all the fixes for the warnings found by gcc 4.3.x from the 1.4 branch, and add the ones needed for all the new code here too
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@153616 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-11-02 18:52:13 +00:00
Russell Bryant
5b168ee34b Merge changes from team/group/appdocsxml
This commit introduces the first phase of an effort to manage documentation of the
interfaces in Asterisk in an XML format.  Currently, a new format is available for
applications and dialplan functions.  A good number of conversions to the new format
are also included.

For more information, see the following message to asterisk-dev:

http://lists.digium.com/pipermail/asterisk-dev/2008-October/034968.html


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@153365 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-11-01 21:10:07 +00:00
Terry Wilson
5fe37e47c6 Recent CDR fixes moved execution of the 'h' exten into the bridging code, so variables that were set after ast_bridge_call was called would not show up in the 'h' exten. Added a callback function to handle setting variables, etc. from w/in the bridging code. Calls back into a nested function within the function calling ast_bridge_call
(closes issue #13793)
Reported by: greenfieldtech


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@153181 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-10-31 18:55:33 +00:00
Steve Murphy
d736ac2b19 Merged revisions 152538 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
r152538 | murf | 2008-10-28 23:19:04 -0600 (Tue, 28 Oct 2008) | 14 lines

A little documentation cross-ref between features and
dial and queue... I wasted some time (stupidly) trying
to get the one-touch parking stuff working, because it
didn't occur to me that I had to also have the corresponding
options in the dial command! Duh! (In all this time, I never
set this up before!)
So, to keep some poor fool from suffering the same fate,
I made the features.conf.sample file mention the corresponding
opts in dial/queue; and the docs for dial/app specifically
mention the corresponding decls in the feature.conf file.

I hope this doesn't spoil some vast, eternal plan...


........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@152605 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-10-29 05:47:13 +00:00
Steve Murphy
6fad66dfb3 Merged revisions 152535 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
r152535 | murf | 2008-10-28 22:36:32 -0600 (Tue, 28 Oct 2008) | 46 lines

The magic trick to avoid this crash is not to
try to find the channel by name in the list,
which is slow and resource consuming, but rather
to pay attention to the result codes from the
ast_bridge_call, to which I added the 
AST_PBX_NO_HANGUP_PEER_PARKED value, which
now are returned when a channel is parked.
Why? because CDR's aren't generated via parking,
so nothing is needed, but if a transfer occurred,
there are critical things I need.

If you get AST_PBX_KEEPALIVE,
then don't touch the channel pointer.

If you get AST_PBX_NO_HANGUP_PEER, or
AST_PBX_NO_HANGUP_PEER_PARKED, then don't
touch the peer pointer.

Updated the several places where the results
from a bridge were not being properly obeyed,
and fixed some code I had introduced so that
the results of the bridge were not overridden 
(in trunk).

All the places that previously tested for 
AST_PBX_NO_HANGUP_PEER now have to check for
both AST_PBX_NO_HANGUP_PEER and AST_PBX_NO_HANGUP_PEER_PARKED.

I tested this against the 4 common parking
scenarios:


1. A calls B; B answers; A parks B; B hangs up while A is getting the parking
slot announcement, immediately after being put on hold.

2. A calls B; B answers; A parks B; B hangs up after A has been hung up, but
before the park times out.

3. A calls B; B answers; B parks A; A hangs up while B is getting the parking slot announcement, immediately after being put on hold.

4. A calls B; B answers; B parks A; A hangs up after B has been hung up, but before the park times out.


No crash.

I also ran the scenarios above against valgrind, and accesses looked good.



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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@152536 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-10-29 05:01:00 +00:00
BJ Weschke
7a8344bac6 The QueueEntry event now has the uniqueid of the channel included.
(closes issue #13731)
 reported and patched by: caio1982



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@150773 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-10-18 00:25:18 +00:00
Mark Michelson
184fe6f87b Merged revisions 149200 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r149200 | mmichelson | 2008-10-14 17:40:42 -0500 (Tue, 14 Oct 2008) | 12 lines

Update the queue with the correct number of calls and
whether the call was completed within the service level
when a transfer takes place. This way, we do not "break"
the leastrecent and fewestcalls strategies by not logging
a call until after the transferred call has ended.

(closes issue #13395)
Reported by: Marquis
Patches:
      app_queue.c.transfer.patch uploaded by Marquis (license 32)


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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@149201 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-10-14 22:41:13 +00:00
Mark Michelson
b8aed684f5 This commit introduces a change to how the "joinempty"
and "leavewhenempty" options are configured in queues.conf.

Instead of using vague terms like "yes," "no," "loose," and
"strict," we now accept a comma-separated list of values
to determine when to consider a member available.

Extended details can be found in the queues.conf.sample
file. Note also that the above four referenced values are
still accepted for backwards-compatibility, but are mapped
internally to the new method of representing the option.

AST-105



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@146640 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-10-06 15:29:56 +00:00
Michiel van Baak
90751b16ca Merge the cli_cleanup branch.
This work is done by lmadsen, junky and mvanbaak
during AstriDevCon.

This is the second audit the CLI got, and
this time lmadsen made sure he had _ALL_ modules
loaded that have CLI commands in them.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@145121 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-09-28 23:32:14 +00:00
Michiel van Baak
f1b673eef0 fix a couple of CLI commands that did not have a help description.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@144879 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-09-27 00:49:24 +00:00
Tilghman Lesher
08af5bb312 Create a new config file status, CONFIG_STATUS_FILEINVALID for differentiating
when a file is invalid from when a file is missing.  This is most important when
we have two configuration files.  Consider the following example:

Old system:
sip.conf     users.conf     Old result               New result
========     ==========     ==========               ==========
Missing      Missing        SIP doesn't load         SIP doesn't load
Missing      OK             SIP doesn't load         SIP doesn't load
Missing      Invalid        SIP doesn't load         SIP doesn't load
OK           Missing        SIP loads                SIP loads
OK           OK             SIP loads                SIP loads
OK           Invalid        SIP loads incompletely   SIP doesn't load
Invalid      Missing        SIP doesn't load         SIP doesn't load
Invalid      OK             SIP doesn't load         SIP doesn't load
Invalid      Invalid        SIP doesn't load         SIP doesn't load

So in the case when users.conf doesn't load because there's a typo that
disrupts the syntax, we may only partially load users, instead of failing with
an error, which may cause some calls not to get processed.  Worse yet, the old
system would do this with no indication that anything was even wrong.

(closes issue #10690)
 Reported by: dtyoo
 Patches: 
       20080716__bug10690.diff.txt uploaded by Corydon76 (license 14)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@142992 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-09-12 23:30:03 +00:00
Steve Murphy
67f7ac0499 Merged revisions 142675 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
r142675 | murf | 2008-09-11 22:29:34 -0600 (Thu, 11 Sep 2008) | 29 lines

Tested by: sergee, murf, chris-mac, andrew, KNK

This is a "second attempt" to restore the previous "endbeforeh" behavior
in 1.4 and up. In order to capture information concerning all the
legs of transfers in all their infinite combinations, I was forced
to this particular solution by a chain of logical necessities, the
first being that I was not allowed to rewrite the CDR mechanism from 
the ground up!

This change basically leaves the original machinery alone, which allows
IVR and local channel type situations to generate CDR's as normal, but
a channel flag can be set to suppress the normal running of the h exten.
That flag would be set by the code that runs the h exten from the
ast_bridge_call routine, to prevent the h exten from being run twice.
Also, a flag in the ast_bridge_config struct passed into ast_bridge_call
can be used to suppress the running of the h exten in that routine. This
would happen, for instance, if you use the 'g' option in the Dial app.

Running this routine 'early' allows not only the CDR() func to be used
in the h extension for reading CDR variables, but also allows them to
be modified before the CDR is posted to the backends.

While I dearly hope that this patch overcomes all problems, and 
introduces no new problems, reality suggests that surely someone
will have problems. In this case, please re-open 13251 (or 13289),
and we'll see if we can't fix any remaining issues.

** trunk note: some code to suppress the h exten being run 
from app_queue was added; for the 'continue' option available
only in trunk/1.6.x.


........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@142676 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-09-12 04:50:48 +00:00
Mark Michelson
4ad993ed8f This is the trunk version of the patch to close
issue 12979. The difference between this and the
1.6.0 and 1.6.1 versions is that this is a much more
invasive change. With this, we completely get rid
of the interfaces list, along with all its helper
functions.

Let me take a moment to say that this change personally
excites me since it may mean huge steps forward regarding
proper lock order in app_queue without having to strew
seemingly unnecessary locks all over the place. It also
results in a huge reduction in lines of code and complexity.
Way to go Brett!

(closes issue #12979)
Reported by: sigxcpu
Patches:
      20080710_issue12979_queue_custom_state_interface_trunk_2.diff uploaded by bbryant (license 36)
Tested by: sigxcpu, putnopvut



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@142146 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-09-09 17:15:29 +00:00
Mark Michelson
16d2c41701 Use ast_debug for debug messages. I was wondering why debug
messages weren't showing up when I had set the debug level
high for just app_queue.c. It's because we were only checking
the global option_debug variable instead of using the awesome
macro which checks both the global and file-specific value



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@141998 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-09-09 12:32:38 +00:00
Mark Michelson
4356d794e3 Optimization: The only reason we should check member
status is if the queue has a joinempty or a leavewhenempty
setting which could cause the caller to not join the queue
or exit the queue. Prior to this patch, we could potentially
traverse the entire queue's member list for no reason since
even if the members are currently not available in some way
we're going to let the caller join the queue anyway.



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@141906 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-09-08 23:00:06 +00:00
Mark Michelson
43285b9273 Fix some locking order issues in app_queue. This was
brought up by atis on IRC a while ago.



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@140975 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-09-03 20:16:12 +00:00
Mark Michelson
5dfefa5ee6 Merged revisions 140488 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
r140488 | mmichelson | 2008-08-29 12:34:17 -0500 (Fri, 29 Aug 2008) | 22 lines

After working on the ao2_containers branch, I noticed
something a bit strange. In all cases where we provide
a callback function to ao2_container_alloc, the callback
function would only return 0 or CMP_MATCH. After inspecting
the ao2_callback() code carefully, I found that if you're
only looking for one specific item, then you should return
CMP_MATCH | CMP_STOP. Otherwise, astobj2 will continue
traversing the current bucket until the end searching for
more matches.

In cases like chan_iax2 where in 1.4, all the peers are
shoved into a single bucket, this makes for potentially
terrible performance since the entire bucket will be
traversed even if the peer is one of the first ones come
across in the bucket.

All the changes I have made were for cases where the 
callback function defined was passed to ao2_container_alloc
so that calls to ao2_find could find a unique instance
of whatever object was being stored in the container.


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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@140489 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-08-29 17:47:17 +00:00
Mark Michelson
e833fd2ab1 Add output of variables to AgentRingNoAnswer manager
event if eventwhencalled is set to "vars" in 
queues.conf. Yay for consistency.

(closes issue #13369)
Reported by: srt
Patches:
      13369_agentringnoanswer_variables.diff uploaded by srt (license 378)



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@139832 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-08-25 17:24:02 +00:00
Mark Michelson
612f8c85b4 Change the queue timeout priority logic into less ugly
and confusing code pieces. Clarify the logic within
queues.conf.sample.

(closes issue #12690)
Reported by: atis
Patches:
      queue_timeoutpriority.patch uploaded by atis (license 242)



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@138694 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-08-18 20:23:11 +00:00
Mark Michelson
16a0e23110 Merged revisions 138685 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
r138685 | mmichelson | 2008-08-18 15:01:14 -0500 (Mon, 18 Aug 2008) | 10 lines

Change the inequalities used in app_queue with regards
to timeouts from being strict to non-strict for more
accuracy.

(closes issue #13239)
Reported by: atis
Patches:
      app_queue_timeouts_v2.patch uploaded by atis (license 242)


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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@138687 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-08-18 20:04:10 +00:00