Commit Graph

204 Commits

Author SHA1 Message Date
Joshua Colp 46d2c050c5 Merged revisions 90548 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r90548 | file | 2007-12-03 14:40:56 -0400 (Mon, 03 Dec 2007) | 2 lines

Preserve the indication currently playing on a channel when a masquerade operation happens. (issue #BE-88)

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@90550 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-12-03 18:44:16 +00:00
Russell Bryant 53a5f22849 Merged revisions 90145 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r90145 | russell | 2007-11-28 18:20:34 -0600 (Wed, 28 Nov 2007) | 5 lines

This set of changes is to make some callerID handling thread-safe.
The ast_set_callerid() function needed to lock the channel.  Also, the handlers
for the CALLERID() dialplan function needed to lock the channel when reading
or writing callerid values directly on the channel structure.

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@90146 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-29 00:28:10 +00:00
Joshua Colp 867b734868 Merged revisions 89577 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r89577 | file | 2007-11-26 11:34:38 -0400 (Mon, 26 Nov 2007) | 6 lines

If channel allocation fails because the alert pipe could not be created also free the scheduler context.
(closes issue #11355)
Reported by: eliel
Patches:
      main.channel.c.patch uploaded by eliel (license 64)

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89578 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-26 15:36:27 +00:00
Olle Johansson 595961655a Try to get channel.h and channel.c aligned in regards to ast_set_callerid as well
as change name of variables to follow the rest of the naming.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89564 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-25 19:33:33 +00:00
Luigi Rizzo e0ff5fef5c remove a bunch of useless #include "options.h"
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89511 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-21 23:09:02 +00:00
Luigi Rizzo a23c055c3d move asterisk/paths.h outside asterisk.h and into those files
who really need it.



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89466 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-20 23:16:15 +00:00
Luigi Rizzo 915b97d300 move internal function declarations to include/asterisk/_private.h
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89465 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-20 22:18:21 +00:00
Luigi Rizzo 9335ace850 another bunch of include removals (errno.h and asterisk/logger.h)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89425 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-19 19:09:03 +00:00
Luigi Rizzo fdb7f7ba3d Start untangling header inclusion in a way that does not affect
build times - tested, there is no measureable difference before and
after this commit.

In this change:

use asterisk/compat.h to include a small set of system headers:
inttypes.h, unistd.h, stddef.h, stddint.h, sys/types.h, stdarg.h,
stdlib.h, alloca.h, stdio.h

Where available, the inclusion is conditional on HAVE_FOO_H as determined
by autoconf.

Normally, source files should not include any of the above system headers,
and instead use either "asterisk.h" or "asterisk/compat.h" which does it
better. 

For the time being I have left alone second-level directories
(main/db1-ast, etc.).



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89333 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-16 20:04:58 +00:00
Luigi Rizzo 09d9cce1d8 access channel locks through ast_channel_lock/unlock/trylock and not
through ast_mutex primitives.

To detect all occurrences, I have renamed the lock field in struct ast_channel
so it is clear that it shouldn't be used directly.

There are some uses in res/res_features.c (see details of the diff)
that are error prone as they try and lock two channels without
caring about the order (or without explaining why it is safe).



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89293 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-15 16:20:47 +00:00
Tilghman Lesher 5112f2b7e9 Typo
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89274 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-14 23:08:54 +00:00
Tilghman Lesher 36d9a5aaf3 Add callerid to the Hangup manager event.
Reported by: outtolunc
Patch by: outtolunc
Closes issue #11248


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89273 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-14 22:59:05 +00:00
Kevin P. Fleming a34bbab643 use simpler technique for removing known entries from lists
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89259 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-14 00:54:38 +00:00
Tilghman Lesher 061e5a1674 Merged revisions 89184 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r89184 | tilghman | 2007-11-12 11:29:17 -0600 (Mon, 12 Nov 2007) | 5 lines

Fix two cases of memory corruption caused by background threads.
Reported by: atis
Patch by: tilghman
Fixes issue #10923

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89185 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-12 17:44:04 +00:00
Kevin P. Fleming edc78d6023 improve linked-list macros in two ways:
- the *_CURRENT macros no longer need the list head pointer argument
  - add AST_LIST_MOVE_CURRENT to encapsulate the remove/add operation when moving entries between lists


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89106 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-08 05:28:47 +00:00
Joshua Colp 0971b51cfc Remove native bridging check for DTMF based transfers. Thanks to the last batch of RTP changes it is no longer required for the media stream to go through Asterisk if DTMF is going over signalling. It will simply reinvite back as needed.
(closes issue #11172)
Reported by: ibc


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89057 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-06 20:55:58 +00:00
Mark Michelson 57c4409689 Merged revisions 88826 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r88826 | mmichelson | 2007-11-05 17:29:29 -0600 (Mon, 05 Nov 2007) | 6 lines

Reworked deadlock avoidance in __ast_read. Restored audio to 
callback agents.

(closes issue #11071, reported by callguy, patched by me, tested by callguy and Ted Brown)


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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@88827 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-05 23:31:00 +00:00
Russell Bryant 7df9905e71 Merged revisions 88709 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r88709 | russell | 2007-11-05 14:11:04 -0600 (Mon, 05 Nov 2007) | 20 lines

Merge the last bit of changes from asterisk/team/russell/readq-1.4

The issue here is that the channel frame readq handling got broken when the
code was converted to use the linked list macros.  It caused corruption of the
list head and tail pointers.  So, I fixed up the usage of the linked list
macros and in passing, simplified the code.  I also documented what the code
is doing, as it was a bit difficult to figure out at first.

This bug showed itself with crashes showing messed up head/tail pointers for
the readq.  However, there are a couple of crashes that aren't quite as obvious,
but I think may be related.  So, if your bug gets closed by this commit, but
you still have a problem, please reopen or create a new bug report.

(closes issue #10936)
(closes issue #10595)
(closes issue #10368)
(closes issue #11084)
(closes issue #10040)
(closes issue #10840)

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@88710 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-05 20:14:48 +00:00
Russell Bryant 6677325afc Merged revisions 88624 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r88624 | russell | 2007-11-05 11:46:02 -0600 (Mon, 05 Nov 2007) | 5 lines

Fix up datastore handling in ast_do_masquerade().  The code is intended to move
any channel datastores from the old channel to the new one.  However, it did
not use the linked list macros properly to accomplish the task.  The existing
code would only work if there was only a single datastore on the old channel.

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@88651 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-05 17:52:08 +00:00
Luigi Rizzo 08b10da53b Simplify the implementation and the API for stringfields;
details and examples are in include/asterisk/stringfields.h.

Not applicable to older branches except for 1.4 which will
receive a fix for the routines that free memory pools.



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@88454 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-04 19:44:31 +00:00
Kevin P. Fleming c6a8d45f43 Merged revisions 87069 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r87069 | kpfleming | 2007-10-25 18:03:11 -0500 (Thu, 25 Oct 2007) | 2 lines

appending one list to another should leave the first list empty, and not require the user to do that

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@87070 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-10-25 23:11:28 +00:00
Jason Parker ebe4050128 Switch from AST_CLI (formerly NEW_CLI) to AST_CLI_DEFINE, since the former didn't make much sense
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@86820 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-10-22 20:05:18 +00:00
Russell Bryant a54eca55cf Merged revisions 86750 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r86750 | russell | 2007-10-22 10:52:48 -0500 (Mon, 22 Oct 2007) | 8 lines

Don't leak a frame in the case that an END frame is received and the time since
the BEGIN is less than that of the defined minimum DTMF duration.

(closes issue #11051)
Reported by: casper
Patches:
      channel.c.86664.diff uploaded by casper (license 55)

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@86751 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-10-22 15:57:02 +00:00
Joshua Colp 150964ab0e Merged revisions 86663 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r86663 | file | 2007-10-22 11:27:03 -0300 (Mon, 22 Oct 2007) | 6 lines

Move log message to before the frame it references is freed.
(closes issue #11050)
Reported by: slavon
Patches:
      channel.c.86662.diff uploaded by casper (license 55)

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@86664 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-10-22 14:28:07 +00:00
Jason Parker b0f3e6097e Convert NEW_CLI to AST_CLI.
Closes issue #11039, as suggested by seanbright.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@86536 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-10-19 18:29:40 +00:00
Russell Bryant 155aaf947f Merged revisions 86330 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r86330 | russell | 2007-10-18 13:03:10 -0500 (Thu, 18 Oct 2007) | 10 lines

The channel needs to stay locked while running timer callbacks, as they access
and modify channel data that may change elsewhere.  I went through every timer
callback in the source tree to make sure that none of them did any additional
locking that could introduce deadlocks, and all is well.

(closes issue #10765)
Reported by: Ivan
Patches:
      ast_1_4_11_svn_patch_channel_rc.diff uploaded by Ivan (license 229)

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@86331 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-10-18 18:06:49 +00:00
Russell Bryant e97a723cf1 Merge a ton of NEW_CLI conversions. Thanks to everyone that helped out! :)
(closes issue #10724)
Reported by: eliel
Patches: 
      chan_skinny.c.patch uploaded by eliel (license 64)
      chan_oss.c.patch uploaded by eliel (license 64)
      chan_mgcp.c.patch2 uploaded by eliel (license 64)
      pbx_config.c.patch uploaded by seanbright (license 71)
      iax2-provision.c.patch uploaded by eliel (license 64)
      chan_gtalk.c.patch uploaded by eliel (license 64)
      pbx_ael.c.patch uploaded by seanbright (license 71)
      file.c.patch uploaded by seanbright (license 71)
      image.c.patch uploaded by seanbright (license 71)
      cli.c.patch uploaded by moy (license 222)
      astobj2.c.patch uploaded by moy (license 222)
      asterisk.c.patch uploaded by moy (license 222)
      res_limit.c.patch uploaded by seanbright (license 71)
      res_convert.c.patch uploaded by seanbright (license 71)
      res_crypto.c.patch uploaded by seanbright (license 71)
      app_osplookup.c.patch uploaded by seanbright (license 71)
      app_rpt.c.patch uploaded by seanbright (license 71)
      app_mixmonitor.c.patch uploaded by seanbright (license 71)
      channel.c.patch uploaded by seanbright (license 71)
      translate.c.patch uploaded by seanbright (license 71)
      udptl.c.patch uploaded by seanbright (license 71)
      threadstorage.c.patch uploaded by seanbright (license 71)
      db.c.patch uploaded by seanbright (license 71)
      cdr.c.patch uploaded by moy (license 222)
      pbd_dundi.c.patch uploaded by moy (license 222)
      app_osplookup-rev83558.patch uploaded by moy (license 222)
      res_clioriginate.c.patch uploaded by moy (license 222)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@85460 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-10-11 19:03:06 +00:00
Tilghman Lesher e9221b062e Merged revisions 85158 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r85158 | tilghman | 2007-10-09 16:55:06 -0500 (Tue, 09 Oct 2007) | 5 lines

This commit fixes the following issues:
- Deadlock in ast_write (issue #10406)
- Deadlock in ast_read (issue #10406)
- Possible mutex initialization error in lock.h (issue #10571)

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@85176 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-10-09 22:21:49 +00:00
Joshua Colp 914cd4a7cf Merged revisions 84990 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r84990 | file | 2007-10-08 12:03:07 -0300 (Mon, 08 Oct 2007) | 4 lines

Don't keep trying to native bridge if either of the channels are involved in a masquerade operation to be done.
(closes issue #10696)
Reported by: tbelder

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@84991 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-10-08 15:05:05 +00:00
Dwayne M. Hubbard 71a0672560 Merged revisions 84274 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r84274 | dhubbard | 2007-10-01 16:25:37 -0500 (Mon, 01 Oct 2007) | 1 line

moved get_base_channel() code from action_redirect to ast_channel_masquerade() for issue 7706 and BE-160
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@84275 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-10-01 21:31:16 +00:00
Tilghman Lesher d3904a21c4 Merged revisions 84049 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r84049 | tilghman | 2007-09-28 00:30:22 -0500 (Fri, 28 Sep 2007) | 3 lines

Avoid a deadlock with ALL of the locks in the masquerade function, not just the
pairs of channels.  (Closes issue #10406)

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@84050 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-09-28 05:37:16 +00:00
Russell Bryant 9f64905d4e Merged revisions 83432 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r83432 | russell | 2007-09-21 09:37:20 -0500 (Fri, 21 Sep 2007) | 4 lines

gcc 4.2 has a new set of warnings dealing with cosnt pointers.  This set of
changes gets all of Asterisk (minus chan_alsa for now) to compile with gcc 4.2.
(closes issue #10774, patch from qwell)

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@83433 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-09-21 14:40:10 +00:00
Joshua Colp a565584d05 Fix memory issue that crept up with Russell's testing. It is *not* proper to free the frame we get in ast_write.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@81858 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-09-06 22:34:44 +00:00
Joshua Colp 352fd9d942 Merged revisions 81437 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r81437 | file | 2007-09-04 10:46:23 -0300 (Tue, 04 Sep 2007) | 4 lines

(closes issue #10476)
Reported by: mdu113
Only look for the end of a digit when waiting for a digit. This in turn disables emulation in the core.

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@81438 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-09-04 13:47:33 +00:00
Steve Murphy b5cd67adc3 This code was in team/murf/bug8684-trunk; it should fix bug 8684 in trunk. I didn't add it to 1.4 yet, because it's not entirely clear to me if this is a bug fix or an enhancement. A lot of files were affected by small changes like ast_variable_new getting an added arg, for the file name the var was defined in; ast_category_new gets added args of filename and lineno; ast_category and ast_variable structures now record file and lineno for each entry; a list of all #include and #execs in a config file (or any of its inclusions are now kept in the ast_config struct; at save time, each entry is put back into its proper file of origin, in order. #include and #exec directives are folded in properly. Headers indicating that the file was generated, are generated also for each included file. Some changes to main/manager.c to take care of file renaming, via the UpdateConfig command. Multiple inclusions of the same file are handled by exploding these into multiple include files, uniquely named. There's probably more, but I can't remember it right now.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@81361 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-08-29 20:55:40 +00:00
Russell Bryant 040a5f20f9 * Constify the uid field of channel datastores
* Convert some spaces to tabs in func_volume
* Add a note in channel.h making it clear that none of the datastore API calls
  lock the channel they are given, so the channel should be locked before
  calling the functions that take a channel argument.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@81260 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-08-28 18:32:56 +00:00
Joshua Colp 9264d5007a (closes issue #10427)
Reported by: pj
Two of the three places ast_waitfor_nandfds could branch off to did not clear outfd and exception. If the calling function did not clear these there was a chance they could get a false positive on testing to see whether they were set.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@79379 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-08-14 14:17:43 +00:00
Steve Murphy 4f98b05cd5 Don't ask me why, but waitfordigit will immediately return a 1 on my system, unless the outfd is initialized to -1 before calling the nandfds func
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@79378 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-08-14 13:46:58 +00:00
Russell Bryant 5f0c3e7dbc constify the return value of reason2str
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@79176 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-08-13 14:23:38 +00:00
Steve Murphy 526d1f39a2 Merged revisions 79099 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r79099 | murf | 2007-08-10 14:53:43 -0600 (Fri, 10 Aug 2007) | 1 line

From a user complaint on #asterisk, I have forced pbx_spool to explain what reason codes mean, when they are logged
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@79100 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-08-10 21:03:06 +00:00
Joshua Colp 22114b509d Add support for using epoll instead of poll. This should increase scalability and is done in such a way that we should be able to add support for other poll() replacements.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@78683 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-08-08 21:44:58 +00:00
Joshua Colp 602198c402 Merge audiohooks branch into trunk. This is a new API for developers to listen and manipulate the audio going through a channel.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@78649 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-08-08 19:30:52 +00:00
Joshua Colp 9ef1b0a974 Extend the ast_senddigit and ast_dtmf_stream API calls to allow the duration of the DTMF digit(s) to be specified and make the SendDTMF application have the capability to use it.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@78278 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-08-06 21:52:30 +00:00
Joshua Colp 431c60f818 Merged revisions 78275 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r78275 | file | 2007-08-06 18:41:13 -0300 (Mon, 06 Aug 2007) | 2 lines

Add additional DTMF log messages to help when debugging issues.

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@78276 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-08-06 21:43:09 +00:00
Luigi Rizzo f66dbeb8f3 print formats as 0x%x instead of %d in a warning message.
Being bitmasks, it is a lot easier to read this way.



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2007-07-30 19:42:25 +00:00
Russell Bryant ad094d5109 Merged revisions 77785 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r77785 | russell | 2007-07-30 13:55:15 -0500 (Mon, 30 Jul 2007) | 3 lines

file and I both committed changes for issue #10301.  Remove a duplicated
assignment to restore the original value of the previous channel.

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2007-07-30 18:56:29 +00:00
Russell Bryant a2fc80d4a6 Merged revisions 77780 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r77780 | russell | 2007-07-30 12:29:43 -0500 (Mon, 30 Jul 2007) | 16 lines

(closes issue #10301)
Reported by: fnordian
Patches:
      asterisk-1.4.9-channel.c.patch uploaded by fnordian (license 110)
      Additional changes by me

Fix some problems in channel_find_locked() which can cause an infinite loop.
The reference to the previous channel is set to NULL in some cases.  These changes
ensure that the reference to the previous channel gets restored before needing
it again.

I'm not convinced that the code that is setting it to NULL is really the right
thing to do.  However, I am making these changes to fix the obvious problem
and just leaving an XXX comment that it needs a better explanation that what
is there now.

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2007-07-30 17:31:29 +00:00
Joshua Colp 3733874f49 Merged revisions 77771 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r77771 | file | 2007-07-30 12:47:52 -0300 (Mon, 30 Jul 2007) | 6 lines

(closes issue #10301)
Reported by: fnordian
Patches:
      asterisk-1.4.9-channel.c.patch uploaded by fnordian (license 110)
Restore previous behavior where if we failed to lock the channel we wanted we would return to exactly the same point as if we had just reentered the function.

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2007-07-30 15:49:30 +00:00
Joshua Colp 7bd9b905e4 Merged revisions 77460 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r77460 | file | 2007-07-26 20:19:04 -0300 (Thu, 26 Jul 2007) | 4 lines

(closes issue #10302)
Reported by: litnialex
If a DTMF end frame comes from a channel without a begin and it is going to a technology that only accepts end frames (aka INFO) then use the minimum DTMF duration if one is not in the frame already.

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2007-07-26 23:20:25 +00:00
Russell Bryant f8483a0d04 Do a massive conversion for using the ast_verb() macro
(closes issue #10277, patches by mvanbaak)

Basically, this changes ...

if (option_verbose > 2)
   ast_verbose(VERBOSE_PREFIX_3, "Something\n");

to ...

ast_verb(3, "Something\n");


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2007-07-26 15:49:18 +00:00
Mark Michelson b37f7ab577 Merged revisions 77154 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r77154 | mmichelson | 2007-07-25 16:52:47 -0500 (Wed, 25 Jul 2007) | 3 lines

chan->emulate_dtmf_duration is an unsigned int, not a signed int, so use %u instead of %d in the format string


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2007-07-25 21:53:35 +00:00
Russell Bryant d5b4fde291 Merged revisions 76132 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r76132 | russell | 2007-07-20 13:22:24 -0500 (Fri, 20 Jul 2007) | 6 lines

Use the define that specifies the default length of an artificially created
DTMF digit in the ast_senddigit() function.  The define is set to 100ms by
default, which is the same thing that this function was using.  But, using
the define lets changes take effect in this case, as well as the others where
it was already used.

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2007-07-20 18:28:15 +00:00
Steve Murphy 0e969271ae After some study, thought, comparing, etc. I've backed out the previous universal mod to make ast_flags a 64 bit thing. Instead, I added a 64-bit version of ast_flags (ast_flags64), and 64-bit versions of the test-flag, set-flag, etc. macros, and an app_parse_options64 routine, and I use these in app_dial alone, to eliminate the 30-option limit it had grown to meet. There is room now for 32 more options and flags. I was heavily tempted to implement some of the other ideas that were presented, but this solution does not intro any new versions of dial, doesn't have a different API, has a minimal/zero impact on code outside of dial, and doesn't seriously (I hope) affect the code structure of dial. It's the best I can think of right now. My goal was NOT to rewrite dial. I leave that to a future, coordinated effort.
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2007-07-19 23:24:27 +00:00
Steve Murphy 5ac24b25d3 This corrects the problem with flags and %lld formats on 64-bit machines, where uint64_t is NOT acceptable for %lld, and also works on 32-bit machines. At least, with gcc.
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2007-07-18 14:35:07 +00:00
Steve Murphy 8a7732f067 via 10206, I have added an option (e) to Dial to allow the h exten to get run on peer. Had to upgrade ast_flag stuff to 64 bits to do this.
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2007-07-17 19:40:29 +00:00
Joshua Colp 1f86a8bc2d Merged revisions 74922 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r74922 | file | 2007-07-12 16:17:59 -0300 (Thu, 12 Jul 2007) | 2 lines

Whoops... didn't want this to be returned to 0 each iteration.

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2007-07-12 19:19:03 +00:00
Joshua Colp 8f3a5481dc Merged revisions 74888 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r74888 | file | 2007-07-12 14:16:28 -0300 (Thu, 12 Jul 2007) | 2 lines

When waiting for a digit ensure that a begin frame was received with it, not just an end frame. (issue #10084 reported by rushowr)

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2007-07-12 17:17:56 +00:00
Joshua Colp f4943f3211 Merged revisions 73355 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r73355 | file | 2007-07-05 11:21:44 -0300 (Thu, 05 Jul 2007) | 10 lines

Merged revisions 73349 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.2

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r73349 | file | 2007-07-05 11:19:14 -0300 (Thu, 05 Jul 2007) | 2 lines

Tweak spy locking. (issue #9951 reported by welles)

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2007-07-05 14:22:58 +00:00
Joshua Colp 73d33590ba Merged revisions 72888 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r72888 | file | 2007-07-02 11:32:59 -0300 (Mon, 02 Jul 2007) | 2 lines

Added additional DTMF debug messages for when emulation occurs.

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2007-07-02 14:39:49 +00:00
Joshua Colp 809c1398d7 Merged revisions 72257 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r72257 | file | 2007-06-27 16:25:24 -0400 (Wed, 27 Jun 2007) | 10 lines

Merged revisions 72256 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.2

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r72256 | file | 2007-06-27 16:23:24 -0400 (Wed, 27 Jun 2007) | 2 lines

I may possibly get shot for doing this... but... defer CDR processing until after the channel has been dealt with. This should eliminate all of the issues with channels going funky (SIP/PRI) when you are posting CDRs to a database that is either slow or unavailable and do not want to enable batching.

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2007-06-27 20:26:53 +00:00
Joshua Colp 7feaaaaf04 Merged revisions 72148 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r72148 | file | 2007-06-27 13:31:50 -0400 (Wed, 27 Jun 2007) | 2 lines

Make the ast_read_noaudio API call behave better under circumstances where DTMF emulation was happening and a generator was setup. (issue #10065 reported by stevefeinstein)

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2007-06-27 17:34:26 +00:00
Steve Murphy cd97d6a687 Merged revisions 70062 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r70062 | murf | 2007-06-19 12:23:23 -0600 (Tue, 19 Jun 2007) | 9 lines

Merged revisions 70053 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.2

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r70053 | murf | 2007-06-19 12:07:59 -0600 (Tue, 19 Jun 2007) | 1 line

This fixes 9246, where channel variables are not available in the 'h' exten, on a 'ZOMBIE' channel. The fix is to consolidate the channel variables during a masquerade, and then copy the merged variables back onto the clone, so the zombie has the same vars that the 'original' has.
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2007-06-19 18:31:29 +00:00
Joshua Colp afe63aec33 Merged revisions 69987 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r69987 | file | 2007-06-19 12:24:31 -0400 (Tue, 19 Jun 2007) | 10 lines

Merged revisions 69986 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.2

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r69986 | file | 2007-06-19 12:21:29 -0400 (Tue, 19 Jun 2007) | 2 lines

Update BRIDGEPEER variable if set to the new channel name when a masquerade happens. (issue #9699 reported by dimas)

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2007-06-19 16:25:57 +00:00
Russell Bryant b179e2155f Convert uses of strdup() to ast_strdup()
(issue #9983, eliel)


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2007-06-14 23:01:01 +00:00
Russell Bryant 055d82cbce Add a massive set of changes for converting to use the ast_debug() macro.
(issue #9957, patches from mvanbaak, caio1982, critch, and dimas)


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2007-06-14 19:39:12 +00:00
Russell Bryant cf06bdb312 Merged revisions 69010 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r69010 | russell | 2007-06-12 14:13:41 -0500 (Tue, 12 Jun 2007) | 12 lines

In ast_channel_make_compatible(), just return if the channels' read and write
formats already match up.  There are code paths that call this function on a
pair of channels multiple times.  This made calls fail that were using g729
in some cases.  The reason is that codec_g729a will unregister itself from the
list of available translators will all licenses are in use.  So, the first
time the function got called, the right translation path was allocated.
However, the second time it got called, the code would not find a translation
path to/from g729 and make the call fail, even if the channel actually already
had a g729 translation path allocated.

(SPD-32)

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2007-06-12 19:19:09 +00:00
Joshua Colp b3475e429a Minor code cleanup.
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2007-06-12 13:58:13 +00:00
Joshua Colp a9043635ed Change channel list to read/write list... I'm crazy.
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2007-06-11 14:41:39 +00:00
Joshua Colp eb56cb6b92 Merged revisions 68683 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r68683 | file | 2007-06-11 10:33:12 -0400 (Mon, 11 Jun 2007) | 10 lines

Merged revisions 68682 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.2

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r68682 | file | 2007-06-11 10:29:58 -0400 (Mon, 11 Jun 2007) | 2 lines

Improve deadlock handling of the channel list. (issue #8376 reported by one47)

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2007-06-11 14:35:02 +00:00
Joshua Colp ef7b38e065 Merged revisions 68157 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r68157 | file | 2007-06-07 14:39:52 -0400 (Thu, 07 Jun 2007) | 2 lines

Fix logic when doing a name based channel search for a structure when you want to start from a specific point in the channel list. (issue #9324 reported by slavon)

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2007-06-07 18:41:17 +00:00
Tilghman Lesher 9d05ff8ed5 Issue 9869 - replace malloc and memset with ast_calloc, and other coding guidelines changes
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2007-06-06 21:20:11 +00:00
Russell Bryant 18101ff0e3 Merged revisions 67716 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r67716 | russell | 2007-06-06 11:55:59 -0500 (Wed, 06 Jun 2007) | 13 lines

Merged revisions 67715 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.2

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r67715 | russell | 2007-06-06 11:40:51 -0500 (Wed, 06 Jun 2007) | 5 lines

We have some bug reports showing crashes due to a double free of a channel.
Add a sanity check to ast_channel_free() to make sure we don't go on trying
to free a channel that wasn't found in the channel list.
(issue #8850, and others...)

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2007-06-06 16:58:28 +00:00
Russell Bryant 2b25070414 Merged revisions 66076 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r66076 | russell | 2007-05-24 17:23:59 -0500 (Thu, 24 May 2007) | 1 line

if the string field init fails, clean up the stuff that was allocated already
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2007-05-24 22:25:55 +00:00
Russell Bryant 9065eb1054 Merged revisions 66070 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r66070 | russell | 2007-05-24 17:07:39 -0500 (Thu, 24 May 2007) | 2 lines

Check the result of ast_string_field_init() in ast_channel_alloc()

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2007-05-24 22:08:33 +00:00
Russell Bryant 90d6885701 Add a new feature for Music on Hold. If you set the "digit" option for a
class in musiconhold.conf, a caller on hold may press this digit to switch
to listening to that music class.
  This involved adding a new callback for generators, which allow generators
to get notified of DTMF from the channel they are running on.  Then, a callback
was implemented for the music on hold generators.
(patch from bbryant)


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2007-05-22 18:52:59 +00:00
Joshua Colp f02ea9e076 Merged revisions 64240 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r64240 | file | 2007-05-14 13:23:51 -0400 (Mon, 14 May 2007) | 2 lines

Fix scenario where if a phone that simply called Echo() put itself on hold it could never get off hold.

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2007-05-14 17:25:25 +00:00
Olle Johansson 78d3244554 Merged revisions 64157 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r64157 | oej | 2007-05-14 12:39:12 +0200 (Mon, 14 May 2007) | 2 lines

Add hangupcause when we lack codecs for transcoding

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2007-05-14 10:40:50 +00:00
Joshua Colp 9427cf5dd6 Merged revisions 63698 via svnmerge from
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r63698 | file | 2007-05-09 15:22:39 -0400 (Wed, 09 May 2007) | 2 lines

Use the DTMF frame on the channel when returning a DTMF frame from AST_FRAME_NULL or AST_FRAME_VOICE.

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2007-05-09 19:24:27 +00:00
Russell Bryant 56254ee491 Merged revisions 63612 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r63612 | russell | 2007-05-09 11:55:27 -0500 (Wed, 09 May 2007) | 5 lines

Modify ast_senddigit_begin() to use the same assumptions used elsewhere in the
code in that if a channel does not have a send_digit_begin() callback, it only
cares about DTMF END events.  (pointed out by Michael Neuhauser on the
asterisk-dev list)

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2007-05-09 19:21:35 +00:00
Russell Bryant 39dd02d542 Merged revisions 63608 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r63608 | russell | 2007-05-09 11:43:50 -0500 (Wed, 09 May 2007) | 5 lines

Only call ast_senddigit_begin() in ast_senddigit() if the channel has a 
send_digit_begin() callback.  Checking the END_DTMF_ONLY flag was the
wrong thing to do, because that flag indicates that a *bridged* channel
only wants DTMF END events coming from this channel.

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2007-05-09 16:44:33 +00:00
Joshua Colp 5394364048 Merged revisions 63286 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r63286 | file | 2007-05-07 17:45:01 -0400 (Mon, 07 May 2007) | 10 lines

Merged revisions 63285 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.2

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r63285 | file | 2007-05-07 17:39:52 -0400 (Mon, 07 May 2007) | 2 lines

Properly handle what happens during a masquerade in relation to group counting. (issue #9657 reported by ramonpeek)

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2007-05-07 21:47:08 +00:00
Russell Bryant 1e3b1a576c Merged revisions 62942 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r62942 | russell | 2007-05-03 10:23:13 -0500 (Thu, 03 May 2007) | 17 lines

Fix YADB (Yet Another DTMF Bug) ((C) Russell Bryant, 2007, TM, Patent Pending).

This set of changes came from a debugging session I had with Dwayne Hubbard.
When he called into his home FXO, ran the Echo application, and pressed a
digit, the digit would be echoed back and would never end.  This is fixed,
along with a couple other little improvements.

* When chan_zap is in the middle of playing a digit to a channel, it feeds
  back null frames, not voice frames.  So, I have modified ast_read to check
  the timing on emulated DTMF when it receives null frames, in addition to
  where it was doing this on voice frames.
* Make a tweak to setting the duration on emulated DTMF digits.  If there was
  no duration specified, it set it to be the minimum, instead of the default.
* Instead of timing the emulated digits off of the number of samples in audio
  frames that pass through, just use time values.  Now there is no code in this
  section that assumes 8kHz audio.

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2007-05-03 15:23:44 +00:00
Russell Bryant 255f151582 Merged revisions 62789 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r62789 | russell | 2007-05-02 17:59:09 -0500 (Wed, 02 May 2007) | 20 lines

Merge changes from team/russell/inband_dtmf ...

Fix some issues related to generating inband DTMF.  There are two changes here:

1)   The list of DTMF tones in the senddigit_begin() function explicitly
   specified 100ms of the tone followed by 100ms of silence.  This really
   broke things with the way that Asterisk now wants complete control
   over when the digit begins and ends.  So, regardless of what Asterisk
   really wanted to do, this was going to play out the tone at the length it
   wanted to.  This caused various problems like DTMF translation to inband to
   be extremely unreliable.
     The list of tones has been changed so that the correct DTMF tone is played
   indefinitely until Asterisk tells it to stop.

2) ast_write() had to be modified to let a DTMF_END frame get processed even
   when a generator is present.  This is how the tone will finally get stopped.

(issues #8944, #9250, #9348, maybe others.  Thanks to mdu113 from #8944 for
 the testing and feedback!)

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2007-05-02 23:00:07 +00:00
Steve Murphy fe7068a51b Merged revisions 62689 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r62689 | murf | 2007-05-02 11:10:50 -0600 (Wed, 02 May 2007) | 1 line

a)In chan_zap, set the clid, src fields in channel_alloc call. b)in the channel_alloc func, set the cid_num and name fields from the arglist[blush]. c) don't update the channel app & app data fields if you are in the 'h' extension. d)the load_module func in cdr_radius needs to return DECLINE, SUCCESS.
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2007-05-02 17:24:03 +00:00
Russell Bryant 007fa5e0be Merged revisions 62005 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r62005 | file | 2007-04-25 22:19:51 -0500 (Wed, 25 Apr 2007) | 2 lines

Missed an ast_app_group_discard during merge. Thanks blitzrage!

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2007-04-26 03:24:01 +00:00
Joshua Colp 8b2b3e172b Merged revisions 61805 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r61805 | file | 2007-04-25 15:21:54 -0400 (Wed, 25 Apr 2007) | 10 lines

Merged revisions 61804 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.2

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r61804 | file | 2007-04-25 14:52:50 -0400 (Wed, 25 Apr 2007) | 2 lines

Merge rewritten group counting support. No more storing data on the variable list of the channels. That was bad, mmmk? (issue #7497 reported by sabbathbh)

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2007-04-25 19:27:42 +00:00
Russell Bryant 94459660a3 Merged revisions 61781 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r61781 | russell | 2007-04-24 14:00:06 -0500 (Tue, 24 Apr 2007) | 6 lines

Improve DTMF handling in ast_read() even more in response to a discussion on
the asterisk-dev mailing list.  I changed the enforced minimum length of a
digit from 100ms to 80ms.  Furthermore, I made it now enforce a gap of 45ms in
between digits.  These values are not configurable in a configuration file
right now, but they can be easily changed near the top of main/channel.c.

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2007-04-24 19:03:16 +00:00
Russell Bryant fa0e814a69 Merged revisions 61763 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r61763 | russell | 2007-04-23 12:57:32 -0500 (Mon, 23 Apr 2007) | 4 lines

Ensure that digits passing through Asterisk have a reasonable minimum length.
It is currently 100 ms.  If someone thinks this should be different, feel free
to speak up.  (related to issues #8944, #9250, and #9348)

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2007-04-23 17:58:15 +00:00
Tilghman Lesher 47dd5a15af Issue 6082 - New DTMF event for manager
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2007-04-10 23:55:26 +00:00
Steve Murphy ecaf781933 Merged revisions 60989 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r60989 | murf | 2007-04-09 12:32:07 -0600 (Mon, 09 Apr 2007) | 1 line

This is a big improvement over the current CDR fixes. It may still need refinement, but this won't have as many folks bothered.
This also adds the mods from 1.4/r.61136;
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2007-04-10 05:41:34 +00:00
Steve Murphy 09c0d56c5c Merged revisions 59522 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r59522 | murf | 2007-03-30 11:51:17 -0600 (Fri, 30 Mar 2007) | 1 line

several changes via kpflemings review
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2007-03-30 17:57:47 +00:00
Steve Murphy 0f11d3c8c3 Merged revisions 59486 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r59486 | murf | 2007-03-30 08:11:59 -0600 (Fri, 30 Mar 2007) | 1 line

These mods fix CDR issues from 8221, 8593, 8680, 8743, and perhaps others. Mainly with CDRs generated from transfer situations.
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2007-03-30 14:37:21 +00:00
Tilghman Lesher 590cb3a6fa Expand datastores to add the notion of inheritance. This will be needed for
the conversion of IAX2 variables from the current custom method to ast_storage.


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2007-03-03 14:40:18 +00:00
Russell Bryant b4a29c3782 Constify the list of codec preferences.
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2007-03-01 20:24:59 +00:00
Tilghman Lesher b0f60e7496 Merged revisions 56685 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r56685 | tilghman | 2007-02-25 08:46:41 -0600 (Sun, 25 Feb 2007) | 11 lines

Merged revisions 56684 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.2

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r56684 | tilghman | 2007-02-25 08:38:03 -0600 (Sun, 25 Feb 2007) | 3 lines

Issue 9130 - If prev is the last item on the channel list, then evaluating
additional conditions (e.g. name prefix) will cause a NULL dereference.

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2007-02-25 14:53:40 +00:00
Olle Johansson 75d387acbc Doxygen additions, corrections
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2007-02-24 20:29:41 +00:00
Joshua Colp afc99294fa Merged revisions 56231 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r56231 | file | 2007-02-22 13:49:39 -0500 (Thu, 22 Feb 2007) | 10 lines

Merged revisions 56230 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.2

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r56230 | file | 2007-02-22 13:44:24 -0500 (Thu, 22 Feb 2007) | 2 lines

Only change the original or clone channel if it's the channel behind the proxy channel, not if it's just a regular bridged channel.

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2007-02-22 18:53:22 +00:00
Olle Johansson ba32ee49d0 Adding Realtime Text support (T.140) to Asterisk
T.140/RFC 2793 is a live communication channel, originally
created for IP based text phones for hearing impaired. 
Feels very much like the old Unix talk application.

This code is developed and disclaimed by John Martin of Aupix, UK.
Tested for interoperability by myself and Omnitor in Sweden,
the company that wrote most of the specifications.

A big thank you to everyone involved in this.


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2007-02-16 13:35:44 +00:00
Joshua Colp 8470ee5cd0 Merged revisions 54290 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r54290 | file | 2007-02-13 20:09:40 -0500 (Tue, 13 Feb 2007) | 2 lines

Add G722 to ast_best_codec. If anyone disagrees with it's placement, feel free to change it. (issue #9045 reported by gork)

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2007-02-14 01:12:21 +00:00
Russell Bryant 10c4a5fef7 Simplify a small bit of logic.
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2007-02-12 15:40:23 +00:00