Commit Graph

226 Commits

Author SHA1 Message Date
Olle Johansson f04d5fb83d Make sure we can set up calls without audio (text+video).
And ... it works!



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@103772 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-02-18 17:12:13 +00:00
Tilghman Lesher 26755e3882 Context tracing for channels
(closes issue #11268)
 Reported by: moy
 Patches: 
       chantrace-datastored-encapsulated-rev94934.patch uploaded by moy (license 222)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@103754 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-02-18 04:43:33 +00:00
Jason Parker 72bc8a7c7d Fix up some doxygen issues.
(closes issue #11996)
Patches:
      bug_11996_doxygen.diff uploaded by snuffy (license 35)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@103723 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-02-15 17:29:08 +00:00
Joshua Colp 9587f09103 Add support for allowing a native bridge to happen when the L option is enabled. The RTP bridging could already handle this, it just needed to be enabled in the main bridging code.
(issue #10647)
Reported by: samdell3


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@103314 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-02-11 16:17:15 +00:00
Russell Bryant 1c74c549d7 Merged revisions 100581 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r100581 | russell | 2008-01-28 11:15:41 -0600 (Mon, 28 Jan 2008) | 9 lines

Make some deadlock related fixes.  These bugs were discovered and reported
internally at Digium by Steve Pitts.
 - Fix up chan_local to ensure that the channel lock is held before the local
   pvt lock.
 - Don't hold the channel lock when executing the timing function, as it can
   cause a deadlock when using chan_local.  This actually changes the code back
   to be how it was before the change for issue #10765.  But, I added some other
   locking that I think will prevent the problem reported there, as well.

........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@100582 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-01-28 17:21:24 +00:00
Russell Bryant 87800e1913 Make a couple more uses of ARRAY_LEN, and convert some spaces to tabs
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@100533 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-01-28 04:43:14 +00:00
Russell Bryant e8f7cbf358 - Simplify a line with ARRAY_LEN()
- Make a few little formatting changes


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@100532 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-01-28 04:30:44 +00:00
Russell Bryant 79be2b8b0e These readlocks always fail for me on my mac, and I saw it happen again
today on another mac.  We ignore the return value of locking operations almost
everywhere in Asterisk.  So, ignore these, as well, so Asterisk will actually
work on systems where this is occurring while I look into what the issue is.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@100514 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-01-28 00:56:14 +00:00
Joshua Colp 1d47c036dd Some more cosmetic changes.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@100095 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-01-24 03:34:57 +00:00
Joshua Colp 15515ec671 Add some spacing.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@100094 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-01-24 03:30:56 +00:00
Olle Johansson b8aa3248ec Add a generic function to set the bridged call PVT unique id string
as a channel variable BRIDGEPVTCALLID

This is important for call tracing in log files and CDRs, so that
the SIP callID can be traced along servers.

The CHANNEL dialplan function won't work here, since the outbound
channel is gone when we need the Call-ID.

Other channel drivers may now implement the same function :-),
but this patch only supports chan_sip.so.

Inspired by (issue #11816)
Reported by: ctooley

Patch by oej



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@99644 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-01-22 20:35:10 +00:00
Russell Bryant d09fdcf61d Add AST_FORMAT_SLINEAR16 to the list for ast_best_codec()
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@99008 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-01-17 23:20:37 +00:00
Mark Michelson e04aa9eed4 Merged revisions 98315 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r98315 | mmichelson | 2008-01-11 13:10:57 -0600 (Fri, 11 Jan 2008) | 5 lines

Properly report the hangup cause as no answer when someone does not answer

(closes issue #10574, reported by boch, patched by moy)


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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@98316 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-01-11 19:12:05 +00:00
Tilghman Lesher afb2031389 Allow the uniqueid to be used for searching for a channel in the list.
Reported and initially patched by: michael-fig
(Closes issue #11340)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@96301 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-01-03 21:27:57 +00:00
Russell Bryant b749217bcb Merged revisions 95024 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r95024 | russell | 2007-12-27 15:40:02 -0600 (Thu, 27 Dec 2007) | 9 lines

Don't report a syntax error when an empty string is passed to ast_get_group.
Just return 0.

(closes issue #11540)
Reported by: tzafrir
Patches: 
      group_empty.diff uploaded by tzafrir (license 46)
	   -- slightly changed by me

........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@95025 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-12-27 21:41:22 +00:00
Russell Bryant f853918c53 Leave a note for a minor bug that was pointed out by d1mas
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@94796 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-12-26 18:19:46 +00:00
Tilghman Lesher 9c1dcfea5a Merged revisions 94767 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r94767 | tilghman | 2007-12-24 13:36:59 -0600 (Mon, 24 Dec 2007) | 5 lines

Race: we need to wait to queue a NewChannel event until after the channel is
inserted into the channel list.  The reason is because some manager users
immediately queue requests from the channel when they see that event and are
confused when Asterisk reports no such channel. (Closes issue #11632)

........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@94768 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-12-24 19:43:36 +00:00
Mark Michelson 5e7f167073 Merged revisions 93625 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r93625 | mmichelson | 2007-12-18 11:02:48 -0600 (Tue, 18 Dec 2007) | 6 lines

Rework deadlock avoidance used in ast_write, since it meant that agent channels which were being monitored
had one audio file recorded and one empty audio file saved.

(closes issue #11529, reported by atis patched by me)


........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@93626 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-12-18 17:05:19 +00:00
Olle Johansson 1e07078c60 Changes to manager events in channel.c
- Newstate event
        - Now has "CalleridNum" for numeric caller id, like Newchannel
        - The event does not send "<unknown>" for unknown caller IDs just an empty field
- Newstate and Newchannel events
        - these have changed headers
        "State"         -> ChannelStateDesc     Text based channel state
                        -> ChannelState         Numeric channel state
        - The events does not send "<unknown>" for unknown caller IDs just an empty field

- Newstate event
        - Now has "CalleridNum" for numeric caller id, like Newchannel
        - The event does not send "<unknown>" for unknown caller IDs just an empty field

- Link and Unlink events
        - The "Link" and "Unlink" bridge events in channel.c are now renamed to "Bridge"
        - The link state is in the bridgestate: header as "Link" or "Unlink"
        - For channel.c bridges, "Bridgetype: core" is added. This opens up for
          bridge events in rtp.c and channel drivers

- The "Rename" manager event has a renamed header, to use the same
        terminology for the current channel as other events
        - Oldname       -> Channel

(Moremanager)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@91409 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-12-06 15:29:05 +00:00
Joshua Colp 3faa47ab5b Instead of iterating through the entire epoll events array just look at the ones that will actually contain data. (props to eliel on IRC for this)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@91328 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-12-06 04:37:36 +00:00
Jason Parker 0ef0259302 Merged revisions 90876 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

(closes issue #11454)
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r90876 | qwell | 2007-12-04 11:28:08 -0600 (Tue, 04 Dec 2007) | 4 lines

If we fail to create a channel after allocating a timing fd, we need to make sure to close it.

Issue 11454, patch by eliel.

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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@90879 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-12-04 17:38:51 +00:00
Tilghman Lesher 0147028b51 Avoid an additional function call. Reported by: eliel Patch by: eliel (Closes issue #11438)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@90591 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-12-03 20:30:02 +00:00
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.

........


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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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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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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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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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.



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@77793 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@77781 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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");


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@77299 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@75983 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@75585 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@75400 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@69988 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-06-19 16:25:57 +00:00
Russell Bryant b179e2155f Convert uses of strdup() to ast_strdup()
(issue #9983, eliel)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@69436 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@69011 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-06-12 19:19:09 +00:00
Joshua Colp b3475e429a Minor code cleanup.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@68901 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-06-12 13:58:13 +00:00
Joshua Colp a9043635ed Change channel list to read/write list... I'm crazy.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@68685 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@68684 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@68158 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@67864 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@67717 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@66077 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@66072 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@65505 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@64242 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@64158 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-05-14 10:40:50 +00:00
Joshua Colp 9427cf5dd6 Merged revisions 63698 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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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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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@63699 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-05-09 19:24:27 +00:00