not work for everyone, but it did for some. This set of changes makes trunk
start again for those having problems. Instead of building libresample as a
static library, it just links the object files in directly with the asterisk
binary.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@95864 65c4cc65-6c06-0410-ace0-fbb531ad65f3
gdb dying while debugging asterisk. The problem seems to be related
with a race in the handling of module_list, which in turn is triggeded
by calling dlopen() on a system which uses initializers to create
locks.
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res_resample, and mark codec_resample as dependent upon res_resample. This
prevents the linker from optimizing away libresample, and also makes it so the
libresample code isn't linked in to multiple places. (I have another module
in a branch that needs it, too.)
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r95577 | mmichelson | 2007-12-31 17:43:13 -0600 (Mon, 31 Dec 2007) | 9 lines
Avoiding a potentially bad locking situation. ast_merge_contexts_and_delete writelocks the conlock, then
calls ast_hint_extension, which attempts to readlock the same lock. Recursion with read-write locks is
dangerous, so the inner lock needs to be removed. I did this by copying the "guts" of ast_hint_extension
into ast_merge_contexts_and_delete (sans the extra lock).
(this change is inspired by the locking problems seen in issue #11080, but I have no idea if this is the
problematic area experienced by the reporters of that issue)
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This commit imports libresample for use in Asterisk. It also adds a new codec
module, codec_resample. This module uses libresample to re-sample signed linear
audio between 8 kHz and 16 kHz.
It also provides an alternative for converting between 16 kHz G.722 and 8 kHz
signed linear when using G.722, which will likely be useful as some people have
complained about volume issues when the current codec_g722 converts to 8 kHz
signed linear. But, to test this, you will have to disable the g722-to-slin and
g722-to-slin16 translators in codec_g722.c.
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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
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r94828 | russell | 2007-12-27 08:33:21 -0600 (Thu, 27 Dec 2007) | 9 lines
Change ast_translator_best_choice() to only pay attention to audio formats.
This fixes a problem where Asterisk claims that a translation path can not be
found for channels involving video.
(closes issue #11638)
Reported by: cwhuang
Tested by: cwhuang
Patch suggested by cwhuang, with some additional changes by me.
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r94829 | russell | 2007-12-27 08:44:29 -0600 (Thu, 27 Dec 2007) | 2 lines
Use the constant that I really meant to use here ...
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r94801 | russell | 2007-12-26 13:04:31 -0600 (Wed, 26 Dec 2007) | 4 lines
Just in case the AST_FLAG_END_DTMF_ONLY flag was already set before starting
autoservice, remember it and ensure that the channel has the same setting when
autoservice gets stopped. (pointed out by d1mas, patched up by me)
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r94797 | russell | 2007-12-26 12:46:39 -0600 (Wed, 26 Dec 2007) | 4 lines
When a channel is in autoservice, mark a flag on the channel that says that
we only care about the END of a digit. That way, no magic digit emulation stuff
will happen when all we're doing is queueing up END frames.
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r94790 | russell | 2007-12-26 11:06:26 -0600 (Wed, 26 Dec 2007) | 5 lines
Don't store DTMF BEGIN frames while a channel is in autoservice. It's just
going to make ast_read() do a lot of extra work when the channel comes back
out of autoservice.
(closes issue #11628, patched by me)
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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)
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r94468 | mmichelson | 2007-12-21 10:49:35 -0600 (Fri, 21 Dec 2007) | 6 lines
Since we are freeing list elements within a list traversal, we need to use the safe
traversal and remove the item from the list before freeing it.
(closes issue 11612, reported by dtyoo)
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so that paths and filename are writable by asterisk.c without
causing segfaults.
This involves defining the variables as const char *,
and having them point to as static, writable buffer
defined in asterisk.c
On passing, fix some errors in using these variables
in some files in utils/ , and in res/snmp/agent.c
which was redefining a variable without using paths.h
(not applicable to 1.4)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@94168 65c4cc65-6c06-0410-ace0-fbb531ad65f3
and likely in other places too.
This is quite useful when placing mail/html stuff in config files.
/*!
\brief Convert some C escape sequences (\b\f\n\r\t) into the
equivalent characters.
\brief s The string to be converted (will be modified).
\return The converted string.
*/
char *ast_unescape_c(char *s);
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Asterisk goes into a busy loop, the machine will be recoverable. We'd still
need to do a restart to put Asterisk back into high priority mode, but at
least a reboot won't be required. (Closes issue #11559)
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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)
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modified.
This requires casting the strings in asterisk.c when writing to
them, so we do it through a macro to do it consistently.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@93603 65c4cc65-6c06-0410-ace0-fbb531ad65f3
globally visible.
Document the fact that DEFAULT_TMP_DIR cannot be overridden
from the default configuration (this needs to be fixed, as you
could have a totally different spooldir configured at runtime,
and yet DEFAULT_TMP_DIR keeps the compile-time default).
Remove two unused entries for sounds and images.
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r93377 | file | 2007-12-17 18:28:09 -0400 (Mon, 17 Dec 2007) | 7 lines
Do not try to access information about a lock when printing out a trylock attempt. It is possible for the lock that it references to no longer be valid. This would have caused segfaults or deadlocks.
(issue #BE-263)
(closes issue #11080)
Reported by: callguy
(closes issue #11100)
Reported by: callguy
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you need to reload after changes. Thanks YS.
Reported by: ys
Patches:
trunk93163_manager_reload.c.diff uploaded by ys (license 281)
(related to issue #11414)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@93166 65c4cc65-6c06-0410-ace0-fbb531ad65f3
manager realtime implementation.
If you change accounts in manager.conf, you now need to reload to activate the
changes (deletions, additions). This was not the case with 1.4.
Reported by: ys
Patches:
trunk93163_manager_reload.c.diff uploaded by ys (license 281)
(closes issue #11414)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@93165 65c4cc65-6c06-0410-ace0-fbb531ad65f3
- Refer to the proper documentation
- Implement separate signalling/media QoS/CoS in many channels using RTP
- Improve warnings and verbose messages
- Deprecate some old settings
Minor modifications by me, a big effort from IgorG.
Thanks!
Reported by: IgorG
Patches:
qoscleanup-89394-4-trunk.patch uploaded by IgorG (license 20)
Tested by: IgorG
(closes issue #11145)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@93163 65c4cc65-6c06-0410-ace0-fbb531ad65f3
equivalent to the check done by ast_verb, I wrote a macro, VERBOSITY_LEVEL, which does this
check. I did a quick look in the source and used this macro in some places where option_verbose
was used.
I also converted some verbose messages in logger.c to use ast_verb instead of ast_verbose.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@93042 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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(closes issue #10690)
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r92696 | qwell | 2007-12-12 18:11:09 -0600 (Wed, 12 Dec 2007) | 7 lines
If a typo is found in a config file, we previous continued on with what was already loaded.
We do not want to do this (see bug below for details).
This makes it so that if a [ is found without a ], the entire config will fail, and nothing in it will be loaded.
Issue 10690.
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get_unaligned functions as const
* In event.c, use get_unaligned_uint32() in a couple of places to fix issues on
architectures that don't allow unaligned access
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r92204 | file | 2007-12-10 12:36:15 -0400 (Mon, 10 Dec 2007) | 6 lines
Add G729A as another possible payload name for G729. Some devices use this instead of G729, which is perfectly normal since the payload number itself is defined and can't be used by anything else so the name doesn't matter that much.
(closes issue #11483)
Reported by: revolution
Patches:
rtp.diff uploaded by revolution (license 346)
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r91830 | russell | 2007-12-07 15:24:33 -0600 (Fri, 07 Dec 2007) | 5 lines
Make the lock protecting each thread's list of locks it currently holds
recursive. I think that this will fix the situation where some people have
said that "core show locks" locks up the CLI.
(related to issue #11080)
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r91777 | russell | 2007-12-07 10:08:35 -0600 (Fri, 07 Dec 2007) | 6 lines
* Add a bit more of a verbose comment as to why a hangup frame needs to be
queued up if autoservice gets a NULL return from ast_read().
* Make the process of queueing the hangup frame more efficient by putting the
frame where it is going to end up and avoiding some locking and extra memory
allocations and freeing.
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r91737 | mmichelson | 2007-12-07 09:39:58 -0600 (Fri, 07 Dec 2007) | 7 lines
Hangups that happen during autoservice were not processed appropriately. This is
because a hangup actually causes a NULL frame to be received, not a hangup frame.
Queueing a hangup if we receive a NULL frame during autoservice corrects this problem
(closes issue #11467, reported by jmls, patched by me)
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r91450 | file | 2007-12-06 12:49:42 -0400 (Thu, 06 Dec 2007) | 6 lines
Fix various in the udptl implementation. It could return empty modem frames, have an incorrect sequence number on packets, and display the wrong sequence number in the debug messages.
(closes issue #11228)
Reported by: Cache
Patches:
udptl-4.patch uploaded by dimas (license 88)
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- Now use Response: success
- New header "Ping: pong" :-)
- The Events action
- Now use Response: Success
- The new status is reported as "Events: On" or "Events: Off"
- Report if manager is enabled in the reload event
Small cleanups...
From moremanager
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@91420 65c4cc65-6c06-0410-ace0-fbb531ad65f3
- 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)
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Maybe belongs in the new cdr category?
---moremanager---
Event: NewAccountCode
Modules: cdr.c
Purpose: To report a change in account code for a live channel
Example:
Event: NewAccountCode
Privilege: call,all
Channel: SIP/olle-01844600
Uniqueid: 1177530895.2
AccountCode: Stinas account 1234848484
OldAccountCode: Olles Account 12345
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Thanks, eliel, for writing the original patch. Modified by me to follow
other manager events and the new "moremanager" style.
(closes issue #11478)
Reported by: eliel
Patches:
manager.c.patch uploaded by eliel (license 64)
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r91192 | russell | 2007-12-05 11:31:42 -0600 (Wed, 05 Dec 2007) | 10 lines
Make the lock in the threadstorage debugging code untracked to avoid a deadlock
on thread destruction.
(closes issue #11207)
Reported by: ys
Patches:
threadstorage.c.diff uploaded by ys (license 281)
Also fixes an open bug report: (closes issue #11446)
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r91074 | russell | 2007-12-04 18:48:47 -0600 (Tue, 04 Dec 2007) | 4 lines
When DEBUG_THREADS is enabled, we only have the details about who is holding
a lock that we are waiting on for a mutex, not rwlocks. This should fix the
problem where people have reported "core show locks" crashing sometimes.
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r90967 | russell | 2007-12-04 13:57:39 -0600 (Tue, 04 Dec 2007) | 7 lines
Make some changes to some additions I made recently for doing channel autoservice
when looking up extensions. This code was added to handle the case where a
dialplan switch was in use that could block for a long time. However, the way
that I added it, it did this for all extension lookups. However, lookups in the
in-memory tree of extensions should _not_ take long enough to matter. So, move
the autoservice stuff to be only around executing a switch.
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This introduces two new options for zapata.conf: mwimonitor and mwimonitornotify.
The mwimonitor option enables MWI monitoring. When the MWI state on a line changes,
then the script specified by mwimonitornotify will be executed for custom handling
of the state change, similar to the externnotify option of voicemail.conf.
Also, when the MWI state on an FXO line changes, an internal Asterisk event is
generated to indicate the new state of the associated mailbox. That may, any
module that cares about MWI information will get notified and can handle it
just as if app_voicemail had sent this notification.
(BE-253, original patch from markster, with some minor modifications by me to
add comments, documentation, and internal event support)
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r90735 | mmichelson | 2007-12-03 17:12:17 -0600 (Mon, 03 Dec 2007) | 22 lines
A big one...
This is the merge of the forward-loop branch. The main change here is that call-forwards can no longer loop.
This is accomplished by creating a datastore on the calling channel which has a linked list of all devices
dialed. If a forward happens, then the local channel which is created inherits the datastore. If, through this
progression of forwards and datastore inheritance, a device is attempted to be dialed a second time, it will simply
be skipped and a warning message will be printed to the CLI. After the dialing has been completed, the datastore
is detached from the channel and destroyed.
This change also introduces some side effects to the code which I shall enumerate here:
1. Datastore inheritance has been backported from trunk into 1.4
2. A large chunk of code has been removed from app_dial. This chunk is the section of code
which handles the call forward case after the channel has been requested but before it has
been called. This was removed because call-forwarding still works fine without it, it makes the
code less error-prone should it need changing, and it made this set of changes much less painful
to just have the forwarding handled in one place in each module.
3. Two new files, global_datastores.h and .c have been added. These are necessary since the datastore
which is attached to the channel may be created and attached in either app_dial or app_queue, so they
need a common place to find the datastore info. This approach was taken in case similar datastores are
needed in the future, there will be a common place to add them.
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r90432 | tilghman | 2007-12-02 03:34:23 -0600 (Sun, 02 Dec 2007) | 7 lines
Clarify the return value on autoservice. Specifically, if you started
autoservice and autoservice was already on, it would erroneously return an
error.
Reported by: adiemus
Patch by: dimas
(Closes issue #11433)
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This works in much the same way as the automonitor, except that instead of using the monitor
app, it uses the mixmonitor app. By providing an 'x' or 'X' as a dial or queue option, a DTMF
sequence may be entered (as defined in features.conf) to start the one-touch mixmonitor.
This patch also introduces some new API calls to the audiohooks code for searching for an audiohook
by type and for searching for a running audiohook by type.
Big thanks to joetester for writing the initial patch, testing it and patiently waiting for it to
be committed.
(closes issue #10185, reported and patched by xmarksthespot)
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r90348 | russell | 2007-11-30 13:26:04 -0600 (Fri, 30 Nov 2007) | 8 lines
Change the behavior of ao2_link(). Previously, in inherited a reference.
Now, it automatically increases the reference count to reflect the reference
that is now held by the container.
This was done to be more consistent with ao2_unlink(), which automatically
releases the reference held by the container. It also makes it so it is
no longer possible for a pointer to be invalid after ao2_link() returns.
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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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Tilghman Lesher
Russell Bryant
Kevin P. Fleming
Luigi Rizzo
Mark Michelson
Jason Parker
Olle Johansson
Joshua Colp
Steve Murphy