No real Jingle implementation being available, testing was made using
two Asterisk servers relaying SIP calls over their Jingle channels:
SIP Phone 1 --- [chan_sip]Asterisk 1[chan_jingle] --- [chan_jingle]Asterisk 2[chan_sip] --- SIP Phone 2
Thus, it was possible to test the code in both ways, and make the
Jingle channel comply with the latest specifications. No sound available yet.
Main modifications include :
- modified the 'jingle_candidate' structure and the
'jingle_create_candidates' function according to XEP-0176 ;
- modified the 'jingle_action' function in order to properly terminate
a Jingle session, in conformance with XEP-0166 ;
- modified username format used in STUN requests ;
- actually make the bindaddr configuration field useable.
Todo :
- set audio paths up (no native bridging) ;
- make the CLI gtalk functions available to jingle ;
- clean up the storage space used in strings.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@83743 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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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r82929 | russell | 2007-09-18 17:42:27 -0500 (Tue, 18 Sep 2007) | 11 lines
Add a new patch to handle interrupting the fgets() call when using FastAGI.
This version of the patch maintains the original behavior of the code when
not using FastAGI.
(closes issue #10553)
Reported by: juggie
Patches:
res_agi_fgets-4.patch uploaded by juggie (license 24)
res_agi_fgets_1.4svn.patch uploaded by juggie (license 24)
Slight mods by me
Tested by: juggie, festr
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r82337 | russell | 2007-09-13 13:45:59 -0500 (Thu, 13 Sep 2007) | 4 lines
Only compile in tracking astobj2 statistics if dev-mode is enabled. Also, when
dev mode is enabled, register the CLI command that can be used to run the astobj2
test and print out statistics.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@82338 65c4cc65-6c06-0410-ace0-fbb531ad65f3
it doesn't free the slinfactory itself. (This isn't related to a bug, i'm just
looking over random code)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@81999 65c4cc65-6c06-0410-ace0-fbb531ad65f3
ao2_hash_fn and ao2_callback_fn typedefs, in preparation
to more cleanup of the _search_flags
Please do not merge this change to 1.4 yet - there are no
functional changes anyways.
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r81599 | russell | 2007-09-05 15:53:41 -0500 (Wed, 05 Sep 2007) | 11 lines
Fix an issue that can occur when you do an attended transfer to parking. If
you complete the transfer before the announcement of the parking spot finishes,
then the channel being parked will hear the remainder of the announcement.
These changes make it so that will not happen anymore.
Basically, res_features sets a flag on the channel is playing the announcement
to so that the file streaming core knows that it needs to watch out for a
channel masquerade, and if it occurs, to abort the announcement.
(closes BE-182)
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r81448 | russell | 2007-09-04 13:37:44 -0500 (Tue, 04 Sep 2007) | 4 lines
Remove the typedefs on ao2_container and ao2_iterator. This is simply because
we don't typedef objects anywhere else in Asterisk, so we might as well make
this follow the same convention.
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functions. As inline functions, the lock debug information will show that
these are always locked in audiohooks.h instead of the file where the lock was
actually acquired.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@81264 65c4cc65-6c06-0410-ace0-fbb531ad65f3
* 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
Reported by: nic_bellamy
Patches:
2006-10-03_svn_44249_voicemail_lockmode_v3.patch uploaded by nic_bellamy (license 213)
Add support for configurable file locking methods. The default is "lockfile",
which is the old behavior. There is an additional option, "flock", which is
intended for use in situations where the lockfile method will not work, such as
with SMB/CIFS mounts.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@81233 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r80573 | russell | 2007-08-23 15:16:41 -0500 (Thu, 23 Aug 2007) | 5 lines
When executing a dynamic feature, don't look it up a second time by digit pattern
after we already looked it up by name. This causes broken behavior if there is
more than one feature defined with the same digit pattern.
(closes issue #10539, reported by bungalow, patch by me)
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r80426 | russell | 2007-08-22 17:54:03 -0500 (Wed, 22 Aug 2007) | 6 lines
Add some more documentation on iterating ao2 containers. The documentation
implies that is possible to miss an object or see an object twice while
iterating. After looking through the code and talking with mmichelson, I have
documented the exact conditions under which this can happen (which are rare and
harmless in most cases).
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r80362 | russell | 2007-08-22 15:21:36 -0500 (Wed, 22 Aug 2007) | 34 lines
Merge changes from team/russell/iax_refcount.
This set of changes fixes problems with the handling of iax2_user and iax2_peer
objects. It was very possible for a thread to still hold a reference to one of
these objects while a reload operation tries to delete them. The fix here is to
ensure that all references to these objects are tracked so that they can't go away
while still in use.
To accomplish this, I used the astobj2 reference counted object model. This
code has been in one of Luigi Rizzo's branches for a long time and was primarily
developed by one of his students, Marta Carbone. I wanted to go ahead and bring
this in to 1.4 because there are other problems similar to the ones fixed by these
changes, so we might as well go ahead and use the new astobj if we're going to go
through all of the work necessary to fix the problems.
As a nice side benefit of these changes, peer and user handling got more efficient.
Using astobj2 lets us not hold the container lock for peers or users nearly as long
while iterating. Also, by changing a define at the top of chan_iax2.c, the objects
will be distributed in a hash table, drastically increasing lookup speed in these
containers, which will have a very big impact on systems that have a large number of
users or peers.
The use of the hash table will be made the default in trunk. It is not the default
in 1.4 because it changes the behavior slightly. Previously, since peers and users
were stored in memory in the same order they were specified in the configuration file,
you could influence peer and user matching order based on the order they are specified
in the configuration. The hash table does not guarantee any order in the container,
so this behavior will be going away. It just means that you have to be a little
more careful ensuring that peers and users are matched explicitly and not forcing
chan_iax2 to have to guess which user is the right one based on secret, host, and
access list settings, instead of simply using the username.
If you have any questions, feel free to ask on the asterisk-dev list.
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https://origsvn.digium.com/svn/asterisk/branches/1.4
(closes issue #10430)
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r79904 | qwell | 2007-08-17 14:12:19 -0500 (Fri, 17 Aug 2007) | 11 lines
Don't send a semicolon over the wire in sip notify messages.
Caused by fix for issue 9938.
I basically took the code that existed before 9938 was fixed, and
copied it into a new function - ast_unescape_semicolon
There should be very few places this will be needed (pbx_config
does NOT need this (see issue 9938 for details))
Issue 10430, patch by me, with help/ideas from murf (thanks murf).
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in place of a very common construct. I also used it in a number of places
in chan_sip.
if (id > -1)
ast_sched_del(sched, id);
id = ast_sched_add(sched, ...);
changes to:
ast_sched_replace(id, sched, ...);
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@79861 65c4cc65-6c06-0410-ace0-fbb531ad65f3
The way a device state change propagates is kind of silly, in my opinion. A
device state provider calls a function that indicates that the state of a
device has changed. Then, another thread goes back and calls a callback for
the device state provider to find out what the new state is before it can go
send it off to whoever cares.
I have changed it so that you can include the state that the device has changed
to in the first function call from the device state provider. This removes the
need to have to call the callback, which locks up critical containers to go find
out what the state changed to.
This change set changes the "simple" device state providers to use the new method.
This includes parking, meetme, and SLA.
I have also mostly converted chan_agent in my branch, but still have some more
things to think through before presenting the plan for converting channel drivers
to ensure all of the right events get generated ...
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r78995 | russell | 2007-08-10 10:20:09 -0500 (Fri, 10 Aug 2007) | 4 lines
The last set of changes that I made to "core show locks" made it not able to
track mutexes unless they were declared using AST_MUTEX_DEFINE_STATIC. Locks
initialized with ast_mutex_init() were not tracked. It should work now.
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r78103 | mmichelson | 2007-08-03 15:25:22 -0500 (Fri, 03 Aug 2007) | 7 lines
Changed the behavior of sip's realtime_peer function to match the corresponding way of matching for non-realtime peers.
Now matches are made on both the IP address and port number, or if the insecure setting is set to "port" then just match on the
IP address.
In order to accomplish this, I also added a new API call, ast_category_root, which returns the first variable of an ast_category struct
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r78184 | russell | 2007-08-06 11:50:54 -0500 (Mon, 06 Aug 2007) | 5 lines
Fix the return value of AST_LIST_REMOVE(). This shouldn't be causing any
problems, though, because the only code that uses the return value only checks
to see if it is NULL.
(closes issue #10390, pointed out by mihai)
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The problem was that res_crypto now has a RWLIST named "keys". The macro
for defining this list defines a function used as a constructor for the list
called "init_keys". However, there was another function called init_keys in
this module for a CLI command. The fix is just to prepend the generated
functions with underscores.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@78138 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r78095 | russell | 2007-08-03 14:39:49 -0500 (Fri, 03 Aug 2007) | 28 lines
Add some improvements to lock debugging. These changes take effect
with DEBUG_THREADS enabled and provide the following:
* This will keep track of which locks are held by which thread as well as
which lock a thread is waiting for in a thread-local data structure. A
reference to this structure is available on the stack in the dummy_start()
function, which is the common entry point for all threads. This information
can be easily retrieved using gdb if you switch to the dummy_start() stack
frame of any thread and print the contents of the lock_info variable.
* All of the thread-local structures for keeping track of this lock information
are also stored in a list so that the information can be dumped to the CLI
using the "core show locks" CLI command. This introduces a little bit of a
performance hit as it requires additional underlying locking operations
inside of every lock/unlock on an ast_mutex. However, the benefits of
having this information available at the CLI is huge, especially considering
this is only done in DEBUG_THREADS mode. It means that in most cases where
we debug deadlocks, we no longer have to request access to the machine to
analyze the contents of ast_mutex_t structures. We can now just ask them
to get the output of "core show locks", which gives us all of the information
we needed in most cases.
I also had to make some additional changes to astmm.c to make this work when
both MALLOC_DEBUG and DEBUG_THREADS are enabled. I disabled tracking of one
of the locks in astmm.c because it gets used inside the replacement memory
allocation routines, and the lock tracking code allocates memory. This caused
infinite recursion.
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r77788 | russell | 2007-07-30 14:13:31 -0500 (Mon, 30 Jul 2007) | 10 lines
(closes issue #10279)
Reported by: seanbright
Patches:
res_agi.carefulwrite.1.4.07252007.patch uploaded by seanbright (license 71)
res_agi.carefulwrite.trunk.07252007.patch uploaded by seanbright (license 71)
Allow the "agi_network: yes" line to be printed out in the AGI debug output.
Also, allow partial writes to be handled when writing out this line just like
it is for all of the others.
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- Makes the structures handling external AGI commands a bit more thread-safe
- Makes AGI transparently work with both live and hungup channels
- DeadAGI is hence no longer necessary and is deprecated
- CLI bug fixes
- Commands will refuse to run if the channel is dead and the command is nonsensical
for dead channels.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@76707 65c4cc65-6c06-0410-ace0-fbb531ad65f3
using old methods of parsing arguments to using the standard macros. However, the big
change is that the really old way of specifying application and arguments separated by
a comma will no longer work (e.g. NoOp,foo|bar). Instead, the way that has been
recommended since long before 1.0 will become the only method available (e.g. NoOp(foo,bar).
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@76703 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Reported by: snuffy
Patches:
doxygen-updates.diff uploaded by snuffy (license 35)
Another big batch of doxygen documentation updates
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@76559 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Reported by: bbryant
Patches:
20070720__core_debug_by_file.patch uploaded by bbryant (license 36)
(with some modifications by me)
Tested by: russell, bbryant
This set of changes introduces the ability to set the core debug or verbose
levels on a per-file basis. Interestingly enough, in 1.4, you have the ability
to set core debug for a single file, but that functionality was accidentally
lost in the conversion of the CLI commands to the new format.
This patch improves upon what was in 1.4 by letting you set it for more than 1
file, and by also supporting verbose.
*** Janitor Project ***
This patch also introduces a new macro, ast_verb(), which is similar
to ast_debug(). Setting the per file verbose value only works for messages that
use this macro. Converting existing uses of ast_verbose() can be done like:
if (option_debug > 2)
ast_verbose(VERBOSE_PREFIX_3 "Something useful\n");
...
ast_verb(3, "Something useful\n");
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@76555 65c4cc65-6c06-0410-ace0-fbb531ad65f3
(e.g. chan_sip.c in a subsequent commit).
Obviously exposing the internals of a data structure is far from ideal
(especially in a case like this where the implementation is very
inefficient and will need to be changed at some point).
On the other hand, it was also unclear what additional APIs should
we provide instead, and because exposing the stucture has no impact
on source and binary compatibility, this seemed to me the best option at
this time.
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in a consistent way. This is meant to replace the custom code
which is repeated all over the place in the various files when
parsing config files, CLI entries and other string information.
Right now the code supports parsing int32, uint32 and sockaddr_in with
optional default values and bound checks. It contains minimal error
checking, but that can be easily extended as the need arises.
Being a new API i am introducing this only in trunk, though I believe
that once the interface has been ironed out it might become a
worthwhile addition to 1.4 as well - basically, the first time
we will need to fix a piece of argument parsing code, we might as
well bring in this change and use the new API instead.
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use the \retval tag for documenting return values, fixing various warnings
when generating the documentation, and various other things.
(closes issue #10203, snuffy)
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sockets other than RTP ones.
The main change is a new API function in main/rtp.c (see there
for a description)
int ast_stun_request(int s, struct sockaddr_in *dst,
const char *username, struct sockaddr_in *answer)
which can be used to send an STUN request on a socket, and
optionally wait for a reply and store the STUN_MAPPED_ADDRESS
into the 'answer' argument (obviously, the version that
waits for a reply is blocking, but this is no different
from DNS resolutions).
Internally there are minor modifications to let stun_handle_packet()
be somewhat configurable on how to parse the body of responses.
At the moment i am not committing any change to the clients,
but adding STUN client support is extremely simple, e.g. chan_sip.c
could do something like this:
+ add a variable to store the stun server address;
static struct sockaddr_in stunaddr = { 0, }; /*!< stun server address */
+ add code to parse a config file of the form "stunaddr=my.stun.server.org:3478"
(not shown for brevity);
+ right after binding the main sip socket, talk to the stun server to
determine the externally visible address
if (stunaddr.sin_addr.s_addr != 0)
ast_stun_request(sipsock, &stunaddr, NULL, &externip);
so now 'externip' is set with the externally visible address.
so it is really trivial.
Similarly ast_stun_request could be called when creating the RTP
socket (possibly adding a struct sockaddr_in field in the struct
ast_rtp to store the externalip).
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@75034 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
(closes issue #10133)
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r74374 | qwell | 2007-07-10 13:39:30 -0500 (Tue, 10 Jul 2007) | 13 lines
Merged revisions 74373 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r74373 | qwell | 2007-07-10 13:37:23 -0500 (Tue, 10 Jul 2007) | 5 lines
Use res_ndestroy on systems that have it. Otherwise, use res_nclose.
This prevents a memleak on NetBSD - and possibly others.
Issue 10133, patch by me, reported and tested by scw
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Reported by: julien23
Patches submitted by: julien23
Add the ability to disable recording the input or output streams in res_monitor.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@74164 65c4cc65-6c06-0410-ace0-fbb531ad65f3
If the call is answered by another phone, other phones won't display the call as "missed".
You can also add an option to the dial command so that you can have a "followme"
scenario and not count the calls as "missed" when you cancel the call.
Thanks to Ramon and Frank for feedback on this feature.
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*must* write the file to the FILE *, and not the raw fd. Otherwise, it breaks
TLS support.
Thanks to rizzo for catching this!
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Handle transferring large files from the built-in http server. Previously, the
code attempted to malloc a block as large as the file itself. Now it uses the
sendfile() system call so that the file isn't copied into userspace at all if
it is available. Otherwise, it just uses a read/write of small chunks at a time.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@72701 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Also adds a new "reporting" permission for manager, since it can be incredibly spammy.
This permission was discussed on the -dev mailing list some months back.
Issue 8613, patch by johann8384, with some minor changes by me.
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r69702 | russell | 2007-06-18 11:35:02 -0500 (Mon, 18 Jun 2007) | 6 lines
To prevent 92138749238754 more reports of "I have unixodbc installed, but
still can't build *_odbc.so!", check for ltdl directly, instead of just listing
it as another library to include in the unixodbc check in the configure script.
This also makes ltdl show up as a dependency in menuselect so people know what
to go install. (related to issue #9989, patch by me)
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the beginning of the file. Also, add a channel variable that indicates
the location in the file where the Playback was stopped.
(closes issue #7655, patch from sharkey)
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r67993 | oej | 2007-06-07 11:00:44 +0200 (Thu, 07 Jun 2007) | 6 lines
Issue #9738 - Make sure we can unload res_jabber. Patch by phsultan - thanks!
Due to a bug in the iksemel library, this will not work if you are using GTLS
in the connection. That's being investigated. If you figure out a way to handle
that without us having to patch iksemel, let us know in the bug report. Thanks.
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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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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r67492 | russell | 2007-06-05 15:53:28 -0500 (Tue, 05 Jun 2007) | 16 lines
This bug has been hanging over my head ever since I wrote this SLA code.
Every time I tried to go debug it by adding some debug output, the behavior
would change. It turns out I wasn't crazy. I had the following piece of code:
if (remove)
AST_LIST_REMOVE_CURRENT(...);
Well, AST_LIST_REMOVE_CURRENT was not wrapped in braces, so my conditional
statement didn't do much good at all. It always ran at least all of the
macro minus the first statement, so I was seeing list entries magically
disappear when they weren't supposed to.
After many hours of debugging, I have come to this extremely irritating fix. :)
(issues #9581, #9497)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@67493 65c4cc65-6c06-0410-ace0-fbb531ad65f3
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r67308 | russell | 2007-06-05 10:51:53 -0500 (Tue, 05 Jun 2007) | 5 lines
When shutting down "gracefully", go through and run the unload() callbacks for
all of the modules. "stop now" is considered a non-graceful shutdown and will
not go through this process.
(issue #9804, reported by chrisost, patch by me)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@67310 65c4cc65-6c06-0410-ace0-fbb531ad65f3
places that cared about device states were app_queue and the hint code in pbx.c.
The changes include converting it to use the Asterisk event system, as well as
other efficiency improvements.
* app_queue: This module used to register a callback into devicestate.c to
monitor device state changes. Now, it is just a subscriber to Asterisk
events with the type, device state.
* pbx.c hints: Previously, the device state processing thread in devicestate.c
would call ast_hint_state_changed() each time the state of a device changed.
Then, that code would go looking for all the hints that monitor that device,
and call their callbacks. All of this blocked the device state processing
thread. Now, the hint code is a subscriber of Asterisk events with the
type, device state. Furthermore, when this code receives a device state
change event, it queues it up to be processed by another thread so that it
doesn't block one of the event processing threads.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@66958 65c4cc65-6c06-0410-ace0-fbb531ad65f3
places in the code where the same block of code for creating detached threads
was replicated. (patch from bbryant)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@65968 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r65200 | murf | 2007-05-18 16:06:27 -0600 (Fri, 18 May 2007) | 9 lines
Merged revisions 65172 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r65172 | murf | 2007-05-18 14:56:20 -0600 (Fri, 18 May 2007) | 1 line
This update will fix the situation that occurs as described by 9717, where when several targets are specified for a dial, if any one them reports FAIL, the whole call gets FAIL, even though others were ringing OK. I rearranged the priorities, so that a new disposition, NULL, is at the lowest level, and the disposition get init'd to NULL. Then, next up is FAIL, and next up is BUSY, then NOANSWER, then ANSWERED. All the related set routines will only do so if the disposition value to be set to is greater than what's already there. This gives the intended effect. So, if all the targets are busy, you'd get BUSY for the call disposition. If all get BUSY, but one, and that one rings is not answered, you get NOANSWER. If by some freak of nature, the NULL value doesn't get overridden, then the disp2str routine will report NOANSWER as before.
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allow you to initiate an ENUM query using ENUMQUERY, and then access the
details of all of the results using ENUMRESULT. Previously, if you wanted
to access multiple results, Asterisk would have to do a new DNS lookup every
time. (patch by bbryant)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@64480 65c4cc65-6c06-0410-ace0-fbb531ad65f3
- Add missing option to options.h
- Add missing variables to asterisk.h
- Move manager version to manager.h include file
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@63030 65c4cc65-6c06-0410-ace0-fbb531ad65f3
This set of changes introduces a new generic event API for use within Asterisk.
I am still working on a way for events to be shared between servers, but this
part is ready and can already be used inside of Asterisk.
This set of changes introduces the first use of the API, as well. I have
restructured the way that MWI (message waiting indication) is handled. It is
now event based instead of polling based. For example, if there are a bunch
of SIP phones subscribed to mailboxes, then chan_sip will not have to
constantly poll the mailboxes for changes. app_voicemail will generate events
when changes occur.
See UPGRADE.txt and CHANGES for some more information on the effects of these
changes from the user perspective. For developer information, see the text in
include/asterisk/event.h.
As always, additional feedback is welcome on the asterisk-dev mailing list.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@62292 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@61782 65c4cc65-6c06-0410-ace0-fbb531ad65f3
Philippe Sultan
Tilghman Lesher
Russell Bryant
Jason Parker
Luigi Rizzo
Steve Murphy
Mark Michelson
Joshua Colp
Doug Bailey
Olle Johansson
Kevin P. Fleming
Dwayne M. Hubbard