Age | Commit message (Collapse) | Author | Files | Lines |
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- restructured build tree and makefiles to eliminate recursion problems
- support for embedded modules
- support for static builds
- simpler cross-compilation support
- simpler module/loader interface (no exported symbols)
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r40227 | russell | 2006-08-17 12:36:21 -0400 (Thu, 17 Aug 2006) | 3 lines
revert bogus change to attempt to fix bug 7506 which actually causes half of
the channels not to get "Newchannel" events at all (issue #7745)
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- instead of defining a free() wrapper in a bunch of files, define it as
ast_free() in utils.h and remove the copies from all the files.
- centralize and abstract the code used for doing thread storage. The code
lives in threadstorage.h, with one function being implemented in utils.c.
This new API includes generic thread storage as well as special functions
for handling thread local dynamic length string buffers.
- update ast_inet_ntoa() to use the new threadstorage API
- update ast_state2str() to use the new threadstorage API
- update ast_cli() to use the new threadstorage API
- Modify manager_event() to use thread storage. Instead of using a buffer of
4096 characters as the workspace for building the manager event, use a thread
local dynamic string. Now there is no length limitation on the length of the
body of a manager event.
- Significantly simplify the handling of ast_verbose() ...
- Instead of using a static char buffer and a lock to make sure only one
thread can be using ast_verbose() at a time, use a thread local dynamic
string as the workspace for preparing the verbose message. Instead of
locking around the entire function, the only locking done now is when the
message has been built and is being deliviered to the list of registered
verbose message handlers.
- This function was doing a strdup() on every message passed to it and
keeping a queue of the last 200 messages in memory. This has been
completely removed. The only place this was used was that if there were
any messages in the verbose queue when a verbose handler was registered,
all of the messages in the queue would be fed to it. So, I just made sure
that the console verbose handler and the network verbose handler (for
remote asterisk consoles) were registered before any verbose messages.
pbx_gtkconsole and pbx_kdeconsole will now lose a few verbose messages at
startup, but I didn't feel the performance hit of this message queue was
worth saving the initial verbose output for these very rarely used modules.
- I have removed the last three arguments to the verbose handlers, leaving
only the string itself because they aren't needed anymore. For example,
ast_verbose had some logic for telling the verbose handler to add
a newline if the buffer was completely full. Now that the buffer can grow
as needed, this doesn't matter anymore.
- remove unused function, ast_verbose_dmesg() which was to dispatch the
message queue
- Convert the list of verbose handlers to use the linked list macros.
- add missing newline characters to a few ast_verbose() calls
- convert the list of log channels to use the linked list macros in logger.c
- fix close_logger() to close all of the files it opened for logging
- update ast_log() to use a thread local dynamic string for its workspace
for preparing log messages instead of a buffer of size BUFSIZ (8kB on my
system) allocated on the stack. The dynamic string in this case is limited
to only growing to a maximum size of BUFSIZ.
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r39056 | file | 2006-08-06 20:15:51 -0400 (Sun, 06 Aug 2006) | 2 lines
Reset our stream and vstream pointers back to NULL so that any generator that uses them (file based MOH) will not try to close them again. (issue #7668 reported by jmls)
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- don't include a newline character when building a response for an unknown
state
(issue #7645, jonty)
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r38982 | russell | 2006-08-05 05:01:37 -0400 (Sat, 05 Aug 2006) | 6 lines
Always generate a Newstate event in ast_setstate() instead of making it a
Newchannel event if the state was AST_STATE_DOWN. The Newchannel event will
always be generated in ast_request(), so this just causes a duplicated
Newchannel event in some cases.
(issue #7506, repoted by capouch, fixed by me)
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r38903 | russell | 2006-08-05 01:07:39 -0400 (Sat, 05 Aug 2006) | 2 lines
suppress a compiler warning about the usage of a potentially uninitialized variable
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r38904 | russell | 2006-08-05 01:08:50 -0400 (Sat, 05 Aug 2006) | 10 lines
Fix an issue that would cause a NewCallerID manager event to be generated
before the channel's NewChannel event. This was due to a somewhat recent
change that included using ast_set_callerid() where it wasn't before. This
function should not be used in the channel driver "new" functions.
(issue #7654, fixed by me)
Also, fix a couple minor bugs in usecount handling. chan_iax2 could have
increased the usecount but then returned an error. The place where chan_sip
increased the usecount did not call ast_update_usecount()
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r38686 | kpfleming | 2006-08-01 18:07:06 -0500 (Tue, 01 Aug 2006) | 2 lines
ensure that the 'feature digit timeout' value is taken into account when deciding how long the bridge should run (this fixes a problem report where a digit press that did not invoke a feature is never passed across the bridge)
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implement whisper mode for ExtenSpy/ChanSpy
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use this way
add infrastructure for whispering onto a channel
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fix prototype for a channel walking function to use a const input pointer
use existing channel walk by name prefix instead of reproducing that code in this app
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r38347 | kpfleming | 2006-07-27 10:40:03 -0500 (Thu, 27 Jul 2006) | 2 lines
do a better job avoiding translation path teardown/setup when not needed
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r38310 | kpfleming | 2006-07-26 21:43:49 -0500 (Wed, 26 Jul 2006) | 2 lines
don't do useless translation destroy/build when the channel is already in the correct format
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storage changes (fixes issue #7595)
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immediately after it gets allocated, which will automatically NULL terminate
the string
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in ast_write(), which is to call the channel's indicate function if it exists
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static buffer for storing the result when the state value is unknown
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handling the way it was decided at AstriDevCon Europe 2006 (and the way people really want it to be)
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split support for G726-32 into RFC3551 and AAL2 packing orders, since both are in use
change "G726-32" to be RFC3551 packing order, in spite of devices that use AAL2 order with this MIME type
add ability to directly transcode between packing orders
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r37361 | kpfleming | 2006-07-10 16:01:35 -0500 (Mon, 10 Jul 2006) | 2 lines
do masquerade-behind-proxy checking with better control over locks
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to native bridge, feature redirect was not being checked.
patch from bug #7296
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minute train ride from Paris to London <G>)
support the new location for zaptel.h and tonezone.h
use the dependency information output by menuselect to build Makefile rules for each module for header files and libraries
combine the common rules into a top-level Makefile.rules file
remove all (now) unnecessary stuff from subdir Makefiles
change translator API so that the newpvt() callback returns an int instead of a pointer (it no longer allocates memory)
alphabetize --with-<foo> options in configure script
enhance Net-SNMP support in configure script to provide a --with-netsnmp option
fix support for --with-pq so that if pg-config is not found when --with-pq is specified, an error will be generated
add 'optional package' usage to modules now that menuselect can output it
allow res_snmp to build by default, since the new loader changes coming soon will solve the function naming problem (and users can disable it via menuselect anyway)
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allocated. These changes caused crashes when using a channel type that did
not support the jitterbuffer. Instead of fixing why it's crashing, I'm going
to implement this in a better way next week. The way I did it caused a
jitterbuffer to be allocated on every channel where the channel type supported
jitterbuffers, even if they were disabled.
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bug 5090 by josh colp. Thanks to everyone who help get this patch through
especially to the author Steven Underwood.
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r33724 | file | 2006-06-12 18:34:38 -0300 (Mon, 12 Jun 2006) | 2 lines
Greatly simply the mixmonitor thread, and move channel reference directly to spy structure so that the core can modify it.
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again :-)
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and some patches (all disclaimed).
- Don't change RTP properties if we reject a re-INVITE
- Don't add video to an outbound channel if there's no video on the inbound channel
- Don't include video in the "preferred codec" list for codec selection
- Clean up and document code that parses and adds SDP attachments
Since we do not transcode video, we can't handle video the same way as audio. This is a
bug fix patch. In future releases, we need to work on a solution for video negotiation,
not codecs but formats and framerates instead.
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(deadlock reported by and debuged with hads on #asterisk-dev on IRC)
(probably fixes issue #7267, maybe others)
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- Change formatting of RTP debug to align better
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with messages about Internal timing when a channel has to make a decision about generating internal timing (eg - MOH)
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r31520 | kpfleming | 2006-06-01 15:27:50 -0500 (Thu, 01 Jun 2006) | 2 lines
handle Zap transfers behind chan_agent properly so the agent doesn't get stuck waiting for the call to hang up
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a new implementation of a fixed size jitterbuffer, as well as support for the
existing adaptive jitterbuffer implementation. (issue #3854, Slav Klenov)
Thank you very much to Slav Klenov of Securax and all of the people involved
in the testing of this feature for all of your hard work!
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in the log
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using the Hangup application. (issue #7160 reported by kmilitzer branch by jcollie)
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https://origsvn.digium.com/svn/asterisk/branches/1.2
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r27468 | kpfleming | 2006-05-16 15:05:17 -0500 (Tue, 16 May 2006) | 2 lines
don't leak frames when deferring DTMF or dropping duplicate ANSWER frames (issue #7041, slightly different fix, reported/patched by clausf)
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paths that set it to AST_BRIDGE_COMPLETE inside the loop (thanks Oskar!)
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As it turns out, all of these checks were useless, because alloca will never
return NULL.
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->indicate()
update iax2_indicate to pass control frame payload to the connected channel
add an API call for sending an indication with payload, and use it for control frames with payload
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asterisk to answer an outbound call
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support sending control frames with payload
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instead of LOG_WARNING as this only has relevance to developers trying to resolve a possible problem. #6949
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