Age | Commit message (Collapse) | Author |
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as RSW since i am too lazy to keep typing it all out). This time a few of
the channels.
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r118953 | tilghman | 2008-05-29 12:20:16 -0500 (Thu, 29 May 2008) | 3 lines
Add some debugging code that ensures that when we do deadlock avoidance, we
don't lose the information about how a lock was originally acquired.
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- make data member of the ast_frame struct a named union instead of a void
Recently the ast_queue_hangup function got a new parameter, the hangupcause
Feedback came in that this is no good and that instead a new function should be created.
This I did.
The hangupcause was stored in the seqno member of the ast_frame struct. This is not very
elegant, and since there's already a data member that one should be used.
Problem is, this member was a void *.
Now it's a named union so it can hold a pointer, an uint32 and there's a padding in case someone
wants to store another type in there in the future.
This commit is so massive, because all ast_frame.data uses have to be
altered to ast_frame.data.data
Thanks russellb and kpfleming for the feedback.
(closes issue #12674)
Reported by: mvanbaak
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r115944 | file | 2008-05-13 17:28:23 -0300 (Tue, 13 May 2008) | 4 lines
Use the right flag to open the audio in non-blocking.
(closes issue #12616)
Reported by: nicklewisdigiumuser
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(closes issue #11328)
Reported by: rain
Patches:
20071207__pass_cause_in_hangup_control_frame.diff.txt uploaded by Corydon76 (license 14)
brought up-to-date to trunk by me
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r106235 | file | 2008-03-05 18:32:10 -0400 (Wed, 05 Mar 2008) | 4 lines
Add a control frame to indicate the source of media has changed. Depending on the underlying technology it may need to change some things.
(closes issue #12148)
Reported by: jcomellas
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remove background thread and all sound generation mechanisms, as the built-in indications can handle everything that is needed
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were included almost everywhere.
Remove some of the instances.
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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.).
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didn't make much sense
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Closes issue #11039, as suggested by seanbright.
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r83974 | kpfleming | 2007-09-26 16:53:03 -0700 (Wed, 26 Sep 2007) | 2 lines
avoid the weird usage of assert() in the ALSA header files that gcc 4.2 wants to complain about
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Reported by: eliel
Patches:
res_features.c.patch uploaded by eliel (license 64)
res_agi.c.patch uploaded by seanbright (license 71)
res_musiconhold.c.patch uploaded by seanbright (license 71)
pbx.c.patch uploaded by moy (license 222)
logger.c.patch uploaded by moy (license 222)
frame.c.patch uploaded by moy (license 222)
manager.c.patch uploaded by moy (license 222)
http.c.patch uploaded by moy (license 222)
dnsmgr.c.patch uploaded by moy (license 222)
res_realtime.c.patch uploaded by eliel (license 64)
res_odbc.c.patch uploaded by seanbright (license 71)
res_jabber.c.patch uploaded by eliel (license 64)
chan_local.c.patch uploaded by eliel (license 64)
chan_agent.c.patch uploaded by eliel (license 64)
chan_alsa.c.patch uploaded by eliel (license 64)
chan_features.c.patch uploaded by eliel (license 64)
chan_sip.c.patch uploaded by eliel (license 64)
RollUp.1.patch (includes all of the above patches) uploaded by seanbright (license 71)
Convert many CLI commands to the NEW_CLI format.
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channel lock wrappers
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scalability and is done in such a way that we should be able to add support for other poll() replacements.
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(closes issue #10277, patches by mvanbaak)
Basically, this changes ...
if (option_verbose > 2)
ast_verbose(VERBOSE_PREFIX_3, "Something\n");
to ...
ast_verb(3, "Something\n");
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(issue #9957, patches from mvanbaak, caio1982, critch, and dimas)
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r64306 | russell | 2007-05-14 14:13:00 -0500 (Mon, 14 May 2007) | 3 lines
Properly handle AST_CONTROL_PROGRESS by just ignoring it. An unknown indication
will trigger an error and cause sounds to stop, which in this case, is ringing.
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r60989 | murf | 2007-04-09 12:32:07 -0600 (Mon, 09 Apr 2007) | 1 line
This is a big improvement over the current CDR fixes. It may still need refinement, but this won't have as many folks bothered.
This also adds the mods from 1.4/r.61136;
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r56888 | russell | 2007-02-26 14:42:21 -0600 (Mon, 26 Feb 2007) | 4 lines
Restore the behavior of Asterisk 1.2 where if a device was not specified in
alsa.conf, then we just use the system default, instead of creating our own
default of hw:0,0. (issue #9139)
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r51788 | file | 2007-01-23 17:46:31 -0500 (Tue, 23 Jan 2007) | 2 lines
Update channel drivers to use module referencing so that unloading them while in use will not result in crashes. (issue #8897 reported by junky)
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r51311 | russell | 2007-01-19 11:49:38 -0600 (Fri, 19 Jan 2007) | 23 lines
Merge the changes from the /team/group/vldtmf_fixup branch.
The main bug being addressed here is a problem introduced when two SIP
channels using SIP INFO dtmf have their media directly bridged. So, when a
DTMF END frame comes into Asterisk from an incoming INFO message, Asterisk
would try to emulate a digit of some length by first sending a DTMF BEGIN
frame and sending a DTMF END later timed off of incoming audio. However,
since there was no audio coming in, the DTMF_END was never generated. This
caused DTMF based features to no longer work.
To fix this, the core now knows when a channel doesn't care about DTMF BEGIN
frames (such as a SIP channel sending INFO dtmf). If this is the case, then
Asterisk will not emulate a digit of some length, and will instead just pass
through the single DTMF END event.
Channel drivers also now get passed the length of the digit to their digit_end
callback. This improves SIP INFO support even further by enabling us to put
the real digit duration in the INFO message instead of a hard coded 250ms.
Also, for an incoming INFO message, the duration is read from the frame and
passed into the core instead of just getting ignored.
(issue #8597, maybe others...)
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r46200 | kpfleming | 2006-10-25 09:32:08 -0500 (Wed, 25 Oct 2006) | 2 lines
apparently developers are still not aware that they should be use ast_copy_string instead of strncpy... fix up many more users, and fix some bugs in the process
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r44378 | kpfleming | 2006-10-04 14:47:22 -0500 (Wed, 04 Oct 2006) | 4 lines
update thread creation code a bit
reduce standard thread stack size slightly to allow the pthreads library to allocate the stack+data and not overflow a power-of-2 allocation in the kernel and waste memory/address space
add a new stack size for 'background' threads (those that don't handle PBX calls) when LOW_MEMORY is defined
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patch provided in bugnote, with minor changes.
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mods done by myself)
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Russell Bryant and the RTP portion done by myself, Muffinlicious Joshua Colp. This has gone through so many discussions/revisions it's not funny but we finally have it!
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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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channel to remove the theoretical race condition that the channel could get
bridged before the channel's jitterbuffer gets configured. This was pointed
out by PCadach on IRC. Thanks!
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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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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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so that channels not using a jitterbuffer don't waste as much memory
- ensure that the channel drivers that use jitterbuffers can handle a failure
from configuring a jitterbuffer on a new channel because of a memory
allocation error
- On passing through these channel drivers, configure the jitterbuffer before
starting the PBX thread instead of afterwards. If the pbx fails to start for
whatever reason, this would have caused a crash.
- Also on passing, move the increase of the usecount to after all of the
possible failure conditions in the function
- fix a place where ast_update_use_count() was not called
- ensure that the owner channel pointer of the channel pvt strcutures is set to
NULL in failure conditions
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again :-)
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These used to be wrapped in a #ifdef
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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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->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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autoconf and menuselect tools for Asterisk!
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description() and key() return values
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const-ify some more APIs
remove 'type' field from ast_channel, in favor of the one in the channel's tech structure
allow string field module users to specify the 'chunk size' for pool allocations
update chan_alsa to be compatible with recent const-ification patches
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(This is being added to the janitor projects list.)
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