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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r80132 | russell | 2007-08-21 10:22:22 -0500 (Tue, 21 Aug 2007) | 3 lines
Don't try to dereference the owner channel when it may not exist
(issue #10507, maxper)
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r79174 | file | 2007-08-13 11:18:04 -0300 (Mon, 13 Aug 2007) | 4 lines
(closes issue #10437)
Reported by: haklin
Don't set the callerid name and number a second time on a newly created channel. ast_channel_alloc itself already sets it and setting it twice would cause a memory leak.
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the mailbox context. Now, all related MWI event dealings pay attention
to the context as well.
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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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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r77869 | file | 2007-08-01 14:56:59 -0300 (Wed, 01 Aug 2007) | 2 lines
Add some fixes for building on Solaris.
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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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microsecond, instead of only to the second
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r69392 | kpfleming | 2007-06-14 16:50:40 -0500 (Thu, 14 Jun 2007) | 2 lines
use ast_localtime() in every place localtime_r() was being used
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(issue #9957, patches from mvanbaak, caio1982, critch, and dimas)
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guidelines changes
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r67121 | qwell | 2007-06-04 17:36:57 -0500 (Mon, 04 Jun 2007) | 4 lines
Fixes for dtmf/dialing with mgcp (similar to the recent fix for chan_skinny)
Issue 9855, patch by DEA.
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places in the code where the same block of code for creating detached threads
was replicated. (patch from bbryant)
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file doc/qos.tex has been updated to document the new functionality.
(issue #9540, patch submitted by IgorG)
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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.
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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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T.140/RFC 2793 is a live communication channel, originally
created for IP based text phones for hearing impaired.
Feels very much like the old Unix talk application.
This code is developed and disclaimed by John Martin of Aupix, UK.
Tested for interoperability by myself and Omnitor in Sweden,
the company that wrote most of the specifications.
A big thank you to everyone involved in this.
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r53046 | russell | 2007-01-31 15:32:08 -0600 (Wed, 31 Jan 2007) | 11 lines
Merged revisions 53045 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r53045 | russell | 2007-01-31 15:25:11 -0600 (Wed, 31 Jan 2007) | 3 lines
Fix a bunch of places where pthread_attr_init() was called, but
pthread_attr_destroy() was not.
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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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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- Use this in chan_sip
- Document ha functions in acl.c
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r47436 | tilghman | 2006-11-10 10:51:55 -0600 (Fri, 10 Nov 2006) | 2 lines
Discussion of these CLI changes resulted in more consistency (Bug 8236)
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r47051 | tilghman | 2006-11-02 17:00:20 -0600 (Thu, 02 Nov 2006) | 2 lines
Reverse change of "show" to "list" and make several other commands more consistent with "category verb arguments"
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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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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https://origsvn.digium.com/svn/asterisk/branches/1.4
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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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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r43933 | file | 2006-09-28 14:05:43 -0400 (Thu, 28 Sep 2006) | 2 lines
Put in missing \ns on the end of ast_logs (issue #7936 reported by wojtekka)
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r43454 | file | 2006-09-21 18:12:09 -0400 (Thu, 21 Sep 2006) | 2 lines
Clean up chan_mgcp's module load function (issue #8001 reported by Mithraen with mods by moi)
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now reports AST_MODULE_LOAD_DECLINE when loading if config file
is not there, also fixed an error in res_config_pgsql where it
had a non static function when it should.
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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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that's what RTP-level packet bridging is all about!
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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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r40057 | kpfleming | 2006-08-16 13:57:44 -0500 (Wed, 16 Aug 2006) | 2 lines
don't allow AUEP responses to overflow the stack during a string copy (reported by Mu Security)
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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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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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inet_ntoa, which uses thread specific data (aka thread local storage) instead
of stack allocatted buffers to store the result.
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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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been requested (Sipura/Grandstream ATAs and others will need this)
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r36725 | russell | 2006-07-03 00:19:09 -0400 (Mon, 03 Jul 2006) | 4 lines
use ast_set_callerid to be more consistent and to make sure that the
"callerid" option in the conf files is always handled the same way and sets ANI
(issue #7285, gkloepfer)
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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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and mgcp channels
- remove the jitterbuffer configuration from the pvt structures in
the sip, zap, and skinny channel drivers, as copying the same global
configuration into each pvt structure has no benefit.
- update and fix some typos in jitterbuffer related documentation
(issue #7257, north, with additional updates and modifications)
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tech structures indicate that they create jitter
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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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