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Combined effort by DEA and mvanbaak.
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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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r58023 | russell | 2007-03-06 12:01:20 -0600 (Tue, 06 Mar 2007) | 3 lines
Return an error of transmit_response is called without a session.
(issue #9002)
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See skinny.conf.sample for configuration example.
Note: Some devices (seen on 12SP+/30VIP) will lock up if they monitor too many hints.
This seems to be a hardware limitation - there isn't anything we can do about it.
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Now you should be able to subscribe to a Skinny device/line.
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r56569 | qwell | 2007-02-23 20:02:53 -0600 (Fri, 23 Feb 2007) | 4 lines
Make sure to set a speeddials parent on creation.
Don't crash if hold is pressed when no call is active.
Don't return in places that we shouldn't..
Update softkey map when call is connected
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r55217 | qwell | 2007-02-17 11:10:09 -0600 (Sat, 17 Feb 2007) | 4 lines
Fix an issue where callerid would not be displayed on some phones.
Issue 8995, initial patch and research done by wedhorn
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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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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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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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r49636 | kpfleming | 2007-01-05 11:09:00 -0600 (Fri, 05 Jan 2007) | 10 lines
Merged revisions 49635 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r49635 | kpfleming | 2007-01-05 10:56:40 -0600 (Fri, 05 Jan 2007) | 2 lines
ensure that threads which are supposed to be detached (because we aren't going to wait on them) are created properly
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- Use this in chan_sip
- Document ha functions in acl.c
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r48888 | qwell | 2006-12-22 15:40:20 -0600 (Fri, 22 Dec 2006) | 2 lines
Note to self: Run make before committing...
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r48870 | qwell | 2006-12-22 14:43:05 -0600 (Fri, 22 Dec 2006) | 2 lines
Fix for issue 7774 - patch by alamantia
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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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r47053 | tilghman | 2006-11-02 17:49:13 -0600 (Thu, 02 Nov 2006) | 2 lines
More changes making the CLI more consistent with "category verb arguments" (continuation of issue 8236)
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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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be called for each thread specific object after they are allocated. Note that
there was already the ability to define a custom cleanup function. Also, if
the custom cleanup function is used, it *MUST* call free on the thread
specific object at the end. There is no way to have this magically done that
I can think of because the cleanup function registered with the pthread
implementation will only call the function back with a pointer to the
thread specific object, not the parent ast_threadstorage object.
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r44764 | qwell | 2006-10-09 11:12:35 -0500 (Mon, 09 Oct 2006) | 4 lines
Fix a problem where phones that go "missing" never got unregistered.
Issue #8067, reported by pj, patch by Anthony LaMantia (with minor whitespace modifications)
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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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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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r43650 | qwell | 2006-09-26 08:33:47 -0700 (Tue, 26 Sep 2006) | 11 lines
Add proper codec support to chan_skinny. Works with at least ulaw, alaw, and g729a.
This is technically a "new feature", but there are justifications for it.
I found a bug with the recent rtp packetization changes, which caused the media setup to
fail under certain circumstances, particularly when using allow=all, or having no allow=
statements (globally or on the device).
I could have either removed the rtp packetization features, or I could add proper codec
support (which, without, I think most people would consider to be a bug anyways).
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r43518 | qwell | 2006-09-22 15:12:12 -0700 (Fri, 22 Sep 2006) | 4 lines
Allow chan_skinny.so to be unloaded properly.
Remove reload support, since it doesn't actually...work.
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r43469 | qwell | 2006-09-21 20:01:16 -0700 (Thu, 21 Sep 2006) | 4 lines
First shot at unload_module in chan_skinny..
More to come.
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option for setting our own packetization as apposed to just doing
what is asked.
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Original patch by wedhorn, with modifications by me.
Issue #7588
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Thanks Josh.
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It still isn't variable length, but it does let you dial again.
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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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Also fix some locking issues I found at the same time.
Issue 7770, original patch by alamantia
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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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Thanks Russell.
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Patch by wedhorn with minor modifications by me.
Issue 7766
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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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