Age | Commit message (Collapse) | Author |
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r77424 | kpfleming | 2007-07-26 17:14:21 -0500 (Thu, 26 Jul 2007) | 2 lines
use new canonical name for download server
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r77429 | kpfleming | 2007-07-26 17:16:42 -0500 (Thu, 26 Jul 2007) | 2 lines
change protocol for downloads as well
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r77410 | russell | 2007-07-26 16:23:23 -0500 (Thu, 26 Jul 2007) | 10 lines
AST_DEVMODE was defined in trunk, but not in 1.4. When Asterisk is compiled
under dev mode, AST_DEVMODE will get defined in buildopts.h. Change 1.4 to
define it in the same way that trunk does. Also, revert the change that added
this define in the Makefile
The advantage to doing it this way is that buildopts.h gets installed when
you install Asterisk. Then, when building any out of tree modules, or
building asterisk-addons, these modules know which options the rest of Asterisk
was built with.
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r77380 | mmichelson | 2007-07-26 15:35:17 -0500 (Thu, 26 Jul 2007) | 7 lines
Fixes to get ast_backtrace working properly. The AST_DEVMODE macro was never defined so the majority of ast_backtrace never
attempted compilation. The makefile now defines AST_DEVMODE if configure was run with --enable-dev-mode. Also, changes were
made to acccomodate 64 bit systems in ast_backtrace.
Thanks to qwell, kpfleming, and Corydon76 for their roles in allowing me to get this committed
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r77350 | tilghman | 2007-07-26 14:32:17 -0500 (Thu, 26 Jul 2007) | 2 lines
Missed one
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r77348 | tilghman | 2007-07-26 14:27:18 -0500 (Thu, 26 Jul 2007) | 2 lines
Oops, that builtin define should be all-lowercase.
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r77318 | mmichelson | 2007-07-26 13:30:29 -0500 (Thu, 26 Jul 2007) | 8 lines
Two consecutive calls to PQfinish could occur, meaning free gets called on the same variable twice.
This patch sets the connection to NULL after calls to PQfinish so that the problem does not occur.
Also in this patch, prashant_jois informed me that it is safe to pass a null pointer to PQfinish, so
I have removed the check for conn's existence from my_unload_module.
(closes issue 10295, reported by junky, patched by me with input from prashant_jois)
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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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where verbatim text went off the end of the page on the PDF, and various
other improvements
(closes issue #10307, IgorG)
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* random() is defined to return a positive result, and now ast_random()
will always do so as well
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(closes issue #10308, reported by yehavi, patched by me)
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(closes #10308)
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and why this does not overwrite useful information.
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AEL; I commented out the code that substitutes commas with vertbars, so we can get apps to parse their args correctly.
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r77191 | murf | 2007-07-25 16:39:27 -0600 (Wed, 25 Jul 2007) | 1 line
This fix solves problem with intense squelch noise when someone joins conf in bug 9430; We repro'd the problem with meetme opts of 'CciMo'; Josh Colp supplied this patch, and I'm applying it. It looks like playing the recorded username will louse up the next thing played into the channel. Josh rearranged the code so as to start things over before playing data directly into the conference.
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r77176 | file | 2007-07-25 19:16:10 -0300 (Wed, 25 Jul 2007) | 4 lines
(closes issue #10303)
Reported by: jtodd
Add SPEECH_DTMF_TERMINATOR variable so the user can specify the digit to terminate a DTMF string with. If none is specified then no terminator will be used.
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At first sight (but the function is very large so i am not 100% sure)
the code seems correct, so maybe my compiler is just not smart
enough to figure that out at the optimization level it has.
Not worthwhile merging to 1.4 i believe.
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r77154 | mmichelson | 2007-07-25 16:52:47 -0500 (Wed, 25 Jul 2007) | 3 lines
chan->emulate_dtmf_duration is an unsigned int, not a signed int, so use %u instead of %d in the format string
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r77071 | file | 2007-07-25 14:14:14 -0300 (Wed, 25 Jul 2007) | 2 lines
Fix autoconf logic for finding OpenH323 when it is not in the first place searched (/usr/share/openh323).
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(3-deep nested loop)
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r77022 | rizzo | 2007-07-25 11:34:01 +0200 (Wed, 25 Jul 2007) | 3 lines
set the sequence number in a frame for all frame types
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section. Also, align the NAT/STUN entry with the others.
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r76983 | murf | 2007-07-24 18:18:32 -0600 (Tue, 24 Jul 2007) | 9 lines
Merged revisions 76978 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r76978 | murf | 2007-07-24 18:07:24 -0600 (Tue, 24 Jul 2007) | 1 line
this fixes bug 10293, where the error message because defaultzone or loadzone was not defined was confusing
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r76937 | tilghman | 2007-07-24 17:12:43 -0500 (Tue, 24 Jul 2007) | 10 lines
Merged revisions 76934 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r76934 | tilghman | 2007-07-24 17:11:33 -0500 (Tue, 24 Jul 2007) | 2 lines
Oops, res contains the error code, not errno. I was wondering why a mutex was reporting "No such file or directory"...
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r76891 | tilghman | 2007-07-24 15:42:05 -0500 (Tue, 24 Jul 2007) | 2 lines
Found another place where we should be using the umask (thanks jcmoore)
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r76803 | qwell | 2007-07-24 11:32:20 -0500 (Tue, 24 Jul 2007) | 3 lines
Don't create the Asterisk channel until we are starting the PBX on it.
(ASA-2007-018)
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r76801 | mmichelson | 2007-07-24 11:26:58 -0500 (Tue, 24 Jul 2007) | 13 lines
Added a membercount variable to call_queue struct which keeps track of the number of logged in members in a particular queue.
This makes it so that the 'n' option for Queue() can act properly depending on which strategy is used. If the strategy is
roundrobin, rrmemory, or ringall, we want to ring each phone once before moving on in the dialplan. However, if any other strategy is
used, we will only ring one phone since it cannot be guaranteed that a different phone will ring on subsequent attempts to ring a phone.
As a side effect of this, the QUEUE_MEMBER_COUNT dialplan function now just reads the membercount variable instead of traversing through
the member list to figure out how many members there are.
Special thanks to blitzrage for helping to test this out.
(closes issue #10127, reported by bcnit, patched by me, tested by blitzrage)
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This allows you to just Dial(Skinny/line), as long as line isn't ambiguous.
Note that this does not remove or deprecate the "old" syntax, as it's still
quite useful - even moreso if shared lines get implemented.
Initial patch by me, with some changes and suggestions from wedhorn.
(closes issue #10263)
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+ externexpire was not initialized properly;
+ stunaddr was not handled properly on a sip reload
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really static... and don't call AST_RWLIST_HEAD_DESTROY on a statically declared list.
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it's just a double check.
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r76708 | tilghman | 2007-07-23 17:38:06 -0500 (Mon, 23 Jul 2007) | 4 lines
It was our stated intention for 1.4 that files created in app_voicemail should
depend upon the umask. Unfortunately, mkstemp() creates files with mode 0600,
regardless of the umask. This corrects that deficiency.
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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.
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should also be thread safe now.
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documentation on the
change to the Local channel name.
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applications
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).
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r76656 | qwell | 2007-07-23 13:59:28 -0500 (Mon, 23 Jul 2007) | 3 lines
Fix some incorrect softkey labels in messages.
Don't try to play dialtone in some unimplemented features.
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r76654 | file | 2007-07-23 15:29:48 -0300 (Mon, 23 Jul 2007) | 12 lines
Merged revisions 76653 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r76653 | file | 2007-07-23 15:28:13 -0300 (Mon, 23 Jul 2007) | 4 lines
(closes issue #5866)
Reported by: tyler
Do not force channel format changes when a generator is present. The generator may have changed the formats itself and changing them back would cause issues.
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https://origsvn.digium.com/svn/asterisk/branches/1.4
(closes issue #10276)
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r76620 | qwell | 2007-07-23 12:57:53 -0500 (Mon, 23 Jul 2007) | 4 lines
Don't try to queue up hold/unhold frames on a non-existent channel.
Issue 10276.
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r76618 | file | 2007-07-23 14:48:51 -0300 (Mon, 23 Jul 2007) | 2 lines
Allow app_morsecode to build on PPC Linux by putting the value of the digit char in an int.
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does not use DTMF BEGIN frames.
1.4 seems correct (it does not have the two fields).
However, as this bug shows, the current way of creating the sip_tech
replica is too error-prone, one can easily forget to update one of
the two entries. Perhaps it would be better to create sip_tech_info
expliclty at module load, by doing
sip_tech_info = sip_tech;
sip_tech_info.send_digit_begin = NULL
(in this case, this is something applicable to 1.4 as well).
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r76561 | file | 2007-07-23 11:34:21 -0300 (Mon, 23 Jul 2007) | 14 lines
Merged revisions 76560 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r76560 | file | 2007-07-23 11:32:07 -0300 (Mon, 23 Jul 2007) | 6 lines
(closes issue #10236)
Reported by: homesick
Patches:
rpid_1.4_75840.patch uploaded by homesick (license 91)
Accept Remote Party ID on guest calls.
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