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Author SHA1 Message Date
tilghman 35d428bcca Convert ast_verbose to ast_verb.
Reported by: snuffy
Patch by: snuffy
(Closes issue #11547)


git-svn-id: http://svn.digium.com/svn/asterisk/trunk@92913 f38db490-d61c-443f-a65b-d21fe96a405b
2007-12-14 14:48:38 +00:00
tilghman 8a1ce32239 Conversions of free to ast_free, where applicable, and several other formatting fixes.
Reported by: eliel
Patch by: eliel,tilghman
(Closes issue #11209)


git-svn-id: http://svn.digium.com/svn/asterisk/trunk@92594 f38db490-d61c-443f-a65b-d21fe96a405b
2007-12-12 20:05:13 +00:00
file c585cac92c Merged revisions 89587 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r89587 | file | 2007-11-26 13:20:58 -0400 (Mon, 26 Nov 2007) | 6 lines

Close the audio file before sending it to the post processing application.
(closes issue #11357)
Reported by: reformed
Patches:
      mixmonitor.patch uploaded by reformed (license 330)

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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89589 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-26 17:23:28 +00:00
rizzo e8a5f98fe8 shuffle a little bit the content of header files to reduce dependencies.
In this commit:
- move the ast_register/unregister_app functions to module.h
  to avoid the need to include pbx.h for the simpler apps;
- move the ast_group structure to channel.h to remove the
  dependency of app.h on linkedlists.h

Note, this is a long process that I am doing in small steps.

The main difficulty is that now for each subsystem we
have a single header (e.g. channel.h) included by the subsystem
provider (usually one file, e.g. channel.c) and by its clients
(dozens of them, e.g. we have some 70+ apps and 30+ functions).

This requires the clients to include all the extra headers
required by the provider (eg. lock.h, linkedlists.h, definitions
of substructures...) even though many of the clients would be
just happy with opaque struct declarations and function prototypes.

The long term plan is to eventually rectify this structure
so that the compilation can become faster, and also APIs
are more stable.



git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89522 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-22 03:50:04 +00:00
rizzo 737b408d52 more removal of redundant headers
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89519 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-22 02:07:33 +00:00
rizzo 150b2c22ef remove another set of redundant #include "asterisk/options.h"
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89512 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-21 23:24:55 +00:00
rizzo 89d8d78652 move asterisk/paths.h outside asterisk.h and into those files
who really need it.



git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89466 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-20 23:16:15 +00:00
rizzo 9cf442d7f7 include "logger.h" and errno.h from asterisk.h - usage shows that they
were included almost everywhere.
Remove some of the instances.



git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89424 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-19 18:52:04 +00:00
rizzo 883346d64a Start untangling header inclusion in a way that does not affect
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.).



git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89333 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-16 20:04:58 +00:00
mmichelson dda901eb6f Merged revisions 89241 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r89241 | mmichelson | 2007-11-13 10:02:02 -0600 (Tue, 13 Nov 2007) | 5 lines

Reverting commit made in revision 89205 since it is unnecessary.

Thanks to Kevin for pointing this out


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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89242 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-13 16:03:10 +00:00
mmichelson fdc273032a There is the potential to copy uninitialized memory into the mixmonitor->post_process
string. This fix prevents that.



git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89207 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-13 01:19:53 +00:00
mmichelson 5dfa041c88 Merged revisions 89205 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r89205 | mmichelson | 2007-11-12 18:56:46 -0600 (Mon, 12 Nov 2007) | 5 lines

Some sanity checking for MixMonitor. If only 1 argument is given, then the args.options
and args.post_process strings are uninitialized and could contain garbage. This change
handles this situation properly by only using arguments that we have parsed.


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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89206 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-13 00:57:34 +00:00
murf 91879350be Based on a note in asterisk-dev by Brian Capouch, I determined I too agressive in not initializing arrays passed to pbx_substitute_variables_xxxx; I reviewed the code (again) and hopefully found every possible spot where substitute_variables is called conditionally, and made sure the char array involved was set to a null string.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89186 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-12 18:44:36 +00:00
mmichelson 92ac6820ee "show application <foo>" changes for clarity.
(closes issue #11171, reported and patched by blitzrage)

Many thanks!



git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89044 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-06 19:04:45 +00:00
murf 47c8ea00b8 This commits the performance mods that give the priority processing engine in the pbx, a 25-30% speed boost. The two updates used, are, first, to merge the ast_exists_extension() and the ast_spawn_extension() where they are called sequentially in a loop in the code, into a slightly upgraded version of ast_spawn_extension(), with a few extra args; and, second, I modified the substitute_variables_helper_full, so it zeroes out the byte after the evaluated string instead of demanding you pre-zero the buffer; I also went thru the code and removed the code that zeroed this buffer before every call to the substitute_variables_helper_full. The first fix provides about a 9% speedup, and the second the rest. These figures come from the 'PIPS' benchmark I describe in blogs, conf. reports, etc.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@88166 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-01 22:26:51 +00:00
file 7af6fda1e0 Add volume adjustment in.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@87851 f38db490-d61c-443f-a65b-d21fe96a405b
2007-10-31 18:03:17 +00:00
file 27c7c3edf5 Restore operation of the option that only writes when the channel is bridged.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@87850 f38db490-d61c-443f-a65b-d21fe96a405b
2007-10-31 17:56:21 +00:00
qwell 7756b987a0 Switch from AST_CLI (formerly NEW_CLI) to AST_CLI_DEFINE, since the former didn't make much sense
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@86820 f38db490-d61c-443f-a65b-d21fe96a405b
2007-10-22 20:05:18 +00:00
qwell d542122e6a Convert NEW_CLI to AST_CLI.
Closes issue #11039, as suggested by seanbright.


git-svn-id: http://svn.digium.com/svn/asterisk/trunk@86536 f38db490-d61c-443f-a65b-d21fe96a405b
2007-10-19 18:29:40 +00:00
russell 13b9c5237c Merge a ton of NEW_CLI conversions. Thanks to everyone that helped out! :)
(closes issue #10724)
Reported by: eliel
Patches: 
      chan_skinny.c.patch uploaded by eliel (license 64)
      chan_oss.c.patch uploaded by eliel (license 64)
      chan_mgcp.c.patch2 uploaded by eliel (license 64)
      pbx_config.c.patch uploaded by seanbright (license 71)
      iax2-provision.c.patch uploaded by eliel (license 64)
      chan_gtalk.c.patch uploaded by eliel (license 64)
      pbx_ael.c.patch uploaded by seanbright (license 71)
      file.c.patch uploaded by seanbright (license 71)
      image.c.patch uploaded by seanbright (license 71)
      cli.c.patch uploaded by moy (license 222)
      astobj2.c.patch uploaded by moy (license 222)
      asterisk.c.patch uploaded by moy (license 222)
      res_limit.c.patch uploaded by seanbright (license 71)
      res_convert.c.patch uploaded by seanbright (license 71)
      res_crypto.c.patch uploaded by seanbright (license 71)
      app_osplookup.c.patch uploaded by seanbright (license 71)
      app_rpt.c.patch uploaded by seanbright (license 71)
      app_mixmonitor.c.patch uploaded by seanbright (license 71)
      channel.c.patch uploaded by seanbright (license 71)
      translate.c.patch uploaded by seanbright (license 71)
      udptl.c.patch uploaded by seanbright (license 71)
      threadstorage.c.patch uploaded by seanbright (license 71)
      db.c.patch uploaded by seanbright (license 71)
      cdr.c.patch uploaded by moy (license 222)
      pbd_dundi.c.patch uploaded by moy (license 222)
      app_osplookup-rev83558.patch uploaded by moy (license 222)
      res_clioriginate.c.patch uploaded by moy (license 222)


git-svn-id: http://svn.digium.com/svn/asterisk/trunk@85460 f38db490-d61c-443f-a65b-d21fe96a405b
2007-10-11 19:03:06 +00:00
file ce30d7306b Merge audiohooks branch into trunk. This is a new API for developers to listen and manipulate the audio going through a channel.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@78649 f38db490-d61c-443f-a65b-d21fe96a405b
2007-08-08 19:30:52 +00:00
tilghman 356721a45c Mostly cleanup of documentation to substitute the pipe with the comma, but a few other formatting cleanups, too.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@77808 f38db490-d61c-443f-a65b-d21fe96a405b
2007-07-31 01:10:47 +00:00
murf 77f799ff1e After some study, thought, comparing, etc. I've backed out the previous universal mod to make ast_flags a 64 bit thing. Instead, I added a 64-bit version of ast_flags (ast_flags64), and 64-bit versions of the test-flag, set-flag, etc. macros, and an app_parse_options64 routine, and I use these in app_dial alone, to eliminate the 30-option limit it had grown to meet. There is room now for 32 more options and flags. I was heavily tempted to implement some of the other ideas that were presented, but this solution does not intro any new versions of dial, doesn't have a different API, has a minimal/zero impact on code outside of dial, and doesn't seriously (I hope) affect the code structure of dial. It's the best I can think of right now. My goal was NOT to rewrite dial. I leave that to a future, coordinated effort.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@75983 f38db490-d61c-443f-a65b-d21fe96a405b
2007-07-19 23:24:27 +00:00
murf cdfb9990ad via 10206, I have added an option (e) to Dial to allow the h exten to get run on peer. Had to upgrade ast_flag stuff to 64 bits to do this.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@75400 f38db490-d61c-443f-a65b-d21fe96a405b
2007-07-17 19:40:29 +00:00
file d17ff1ea42 Applications no longer need to call ast_module_user_add and ast_module_user_remove. This is now taken care of in the pbx_exec function outside of the application.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@75200 f38db490-d61c-443f-a65b-d21fe96a405b
2007-07-16 14:39:29 +00:00
file 9e24ed5ccf It is no longer required for each module that deals with a channel to call ast_module_user_hangup_all in it's unload function. The loader will automatically perform this action for it.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@75183 f38db490-d61c-443f-a65b-d21fe96a405b
2007-07-16 13:35:20 +00:00
file b663e2cc3f Merged revisions 72381 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r72381 | file | 2007-06-27 19:25:12 -0400 (Wed, 27 Jun 2007) | 10 lines

Merged revisions 72378 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.2

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r72378 | file | 2007-06-27 19:24:01 -0400 (Wed, 27 Jun 2007) | 2 lines

Update documentation to clarify variable usage with MixMonitor. (issue #9494 reported by netoguy)

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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@72382 f38db490-d61c-443f-a65b-d21fe96a405b
2007-06-27 23:26:46 +00:00
tilghman 5e53d26887 Issue 9970 - Ensure directory exists before trying to write an output file
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@71268 f38db490-d61c-443f-a65b-d21fe96a405b
2007-06-24 16:20:18 +00:00
tilghman eb5d461ed4 Issue 9869 - replace malloc and memset with ast_calloc, and other coding guidelines changes
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@67864 f38db490-d61c-443f-a65b-d21fe96a405b
2007-06-06 21:20:11 +00:00
file 5d89164d9a Add support for autocompleting start/stop options of the mixmonitor CLI command. (issue #9862 reported by eliel)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@66998 f38db490-d61c-443f-a65b-d21fe96a405b
2007-06-04 11:48:01 +00:00
russell a42bc96f14 Add a new API call for creating detached threads. Then, go replace all of the
places in the code where the same block of code for creating detached threads
was replicated.  (patch from bbryant)


git-svn-id: http://svn.digium.com/svn/asterisk/trunk@65968 f38db490-d61c-443f-a65b-d21fe96a405b
2007-05-24 18:30:19 +00:00
file 74cf1a7bb8 Merged revisions 52717 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r52717 | file | 2007-01-29 18:43:40 -0500 (Mon, 29 Jan 2007) | 10 lines

Merged revisions 52716 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.2

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r52716 | file | 2007-01-29 18:39:39 -0500 (Mon, 29 Jan 2007) | 2 lines

Now that filename is part of the structure and since it comes before postprocess... we have to add it to our postprocess line. (reported on asterisk-dev by Boris Bakchiev)

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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@52718 f38db490-d61c-443f-a65b-d21fe96a405b
2007-01-29 23:45:39 +00:00
file a25ad350e7 Merged revisions 52163 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r52163 | file | 2007-01-24 20:51:35 -0500 (Wed, 24 Jan 2007) | 10 lines

Merged revisions 52162 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.2

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r52162 | file | 2007-01-24 20:48:52 -0500 (Wed, 24 Jan 2007) | 2 lines

Add another note about audio files being played back to each bridged party. (issue #8718 reported by ppyy)

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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@52168 f38db490-d61c-443f-a65b-d21fe96a405b
2007-01-25 01:54:39 +00:00
file e50a0640bd Merged revisions 51407 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r51407 | file | 2007-01-22 14:13:44 -0500 (Mon, 22 Jan 2007) | 10 lines

Merged revisions 51406 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/branches/1.2

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r51406 | file | 2007-01-22 14:08:52 -0500 (Mon, 22 Jan 2007) | 2 lines

Move filestream creation to Mixmonitor loop. This will prevent a blank file from being created if no frames ever pass through to be recorded. (issue #7589 reported by steve_mcneil)

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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@51408 f38db490-d61c-443f-a65b-d21fe96a405b
2007-01-22 19:22:07 +00:00
murf 322326058e a quick fix to app_sms.c to get rid of cursed compiler warnings so I can compile under --enable-dev-mode
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@48767 f38db490-d61c-443f-a65b-d21fe96a405b
2006-12-21 19:44:20 +00:00
kpfleming 1a08d9e31b Merged revisions 44378 via svnmerge from
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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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@44379 f38db490-d61c-443f-a65b-d21fe96a405b
2006-10-04 19:51:38 +00:00
kpfleming 5aacb6a82d merge qwell's CLI verbification work
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@43212 f38db490-d61c-443f-a65b-d21fe96a405b
2006-09-18 19:54:18 +00:00
file c345657f48 Make the difference clear about what the responsibilities of the core and a spy are when it comes to spying on a channel. The core is responsible for adding a spy to a channel, feeding frames into the spy, removing the spy from the channel, and notifying the spy that is has been detached. The spy is responsible for reading frames in, and cleaning itself up. Each side will not try to do the other's job.
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@41959 f38db490-d61c-443f-a65b-d21fe96a405b
2006-09-03 23:30:37 +00:00
russell 53f8d43e29 Merge team/russell/frame_caching
There are some situations in Asterisk where ast_frame and/or iax_frame
structures are rapidly allocatted and freed (at least 50 times per second
for one call).

This code significantly improves the performance of ast_frame_header_new(), 
ast_frdup(), ast_frfree(), iax_frame_new(), and iax_frame_free() by keeping
a thread-local cache of these structures and using frames from the cache 
whenever possible instead of calling malloc/free every time.

This commit also converts the ast_frame and iax_frame structures to use the
linked list macros.


git-svn-id: http://svn.digium.com/svn/asterisk/trunk@41278 f38db490-d61c-443f-a65b-d21fe96a405b
2006-08-29 20:50:36 +00:00
kpfleming 8b0c007ad9 merge new_loader_completion branch, including (at least):
- 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)



git-svn-id: http://svn.digium.com/svn/asterisk/trunk@40722 f38db490-d61c-443f-a65b-d21fe96a405b
2006-08-21 02:11:39 +00:00
kpfleming 2e887fc053 Merged revisions 33841 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2

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r33841 | kpfleming | 2006-06-13 08:30:06 -0500 (Tue, 13 Jun 2006) | 2 lines

memory allocation optimizations

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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@33842 f38db490-d61c-443f-a65b-d21fe96a405b
2006-06-13 13:40:58 +00:00
file 0b6398c918 Merged revisions 33724 via svnmerge from
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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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@33725 f38db490-d61c-443f-a65b-d21fe96a405b
2006-06-12 21:37:29 +00:00
kpfleming 73c525e6e2 simplify autoconfig include mechanism (make tholo happy he can use lint again :-)
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@32846 f38db490-d61c-443f-a65b-d21fe96a405b
2006-06-07 18:54:56 +00:00
russell d99b677f35 remove almost all of the checks of the result from ast_strdupa() or alloca().
As it turns out, all of these checks were useless, because alloca will never
return NULL.


git-svn-id: http://svn.digium.com/svn/asterisk/trunk@26451 f38db490-d61c-443f-a65b-d21fe96a405b
2006-05-10 13:22:15 +00:00
russell bb7e5c2fa8 use ast_channel_(un)lock
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@25450 f38db490-d61c-443f-a65b-d21fe96a405b
2006-05-08 11:30:32 +00:00
russell f0e9d7485d remove a check of the result from ast_mutex_lock
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@25446 f38db490-d61c-443f-a65b-d21fe96a405b
2006-05-08 11:26:01 +00:00
rizzo 3664249356 This rather large commit changes the way modules are loaded.
As partly documented in loader.c and include/asterisk/module.h,
modules are now expected to return all of their methods and flags
into a structure 'mod_data', and are normally loaded with RTLD_NOW
| RTLD_LOCAL, so symbols are resolved immediately and conflicts
should be less likely.  Only in a small number of cases (res_*,
typically) modules are loaded RTLD_GLOBAL, so they can export
symbols.
 
The core of the change is only the two files loader.c and
include/asterisk/module.h, all the rest is simply adaptation of the
existing modules to the new API, a rather mechanical (but believe
me, time and finger-consuming!) process whose detail you can figure
out by svn diff'ing any single module.

Expect some minor compilation issue after this change, please
report it on mantis http://bugs.digium.com/view.php?id=6968
so we collect all the feedback in one place.

I am just sorry that this change missed SVN version number 20000!



git-svn-id: http://svn.digium.com/svn/asterisk/trunk@20003 f38db490-d61c-443f-a65b-d21fe96a405b
2006-04-14 14:08:19 +00:00
kpfleming e4880150b1 since the module API is changing, it's a good time to const-ify the description() and key() return values
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@18552 f38db490-d61c-443f-a65b-d21fe96a405b
2006-04-08 22:01:19 +00:00
bweschke 09e80638b5 Merged revisions 11778 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2

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r11778 | bweschke | 2006-03-04 06:45:37 -0500 (Sat, 04 Mar 2006) | 3 lines

 Substitute variables in the post_process string (if it exists) before those variables could possibly disappear (channel hangup) #6462


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git-svn-id: http://svn.digium.com/svn/asterisk/trunk@11779 f38db490-d61c-443f-a65b-d21fe96a405b
2006-03-04 11:51:45 +00:00
russell 011fac8ff8 fix up the summary and help text for the mixmonitor cli command
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@10392 f38db490-d61c-443f-a65b-d21fe96a405b
2006-02-17 04:17:35 +00:00