Commit Graph

162 Commits

Author SHA1 Message Date
David Vossel 862ebf4d00 fixes adaptive jitterbuffer configuration
When configuring the adaptive jitterbuffer, the target_extra
value not only could not be set from the configuration, but was
not even being set to its proper default.  This value is required
in order for the adaptive jitterbuffer to work correctly.  To resolve
this a config option has been added to expose this value to the conf
files, and a default value is provided when no config specific value
is present.



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@249893 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-03-02 19:08:38 +00:00
Jason Parker e1ec2df832 Fix mute toggling on OSS channels.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@229750 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-11-12 23:30:10 +00:00
Tilghman Lesher d8e0c58437 Expand codec bitfield from 32 bits to 64 bits.
Reviewboard: https://reviewboard.asterisk.org/r/416/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@227580 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-11-04 14:05:12 +00:00
Tilghman Lesher 642bec4d6f AST-2009-005
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@211539 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-08-10 19:20:57 +00:00
Kevin P. Fleming ba020fc390 Define side-effect-safe MIN and MAX macros and remove duplicate definitions from various files.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@209400 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-07-28 13:49:46 +00:00
Russell Bryant 0264eef115 Merge the new Channel Event Logging (CEL) subsystem.
CEL is the new system for logging channel events.  This was inspired after
facing many problems trying to represent what is possible to happen to a call
in Asterisk using CDR records.  For more information on CEL, see the built in
HTML or PDF documentation generated from the files in doc/tex/.

Many thanks to Steve Murphy (murf) and Brian Degenhardt (bmd) for their hard
work developing this code.  Also, thanks to Matt Nicholson (mnicholson) and
Sean Bright (seanbright) for their assistance in the final push to get this
code ready for Asterisk trunk.

Review: https://reviewboard.asterisk.org/r/239/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@203638 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-06-26 15:28:53 +00:00
Kevin P. Fleming e6b2e9a750 Const-ify the world (or at least a good part of it)
This patch adds 'const' tags to a number of Asterisk APIs where they are appropriate (where the API already demanded that the function argument not be modified, but the compiler was not informed of that fact). The list includes:

- CLI command handlers
- CLI command handler arguments
- AGI command handlers
- AGI command handler arguments
- Dialplan application handler arguments
- Speech engine API function arguments

In addition, various file-scope and function-scope constant arrays got 'const' and/or 'static' qualifiers where they were missing.

Review: https://reviewboard.asterisk.org/r/251/



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@196072 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-05-21 21:13:09 +00:00
Tilghman Lesher b0a29390ec Merged revisions 171187 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
  r171187 | tilghman | 2009-01-25 17:44:01 -0600 (Sun, 25 Jan 2009) | 6 lines
  
  Correctly track the hookstate
  (closes issue #13686)
   Reported by: itiliti
   Patches: 
         20081013__bug13686.diff.txt uploaded by Corydon76 (license 14)
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@171188 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-01-25 23:58:00 +00:00
Joshua Colp 654ea55a65 Numerous documentation updates.
(closes issue #13970)
Reported by: pkempgen
Patches:
      __20081217_cli_usage_fixes.patch.txt uploaded by blitzrage (license 10)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@165792 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-12-18 21:21:44 +00:00
Eliel C. Sardanons 1e8e12efcf Janitor, use ARRAY_LEN() when possible.
(closes issue #13990)
Reported by: eliel
Patches:
      array_len.diff uploaded by eliel (license 64)



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@161218 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-12-05 10:31:25 +00:00
Kevin P. Fleming 448562af93 improve configure script to remember the previous value of each dependency in build_tools/menuselect-deps, so that (once it has been written) menuselect can use this information to warn the user when a previously met dependency is no longer met
along the way, change tags used in configure script, menuselect-deps and code for various dependencies to be consistently named



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@154151 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-11-04 15:07:54 +00:00
Kevin P. Fleming bd4eb070f3 bring over all the fixes for the warnings found by gcc 4.3.x from the 1.4 branch, and add the ones needed for all the new code here too
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@153616 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-11-02 18:52:13 +00:00
Steve Murphy e235a07376 (closes issue #13557)
Reported by: nickpeirson
Patches:
      pbx.c.patch uploaded by nickpeirson (license 579)
      replace_bzero+bcopy.patch uploaded by nickpeirson (license 579)
Tested by: nickpeirson, murf

1. replaced all refs to bzero and bcopy to memset and memmove instead.
2. added a note to the CODING-GUIDELINES
3. add two macros to asterisk.h to prevent bzero, bcopy from creeping
   back into the source
4. removed bzero from configure, configure.ac, autoconfig.h.in




git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@147807 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-10-09 14:17:33 +00:00
Michiel van Baak 90751b16ca Merge the cli_cleanup branch.
This work is done by lmadsen, junky and mvanbaak
during AstriDevCon.

This is the second audit the CLI got, and
this time lmadsen made sure he had _ALL_ modules
loaded that have CLI commands in them.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@145121 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-09-28 23:32:14 +00:00
Steve Murphy e74584ca3c (closes issue #13557)
Reported by: nickpeirson

The user attached a patch, but the license is not yet
recorded. I took the liberty of finding and replacing
ALL index() calls with strchr() calls, and that
involves more than just main/pbx.c;

chan_oss, app_playback, func_cut also had calls
to index(), and I changed them out. 1.4 had no
references to index() at all.




git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@144569 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-09-25 22:21:28 +00:00
Tilghman Lesher 08af5bb312 Create a new config file status, CONFIG_STATUS_FILEINVALID for differentiating
when a file is invalid from when a file is missing.  This is most important when
we have two configuration files.  Consider the following example:

Old system:
sip.conf     users.conf     Old result               New result
========     ==========     ==========               ==========
Missing      Missing        SIP doesn't load         SIP doesn't load
Missing      OK             SIP doesn't load         SIP doesn't load
Missing      Invalid        SIP doesn't load         SIP doesn't load
OK           Missing        SIP loads                SIP loads
OK           OK             SIP loads                SIP loads
OK           Invalid        SIP loads incompletely   SIP doesn't load
Invalid      Missing        SIP doesn't load         SIP doesn't load
Invalid      OK             SIP doesn't load         SIP doesn't load
Invalid      Invalid        SIP doesn't load         SIP doesn't load

So in the case when users.conf doesn't load because there's a typo that
disrupts the syntax, we may only partially load users, instead of failing with
an error, which may cause some calls not to get processed.  Worse yet, the old
system would do this with no indication that anything was even wrong.

(closes issue #10690)
 Reported by: dtyoo
 Patches: 
       20080716__bug10690.diff.txt uploaded by Corydon76 (license 14)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@142992 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-09-12 23:30:03 +00:00
Mark Michelson 68bc5b446d Fix a memory leak in chan_oss
(closes issue #13311)
Reported by: eliel
Patches:
      chan_oss.c.patch uploaded by eliel (license 64)



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@141995 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-09-09 10:20:58 +00:00
Luigi Rizzo ff8ad6b497 implement the 'freeze' function for incoming frames;
fix a bug which caused a crash when a videodevice was
specified after startgui=1 in the config file. This also
involves a slightly different method to determine if the
gui is active or not.



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@126572 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-06-30 15:45:15 +00:00
Luigi Rizzo a768ae2122 implement a 'toggle' option for 'console mute' and 'console unmute'
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@126311 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-06-29 13:02:54 +00:00
Michiel van Baak f1e9371da8 - revert change to ast_queue_hangup and create ast_queue_hangup_with_cause
- 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


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@117802 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-05-22 16:29:54 +00:00
Michiel van Baak 08e674bce0 Pass the hangup cause all the way to the calling app/channel.
(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


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114637 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-04-24 22:16:48 +00:00
Russell Bryant 5bd4e32a4c Merged revisions 108796 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
r108796 | russell | 2008-03-14 15:09:22 -0500 (Fri, 14 Mar 2008) | 5 lines

Fix a channel name issue.  chan_oss registers the "Console" channel type,
but it created channels with an "OSS" prefix.

(closes issue #12194, reported by davidw, patched by me)

........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@108797 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-14 20:09:37 +00:00
Joshua Colp 496adc6fc0 Merged revisions 106235 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
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

........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@106239 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-03-05 22:43:22 +00:00
Luigi Rizzo bbbef73276 make get_video_desc() return the active console if
passed a null argument (channel).



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@97389 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-01-09 12:13:32 +00:00
Kevin P. Fleming e566f4eda7 eliminiate sound_thread() and other stuff from chan_oss since Asterisk indications can handle it
remove gentone and all the headers containing tones that are no longer needed


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@96270 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-01-03 19:56:29 +00:00
Luigi Rizzo fb487bbc89 remove useless casts
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@95068 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-12-27 23:00:55 +00:00
Luigi Rizzo 7a1dcff6a6 Build console_video support by linking in, as opposed to including,
console_video.c

This will ease the task of splitting console_video.c into its components
(V4L and X11 grabbers, various video codecs and packetizers, SDL),
as well as ease future extensions (e.g. additional video sources,
codecs and rendering engines).

For the time being nothing changes for users: video support is off by
default, and requires -DHAVE_VIDEO_CONSOLE on the command line to be included
(if SDL and FFMPEG are available).



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@94615 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-12-22 18:28:40 +00:00
Luigi Rizzo b7adaa023c add some macros to simplify parsing the config file,
see description in config.h .

They are a variant of the set of macros i used in chan_oss.c,
structured in a way to be more robust to the presence of
spurious ';' - basically, they define wrappers for 'do {'
and '} while (0)', plus some helper functions to deal with simple
cases such as ast_copy_string, ast_malloc, strtoul, ast_true ...

The prefix (CV_ as 'Config Variable') tries to be easy to remember
and has been chosen to not conflict with other existing macros in the tree.

For the time being, I have only updated the three source files in the
tree that used the old M_* macros. Hopefully, more files will be
converted.

NOTE:

    I understand that inventing my own dialect of C is generally wrong;
    however, the lack of adequate support in the language encourages
    lazy programming practices (such as ignoring errors, bounds, etc.)
    and this increases the chance of vulnerability in the code, especially
    because we are parsing user input here.
    Hopefully, these macros and the use of ast_parse_arg (in config.h)
    should encourage the programmer to write more robust code.



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@94191 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-12-20 12:56:07 +00:00
Luigi Rizzo 4a3c4095ba surprising as it may be, chan_oss compiles correctly under cygwin as well,
provided you look for soundcard.h in the right place...



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@93380 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-12-17 22:38:48 +00:00
Kevin P. Fleming 01f8a6959b make the configure script detect that it is running on a Windows platform, and report that information so that menuselect can use it (all information that is used to decide whether to build modules or not must be fed to menuselect so the user knows what will be built and why... don't make module build decisions in the makefiles, please)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@93211 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-12-17 17:11:06 +00:00
Luigi Rizzo d5b6ce0d6d remove some redundant headers
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@93143 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-12-15 00:11:04 +00:00
Luigi Rizzo 1a38b870cd put in the necessary hooks for video support in the console.
This is a NOP as far as the current code is concerned,
but there is already support in ./configure and the
Makefiles for the various libraries used by console_video.c
(not yet in the tree) so addition is trivial.



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89533 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-23 15:54:13 +00:00
Luigi Rizzo 7e8835e0d7 remove another set of redundant #include "asterisk/options.h"
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89512 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-21 23:24:55 +00:00
Luigi Rizzo 0595b5e2aa 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: https://origsvn.digium.com/svn/asterisk/trunk@89424 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-19 18:52:04 +00:00
Luigi Rizzo fdb7f7ba3d 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: https://origsvn.digium.com/svn/asterisk/trunk@89333 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-16 20:04:58 +00:00
Luigi Rizzo e6e98982c9 move the inner part of config file parsing to a separate function,
so it can be reused in the implementation of cli commands when
they have a similar syntax.



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89320 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-16 09:51:41 +00:00
Luigi Rizzo 7f8ecd2cd3 make the 'name' and 'value' fields in ast_variable const char *
This prevents modifying the strings in the stored variables, 
and catched a few instances where this was actually done.

Given the differences between trunk and 1.4 (and the fact that this
is effectively an API change) it is better to fix 1.4 independently.
These are

chan_sip.c::sip_register()
chan_skinny.c:: near line 2847
config.c:: near line 1774
logger.c::make_components()
res_adsi.c:: near line 1049

I may have missed some instances for modules that do not build here.



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89268 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-14 13:18:40 +00:00
Jason Parker ebe4050128 Switch from AST_CLI (formerly NEW_CLI) to AST_CLI_DEFINE, since the former didn't make much sense
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@86820 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-10-22 20:05:18 +00:00
Jason Parker b0f3e6097e Convert NEW_CLI to AST_CLI.
Closes issue #11039, as suggested by seanbright.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@86536 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-10-19 18:29:40 +00:00
Russell Bryant e97a723cf1 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: https://origsvn.digium.com/svn/asterisk/trunk@85460 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-10-11 19:03:06 +00:00
Tilghman Lesher 56b9568164 Don't reload a configuration file if nothing has changed.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@79747 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-08-16 21:09:46 +00:00
Joshua Colp d5eda8709c Merged revisions 79174 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

........
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.

........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@79175 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-08-13 14:22:46 +00:00
Joshua Colp 22114b509d Add support for using epoll instead of poll. This should increase scalability and is done in such a way that we should be able to add support for other poll() replacements.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@78683 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-08-08 21:44:58 +00:00
Russell Bryant f8483a0d04 Do a massive conversion for using the ast_verb() macro
(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");


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@77299 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-07-26 15:49:18 +00:00
Tilghman Lesher 9d05ff8ed5 Issue 9869 - replace malloc and memset with ast_calloc, and other coding guidelines changes
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@67864 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-06-06 21:20:11 +00:00
Steve Murphy 901413c76c Merged revisions 61644 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4

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r61644 | murf | 2007-04-13 11:01:02 -0600 (Fri, 13 Apr 2007) | 1 line

A fix for chan_oss that resulted from the CDR changes; it helps to use the right info.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@61646 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-04-13 17:11:53 +00:00
Steve Murphy ecaf781933 Merged revisions 60989 via svnmerge from
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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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@61152 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-04-10 05:41:34 +00:00
Kevin P. Fleming 74f31a51bc add -Wundef to the --enable-dev-mode flags, so that mistyped macro names in #if expressions will be caught
convert various #if expressions to #ifdef for macros that may not be defined (and where the value is not important)
Note: two of these changes are in bison generated files which is going to be inconvenient when they are regenerated


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@55329 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-02-18 15:03:42 +00:00
Joshua Colp ee3ab150f6 Merged revisions 51788 via svnmerge from
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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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@51801 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-01-23 22:59:55 +00:00
Russell Bryant dcca8f345f Merged revisions 51311 via svnmerge from
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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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@51314 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-01-19 18:06:03 +00:00