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20 authors
  1. Sep 02, 2010
  3. Jul 29, 2010
  5. Jul 28, 2010
  7. Jul 14, 2010
    • Richard Mudgett's avatar
      Expand the caller ANI field to an ast_party_id · cf7bbcc4
      Richard Mudgett authored 
      Expand the ani field in ast_party_caller and ast_party_connected_line to
an ast_party_id.

This is an extension to the ast_callerid restructuring patch in review:
https://reviewboard.asterisk.org/r/702/

Review: https://reviewboard.asterisk.org/r/744/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@276393 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      cf7bbcc4
    • Richard Mudgett's avatar
      ast_callerid restructuring · ec37ffbd
      Richard Mudgett authored 
      The purpose of this patch is to eliminate struct ast_callerid since it has
turned into a miscellaneous collection of various party information.

Eliminate struct ast_callerid and replace it with the following struct
organization:

struct ast_party_name {
	char *str;
	int char_set;
	int presentation;
	unsigned char valid;
};
struct ast_party_number {
	char *str;
	int plan;
	int presentation;
	unsigned char valid;
};
struct ast_party_subaddress {
	char *str;
	int type;
	unsigned char odd_even_indicator;
	unsigned char valid;
};
struct ast_party_id {
	struct ast_party_name name;
	struct ast_party_number number;
	struct ast_party_subaddress subaddress;
	char *tag;
};
struct ast_party_dialed {
	struct {
		char *str;
		int plan;
	} number;
	struct ast_party_subaddress subaddress;
	int transit_network_select;
};
struct ast_party_caller {
	struct ast_party_id id;
	char *ani;
	int ani2;
};

The new organization adds some new information as well.

* The party name and number now have their own presentation value that can
be manipulated independently.  ISDN supplies the presentation value for
the name and number at different times with the possibility that they
could be different.

* The party name and number now have a valid flag.  Before this change the
name or number string could be empty if the presentation were restricted.
Most channel drivers assume that the name or number is then simply not
available instead of indicating that the name or number was restricted.

* The party name now has a character set value.  SIP and Q.SIG have the
ability to indicate what character set a name string is using so it could
be presented properly.

* The dialed party now has a numbering plan value that could be useful to
have available.

The various channel drivers will need to be updated to support the new
core features as needed.  They have simply been converted to supply
current functionality at this time.


The following items of note were either corrected or enhanced:

* The CONNECTEDLINE() and REDIRECTING() dialplan functions were
consolidated into func_callerid.c to share party id handling code.

* CALLERPRES() is now deprecated because the name and number have their
own presentation values.

* Fixed app_alarmreceiver.c write_metadata().  The workstring[] could
contain garbage.  It also can only contain the caller id number so using
ast_callerid_parse() on it is silly.  There was also a typo in the
CALLERNAME if test.

* Fixed app_rpt.c using ast_callerid_parse() on the channel's caller id
number string.  ast_callerid_parse() alters the given buffer which in this
case is the channel's caller id number string.  Then using
ast_shrink_phone_number() could alter it even more.

* Fixed caller ID name and number memory leak in chan_usbradio.c.

* Fixed uninitialized char arrays cid_num[] and cid_name[] in
sig_analog.c.

* Protected access to a caller channel with lock in chan_sip.c.

* Clarified intent of code in app_meetme.c sla_ring_station() and
dial_trunk().  Also made save all caller ID data instead of just the name
and number strings.

* Simplified cdr.c set_one_cid().  It hand coded the ast_callerid_merge()
function.

* Corrected some weirdness with app_privacy.c's use of caller
presentation.

Review:	https://reviewboard.asterisk.org/r/702/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@276347 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      ec37ffbd
  9. Jun 03, 2010
  11. Mar 25, 2010
  13. Mar 02, 2010
    • David Vossel's avatar
      fixes adaptive jitterbuffer configuration · 862ebf4d
      David Vossel authored 
      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
      862ebf4d
  15. Feb 16, 2010
  17. Feb 08, 2010
  19. Aug 10, 2009
  21. Jul 28, 2009
  23. Jun 26, 2009
    • Russell Bryant's avatar
      Merge the new Channel Event Logging (CEL) subsystem. · 0264eef1
      Russell Bryant authored 
      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
      0264eef1
  25. Dec 05, 2008
  27. Nov 25, 2008
  29. Nov 19, 2008
  31. Nov 04, 2008
  33. Oct 09, 2008
    • Steve Murphy's avatar
      (closes issue #13557) · e235a073
      Steve Murphy authored 
      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
      e235a073
  35. May 22, 2008
  37. May 16, 2008
  39. Mar 31, 2008
  41. Mar 26, 2008
    • Jason Parker's avatar
      Large cleanup of DSP code · 6412a96e
      Jason Parker authored 
      Per comments from dimas:
1. The code now generates DTMF_BEGIN frames in addition to DTMF_END ones.

2. "quelching" rewritten - now each detector (MF/DTMF/generic tone) may mark fragment of a frame for suppression (squelching, muting) with a call to mute_fragment. Actual muting happens only once at the very end of ast_dsp_process where all marked fragments are zeroed. This way every detector sees original data in the frame without any piece of a frame being zeroed by a detector which was run before.

3. DTMF detector tries to "mute" one block before and one block after the block where actual tone was detected. Muting of previois block is something new for this patch. Obviously this operation is not always possible - if current frame does not contain data for previous block - it is too late. But at least we make our best.
Muting of next block was already done by the old code but it only affects part of the next block which is in the frame being processed. New code keeps this information in state structures so it will mute proper number of samples in the next frame(s) too.

4. Removed ast_dsp_digitdetect and ast_dsp_getdigits APIs because these are not used.

5. DSP API extended a bit - ast_dsp_was_muted() function added which returns true if DSP code was muting any fragment in the last frame. chan_zap uses this function to decide it needs to turn on confmute on the channel.
This is to replace AST_FRAME_DTMF 'm'/'u' (mute/unmute) functionality.


(closes issue #11968)
Reported by: dimas
Patches:
      v2-11968-dsp.patch uploaded by dimas (license 88)
      v4-11968-zap.patch uploaded by dimas (license 88)
Tested by: dimas, qwell


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@111022 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      6412a96e
  43. Mar 19, 2008
  45. Feb 08, 2008
    • Russell Bryant's avatar
      Merge changes from team/mvanbaak/cli-command-audit · 1ec8cb41
      Russell Bryant authored 
      (closes issue #8925)

About a year ago, as Leif Madsen and Jim van Meggelen were going over the CLI
commands in Asterisk 1.4 for the next version of their book, they documented
a lot of inconsistencies.  This set of changes addresses all of these issues
and has been reviewed by Leif.

While this does introduce even more changes to the CLI command structure, it
makes everything consistent, which is the most important thing.

Thanks to all that helped with this one!


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@103171 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      1ec8cb41
  47. Feb 05, 2008
  49. Jan 05, 2008
  51. Dec 20, 2007
    • Luigi Rizzo's avatar
      add some macros to simplify parsing the config file, · b7adaa02
      Luigi Rizzo authored 
      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
      b7adaa02
  53. Dec 04, 2007
  55. Nov 25, 2007
  57. Nov 21, 2007
  59. Nov 19, 2007
  61. Nov 16, 2007
    • Luigi Rizzo's avatar
      Start untangling header inclusion in a way that does not affect · fdb7f7ba
      Luigi Rizzo authored 
      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
      fdb7f7ba
  63. Oct 22, 2007