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  1. Jul 14, 2010
    • 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
  2. Apr 29, 2009
    • Tilghman Lesher's avatar
      Merge str_substitution branch. · a866a759
      Tilghman Lesher authored
      This branch adds additional methods to dialplan functions, whereby the result
      buffers are now dynamic buffers, which can be expanded to the size of any
      result.  No longer are variable substitutions limited to 4095 bytes of data.
      In addition, the common case of needing buffers much smaller than that will
      enable substitution to only take up the amount of memory actually needed.
      The existing variable substitution routines are still available, but users
      of those API calls should transition to using the dynamic-buffer APIs.
      Reviewboard: http://reviewboard.digium.com/r/174/
      
      
      git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@191140 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      a866a759
  3. Nov 05, 2008
  4. Nov 01, 2008
  5. Nov 22, 2007
  6. Nov 21, 2007
  7. Nov 19, 2007
  8. Nov 16, 2007
  9. Jul 16, 2007
  10. Jan 07, 2007
  11. Jan 06, 2007
  12. Sep 21, 2006
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