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20 authors
  1. Jul 16, 2012
  3. Feb 28, 2012
  5. Jul 21, 2011
  7. Jul 15, 2011
  9. Jul 13, 2011
  11. Jul 08, 2011
  13. Jul 06, 2011
    • Terry Wilson's avatar
      Replace Berkeley DB with SQLite 3 · efd040cd
      Terry Wilson authored 
      There were some bugs in the very ancient version of Berkeley DB that Asterisk
used. Instead of spending the time tracking down the bugs in the Berkeley code
we move to the much better documented SQLite 3.

Conversion of the old astdb happens at runtime by running the included
astdb2sqlite3 utility. The ast_db API with SQLite 3 backend should behave
identically to the old Berkeley backend, but in the future we could offer a
much more robust interface.

We do not include the SQLite 3 library in the source tree, but instead rely
upon the distribution-provided libraries. SQLite is so ubiquitous that this
should not place undue burden on administrators.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@326589 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      efd040cd
  15. Jun 29, 2011
    • Gregory Nietsky's avatar
      · f99a06d0
      Gregory Nietsky authored 
      Commit "distrotech" app_queue changes to Trunk

 * Added general option negative_penalty_invalid default off. when set
   members are seen as invalid/logged out when there penalty is negative.  
   for realtime members when set remove from queue will set penalty to -1.  
 * Added queue option autopausedelay when autopause is enabled it will be
   delayed for this number of seconds since last successful call if there
   was no prior call the agent will be autopaused immediately.
 * Added member option ignorebusy this when set and ringinuse is not   
   will allow per member control of multiple calls as ringinuse does for
   the Queue.
  
 - Mark QUEUE_MEMBER_PENALTY Deprecated it never worked for realtime members
 - QUEUE_MEMBER is now R/W supporting setting paused, ignorebusy and penalty.

(closes issue ASTERISK-17421)
(closes issue ASTERISK-17391)
Reported by: irroot
Tested by: irroot, jrose
Review: https://reviewboard.asterisk.org/r/1119/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@325483 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      f99a06d0
  17. May 27, 2011
  19. May 06, 2011
  21. May 05, 2011
  23. Apr 21, 2011
  25. Jan 04, 2011
  27. Nov 02, 2010
  29. Oct 29, 2010
  31. Jul 23, 2010
  33. Jul 21, 2010
  35. 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
  37. Jul 06, 2010
  39. Jun 08, 2010
  41. Jun 03, 2010
  43. Jun 01, 2010
  45. May 28, 2010
  47. May 07, 2010
  49. Apr 21, 2010
  51. Feb 12, 2010
  53. Jan 18, 2010
  55. Dec 09, 2009
  57. Dec 02, 2009
  59. Nov 13, 2009
  61. Nov 02, 2009
    • Richard Mudgett's avatar
      DAHDI ISDN channel names will not allow device state to work. (Interim solution.) · 6406f395
      Richard Mudgett authored 
      Since ISDN works like SIP and not analog ports in regard to devices, the
device state based on the ISDN channel number could not work.  This has
not been an issue until the advent of PTMP NT mode.  Previously, ISDN
lines were used as trunks and did not have to keep track of specific
devices.

As an interim solution until device states are properly implemented, the
channel name is being changed to the following format to use the generic
device state support:
DAHDI/i<span>/<number>[:<subaddress>]-<sequence-number>

Dialplan hints would thus be:
exten => xxx,hint,DAHDI/i2/5551212

This will work with the following restrictions:
*  The number of devices/phones cannot exceed the number of B channels.
(i.e., BRI has 2)
*  Each device/phone can only have one number.  No shared MSN's.
*  The phones/devices probably should not use subaddressing.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@226882 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      6406f395
  63. Oct 05, 2009
    • Kevin P. Fleming's avatar
      Allow non-compliant T.38 endpoints to be supportable via configuration option. · 20743ec0
      Kevin P. Fleming authored 
      Many T.38 endpoints incorrectly send the maximum IFP frame size they can accept
as the T38FaxMaxDatagram value in their SDP, when in fact this value is
supposed to be the maximum UDPTL payload size (datagram size) they can accept.
If the value they supply is small enough (a commonly supplied value is '72'),
T.38 UDPTL transmissions will likely fail completely because the UDPTL packets
will not have enough room for a primary IFP frame and the redundancy used for
error correction. If this occurs, the Asterisk UDPTL stack will emit log messages
warning that data loss may occur, and that the value may need to be overridden.

This patch extends the 't38pt_udptl' configuration option in sip.conf to allow
the administrator to override the value supplied by the remote endpoint and
supply a value that allows T.38 FAX transmissions to be successful with that
endpoint. In addition, in any SIP call where the override takes effect, a debug
message will be printed to that effect. This patch also removes the
T38FaxMaxDatagram configuration option from udptl.conf.sample, since it has not
actually had any effect for a number of releases.

In addition, this patch cleans up the T.38 documentation in sip.conf.sample
(which incorrectly documented that T.38 support was passthrough only).

(issue #15586)
Reported by: globalnetinc


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@222110 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      20743ec0
  65. Oct 01, 2009
  67. Sep 24, 2009