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20 authors
  1. Mar 13, 2015
  3. Jun 16, 2014
    • Richard Mudgett's avatar
      chan_dahdi: Adds support for major update to libss7. · 0c896d8b
      Richard Mudgett authored 
      * SS7 support now requires libss7 v2.0 or later.  The new libss7 is not
backwards compatible.

* Added SS7 support for connected line and redirecting.

* Most SS7 CLI commands are reworked as well as new SS7 commands added.
See online CLI help.

* Added several SS7 config option parameters described in
chan_dahdi.conf.sample.

* ISUP timer support reworked and now requires explicit configuration.
See ss7.timers.sample.

Special thanks to Kaloyan Kovachev for his support and persistence in
getting the original patch by adomjan updated and ready for release.

SS7-27 #close
Reported by: adomjan


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@416416 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      0c896d8b
  5. Mar 07, 2014
  7. Oct 02, 2013
  9. Sep 25, 2013
  11. Aug 27, 2013
  13. Nov 08, 2012
  15. Sep 15, 2012
  17. Aug 15, 2012
  19. Aug 09, 2012
  21. Aug 08, 2012
  23. Jul 31, 2012
  25. Jul 20, 2012
  27. Jul 06, 2012
  29. Jun 26, 2012
  31. Jun 22, 2012
  33. Jun 15, 2012
    • Kevin P. Fleming's avatar
      Multiple revisions 369001-369002 · 166b4e2b
      Kevin P. Fleming authored 
      ........
  r369001 | kpfleming | 2012-06-15 10:56:08 -0500 (Fri, 15 Jun 2012) | 11 lines
  
  Add support-level indications to many more source files.
  
  Since we now have tools that scan through the source tree looking for files
  with specific support levels, we need to ensure that every file that is
  a component of a 'core' or 'extended' module (or the main Asterisk binary)
  is explicitly marked with its support level. This patch adds support-level
  indications to many more source files in tree, but avoids adding them to
  third-party libraries that are included in the tree and to source files
  that don't end up involved in Asterisk itself.
........
  r369002 | kpfleming | 2012-06-15 10:57:14 -0500 (Fri, 15 Jun 2012) | 3 lines
  
  Add a script to enable finding source files without support-levels defined.
........

Merged revisions 369001-369002 from http://svn.asterisk.org/svn/asterisk/branches/1.8
........

Merged revisions 369005 from http://svn.asterisk.org/svn/asterisk/branches/10


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@369013 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      166b4e2b
  35. May 30, 2012
  37. Mar 07, 2012
  39. Mar 05, 2012
  41. Mar 01, 2012
  43. Feb 29, 2012
  45. Feb 20, 2012
  47. Feb 18, 2012
  49. Feb 01, 2012
  51. Jan 09, 2012
    • Terry Wilson's avatar
      Replace direct access to channel name with accessor functions · 04da92c3
      Terry Wilson authored 
      There are many benefits to making the ast_channel an opaque handle, from
increasing maintainability to presenting ways to kill masquerades. This patch
kicks things off by taking things a field at a time, renaming the field to
'__do_not_use_${fieldname}' and then writing setters/getters and converting the
existing code to using them. When all fields are done, we can move ast_channel
to a C file from channel.h and lop off the '__do_not_use_'.

This patch sets up main/channel_interal_api.c to be the only file that actually
accesses the ast_channel's fields directly. The intent would be for any API
functions in channel.c to use the accessor functions. No more monkeying around
with channel internals. We should use our own APIs.

The interesting changes in this patch are the addition of
channel_internal_api.c, the moving of the AST_DATA stuff from channel.c to
channel_internal_api.c (note: the AST_DATA stuff will have to be reworked to
use accessor functions when ast_channel is really opaque), and some re-working
of the way channel iterators/callbacks are handled so as to avoid creating fake
ast_channels on the stack to pass in matching data by directly accessing fields
(since "name" is a stringfield and the fake channel doesn't init the
stringfields, you can't use the ast_channel_name_set() function). I went with
ast_channel_name(chan) for a getter, and ast_channel_name_set(chan, name) for a
setter.

The majority of the grunt-work for this change was done by writing a semantic
patch using Coccinelle ( http://coccinelle.lip6.fr/ ).

Review: https://reviewboard.asterisk.org/r/1655/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@350223 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      04da92c3
  53. Oct 11, 2011
  55. Sep 20, 2011
    • Richard Mudgett's avatar
      Merged revisions 337008 via svnmerge from · 7fe331fd
      Richard Mudgett authored 
      https://origsvn.digium.com/svn/asterisk/branches/10

................
  r337008 | rmudgett | 2011-09-20 14:12:24 -0500 (Tue, 20 Sep 2011) | 22 lines
  
  Merged revisions 337007 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.8
  
  ........
    r337007 | rmudgett | 2011-09-20 14:10:30 -0500 (Tue, 20 Sep 2011) | 15 lines
    
    Check if a channel was created before using the pointer in sig_ss7_new_ast_channel().
    
    Fixes the crash in ASTERISK-17955 gdb-11918.txt backtrace.
    
    * Added some missing libss7 access lock protection.
    
    * Prevent cancelling the ss7_linkset() thread at inoportune times just
    like the pri_dchannel() thread.
    
    (issue ASTERISK-17955)
    Reported by: Ian M Sherman
    Patches:
          jira_asterisk_17955_v1.8.patch (license #5621) patch uploaded by rmudgett
          (attached to related ASTERISK-17966)
  ........
................


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@337009 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      7fe331fd
    • Richard Mudgett's avatar
      Merged revisions 336978 via svnmerge from · b3768f04
      Richard Mudgett authored 
      https://origsvn.digium.com/svn/asterisk/branches/10

................
  r336978 | rmudgett | 2011-09-20 13:14:40 -0500 (Tue, 20 Sep 2011) | 28 lines
  
  Merged revisions 336977 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.8
  
  ........
    r336977 | rmudgett | 2011-09-20 13:12:17 -0500 (Tue, 20 Sep 2011) | 21 lines
    
    Fix deadlock from not releasing SS7 linkset lock.
    
    sig_ss7_hangup() failed to release the SS7 linkset lock if the call had
    the alreadyhungup flag set.
    
    * Made unlock the SS7 linkset lock in sig_ss7_hangup() if the
    alreadyhungup flag is set.
    
    * Made ss7_start_call() not hold any locks while creating the channel for
    an incoming call to prevent deadlock.
    
    * Made ss7_grab() a void function, since it could never fail, to simplify
    calling code.
    
    * Made obtain the channel lock to do softhangup in some places.
    
    Patches:
          jira_ast_668_v1.8.patch (license #5621) patch uploaded by rmudgett
    
    JIRA AST-668
  ........
................


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      b3768f04
  57. Sep 09, 2011
    • Matthew Jordan's avatar
      Merged revisions 335078 via svnmerge from · 8b5ba33f
      Matthew Jordan authored 
      https://origsvn.digium.com/svn/asterisk/branches/10

................
  r335078 | mjordan | 2011-09-09 11:27:01 -0500 (Fri, 09 Sep 2011) | 29 lines
  
  Merged revisions 335064 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.8
  
  ........
    r335064 | mjordan | 2011-09-09 11:09:09 -0500 (Fri, 09 Sep 2011) | 23 lines
    
    Updated SIP 484 handling; added Incomplete control frame
    
    When a SIP phone uses the dial application and receives a 484 Address 
    Incomplete response, if overlapped dialing is enabled for SIP, then
    the 484 Address Incomplete is forwarded back to the SIP phone and the
    HANGUPCAUSE channel variable is set to 28.  Previously, the Incomplete
    application dialplan logic was automatically triggered; now, explicit
    dialplan usage of the application is required.
    
    Additionally, this patch adds a new AST_CONTOL_FRAME type called
    AST_CONTROL_INCOMPLETE.  If a channel driver receives this control frame,
    it is an indication that the dialplan expects more digits back from the
    device.  If the device supports overlap dialing it should attempt to 
    notify the device that the dialplan is waiting for more digits; otherwise,
    it can handle the frame in a manner appropriate to the channel driver.
    
    (closes issue ASTERISK-17288)
    Reported by: Mikael Carlsson
    Tested by: Matthew Jordan
    
    Review: https://reviewboard.asterisk.org/r/1416/
  ........
................


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@335079 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      8b5ba33f
  59. May 10, 2011
    • Richard Mudgett's avatar
      Merged revisions 318499 via svnmerge from · d1e27b10
      Richard Mudgett authored 
      https://origsvn.digium.com/svn/asterisk/branches/1.8

........
  r318499 | rmudgett | 2011-05-10 18:41:08 -0500 (Tue, 10 May 2011) | 15 lines
  
  Unable to pickup DAHDI/PRI call because call state is reported as DIALING.
  
  The channel state is not updated to RINGING when an ALERTING message is
  received.  Regression caused when sig_pri.c (also sig_ss7.c) extracted
  from chan_dahdi.c.
  
  * Added missing channel state update to RINGING when the
  AST_CONTROL_RINGING frame is queued for ISDN and SS7.
  
  (closes issue #19257)
  Reported by: alecdavis
  Patches:
        issue19257_v1.8_v2.patch uploaded by rmudgett (license 664)
  Tested by: alecdavis, rmudgett
........


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      d1e27b10
  61. Apr 08, 2011
  63. Jan 25, 2011
    • Richard Mudgett's avatar
      Merged revisions 303771 via svnmerge from · 7889af7c
      Richard Mudgett authored 
      https://origsvn.digium.com/svn/asterisk/branches/1.8

................
  r303771 | rmudgett | 2011-01-25 11:49:20 -0600 (Tue, 25 Jan 2011) | 54 lines
  
  Merged revisions 303769 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
  
  ................
    r303769 | rmudgett | 2011-01-25 11:42:42 -0600 (Tue, 25 Jan 2011) | 47 lines
    
    Merged revisions 303765 via svnmerge from 
    https://origsvn.digium.com/svn/asterisk/branches/1.4
    
    ........
      r303765 | rmudgett | 2011-01-25 11:36:50 -0600 (Tue, 25 Jan 2011) | 40 lines
      
      Sending out unnecessary PROCEEDING messages breaks overlap dialing.
      
      Issue #16789 was a good idea.  Unfortunately, it breaks overlap dialing
      through Asterisk.  There is not enough information available at this point
      to know if dialing is complete.  The ast_exists_extension(),
      ast_matchmore_extension(), and ast_canmatch_extension() calls are not
      adequate to detect a dial through extension pattern of "_9!".
      
      Workaround is to use the dialplan Proceeding() application early in
      non-dial through extensions.
      
      * Effectively revert issue #16789.
      
      * Allow outgoing overlap dialing to hear dialtone and other early media.
      A PROGRESS "inband-information is now available" message is now sent after
      the SETUP_ACKNOWLEDGE message for non-digital calls.  An
      AST_CONTROL_PROGRESS is now generated for incoming SETUP_ACKNOWLEDGE
      messages for non-digital calls.
      
      * Handling of the AST_CONTROL_CONGESTION in chan_dahdi/sig_pri was
      inconsistent with the cause codes.
      
      * Added better protection from sending out of sequence messages by
      combining several flags into a single enum value representing call
      progress level.
      
      * Added diagnostic messages for deferred overlap digits handling corner
      cases.
      
      (closes issue #17085)
      Reported by: shawkris
      
      (closes issue #18509)
      Reported by: wimpy
      Patches:
            issue18509_early_media_v1.8_v3.patch uploaded by rmudgett (license 664)
            Expanded upon issue18509_early_media_v1.8_v3.patch to include analog
            and SS7 because of backporting requirements.
      Tested by: wimpy, rmudgett
    ........
  ................
................


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@303772 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      7889af7c
  65. 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
  67. Jun 14, 2010
  69. Jun 09, 2010