Commit Graph

7 Commits

Author SHA1 Message Date
Richard Mudgett
e0c235bd9b Setup DSP when SS7 call is connected or early media is available.
Outgoing SS7 calls fail to detect incoming DTMF so any bridged channel
that requires out-of-band DTMF will not work.

* Added sig_ss7_open_media() calls at appropriate places in sig_ss7.c.
The new call converts conditionaled out unconverted code and shows that
the code really did something useful.

* Improved some chan_dahdi DTMF debug messages to help track DTMF
handling.

(closes issue ASTERISK-19312)
Reported by: Igor Nikolaev


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@358260 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-03-05 21:33:26 +00:00
Alec L Davis
1b6601bc0a push 'outgoing' flag from sig_XXX up to chan_dahdi
'p->outgoing' in chan_dahdi and sig_analog wern't kept in sync, particulary FXS ast_hangup didn't clear the 'outgoing' flag.
sig_pri and sig_ss7 were keeping 'outgoing' flag insync.

Now provides a callback for all the low level sig_XXX modules.

(issue ASTERISK-19316)

alecdavis (license 585)
Reported by: Jeremy Pepper
Tested by: alecdavis
 
Review: https://reviewboard.asterisk.org/r/1747/


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@355850 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2012-02-18 07:55:11 +00:00
Richard Mudgett
fbc51bb795 Add protection for SS7 channel allocation and better glare handling.
* Added a CLI "ss7 show channels" command that might prove useful for
future debugging.

* Made the incoming SS7 channel event check and gripe message uniform.

* Made sure that the DNID string for an incoming call is always
initialized.

(issue ASTERISK-17966)
Reported by: Kenneth Van Velthoven
Patches:
      jira_asterisk_17966_v1.8_glare.patch (license #5621) patch uploaded by rmudgett


git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@340365 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-10-11 21:03:15 +00:00
Richard Mudgett
8e51d30b67 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/branches/1.8@303771 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2011-01-25 17:49:20 +00:00
Richard Mudgett
fb5fddd987 Extract sig_ss7_init_linkset() to sig_ss7.
Also found a place where sig_pri_init_pri() was inlined and called it
instead.


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@270298 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-06-14 20:51:09 +00:00
Richard Mudgett
717570899c Add missing API function to sig_ss7: sig_ss7_fixup().
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@269308 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-06-09 17:06:41 +00:00
Richard Mudgett
0122ccd29c Extract sig_ss7 out of chan_dahdi.
Extract the SS7 specific code out of chan_dahdi like what was done to
ISDN/PRI and analog signaling.  The new SS7 structures were modeled on
sig_pri.

The changes to sig_pri are an enhancement and a bug fix made possible
because SS7 was extracted.

1) The sig_pri TRANSFERCAPABILITY channel variable should have been set
unconditionally in sig_pri_new_ast_channel().

2) SS7/PRI transfer capability interaction in dahdi_new() fixed because of
SS7 extraction.

3) Module ref count error in dahdi_new() if startpbx failed to start the
PBX for some reason.

Review:	https://reviewboard.asterisk.org/r/661/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@268774 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-06-07 20:04:42 +00:00