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20 authors
  1. May 03, 2011
  3. Apr 27, 2011
  5. Apr 26, 2011
    • Terry Wilson's avatar
      Merged revisions 315644 via svnmerge from · 8d2a7187
      Terry Wilson authored 
      https://origsvn.digium.com/svn/asterisk/branches/1.8

................
  r315644 | twilson | 2011-04-26 14:39:01 -0700 (Tue, 26 Apr 2011) | 32 lines
  
  Merged revisions 315643 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
  
  ................
    r315643 | twilson | 2011-04-26 14:27:44 -0700 (Tue, 26 Apr 2011) | 25 lines
    
    Merged revisions 315596 via svnmerge from 
    https://origsvn.digium.com/svn/asterisk/branches/1.4
    
    ........
      r315596 | twilson | 2011-04-26 14:16:10 -0700 (Tue, 26 Apr 2011) | 18 lines
      
      Allow transfer loops without allowing forwarding loops
      
      We try to avoid the situation where two phones may be forwarded to each other
      causing an infinite loop by storing each dialed interface in a channel
      datastore and checking the list before dialing out. This works, but currently
      breaks situations like A calls B, A transfers B to C, B transfers C to A, and A
      transfers C to B. Since human interaction is happening here and not an
      automated forwarding loop, it should be allowed.
      
      This patch removes the dialed_interfaces datastore when a call is bridged (a
      suggestion from the brilliant mmichelson). If a call is being bridged, it
      should be safe to assume that we aren't stuck in a loop.
      
      Since we are now handling this is the bridge code, the previous attempts at
      handling it in app_dial and app_queue are removed.
      
      Review: https://reviewboard.asterisk.org/r/1195/
    ........
  ................
................


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@315670 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      8d2a7187
    • Richard Mudgett's avatar
      Merged revisions 315645 via svnmerge from · 24b69394
      Richard Mudgett authored 
      https://origsvn.digium.com/svn/asterisk/branches/1.8

........
  r315645 | rmudgett | 2011-04-26 17:14:31 -0500 (Tue, 26 Apr 2011) | 21 lines
  
  The 'e' special extension fails to trigger in at least two cases.
  
  The 'e' extension is a fall back for the 'i', 't', or 'T' extensions if
  any of them do not exist.  Many of the places the 'e' extension was
  supposed to be invoked fail because the priority was set wrong.  There
  were two places where the 'e' extension was not even checked for fall
  back.
  
  * Made invoke the 'e' extension similarly to the previous 'i', 't', or 'T'
  extension check and added the 'e' extension as a fall back to the two
  missing locations.
  
  * Prioritized and optimized some hangup tests associated with the 'e'
  extension.
  
  (closes issue #19136)
  Reported by: kshumard
  Tested by: rmudgett
  
  Review: https://reviewboard.asterisk.org/r/1196/
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@315649 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      24b69394
  7. Apr 21, 2011
  9. Apr 20, 2011
  11. Apr 19, 2011
  13. Apr 18, 2011
    • David Vossel's avatar
      Merged revisions 314017 via svnmerge from · 4b454910
      David Vossel authored 
      https://origsvn.digium.com/svn/asterisk/branches/1.8

........
  r314017 | dvossel | 2011-04-18 08:41:06 -0500 (Mon, 18 Apr 2011) | 17 lines
  
  sip codec negotiation of dynamic rtp payloads error fix
  
  This patch fixes how chan_sip handles dynamic rtp payload types
  it does not understand.  At the moment if a dynamic payload's mime
  type does not match one we understand, the payload does not get
  removed from our payload table.  As a result of this, the payload
  is set to whatever dynamic codec we use internally for that payload
  number on outgoing INVITES.  This is incorrect.
  
  This patch fixes this by properly checking the rtpmap set function's
  return code to make sure it was found.  The function can return both
  -1 and -2 depending on the source of the mismatch.  We were just
  checking -1 explicitly.
  
  Review: https://reviewboard.asterisk.org/r/1169/
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@314018 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      4b454910
  15. Apr 15, 2011
  17. Apr 14, 2011
    • Richard Mudgett's avatar
      Add Device State Information CCSS for Generic Devices. · ae2926b5
      Richard Mudgett authored 
      Add Asterisk Device State information and callbacks to the Call Completion
Supplemental Services for generic agents.

There are currently not many devices that have native support for CCSS.
Even as the devices become available there may be other reasons why one
may choose to not take advantage of the native abilities and stick with
the generic implementation.  The generic implementation is quite capable
and could be greatly enhanced by adding device state capabilities.  A
phone could then subscribe to the device state with a BLF key in
conjunction with Asterisk hints.

The advantages of the device state information would allow a single button
to: request CCSS, cancel a CCSS request, and display the current state of
a CCSS request.

For example, you may have a single button that when not lit, there is no
active CCSS request.  When you press that button, the dialplan can query
the DEVICE_STATE() associated with that caller to determine whether they
should be calling CallCompletionRequest() or CallCompletionCancel().  If
there is currently a pending request, then the dialplan would cancel it.
This also has the advantage of showing the true state of a request, which
is an asynchronous call, even when CallCompletionRequest() thinks it was
successful.  The actual request could ultimately fail.  Once lit, further
feedback can be provided to the caller about the current state of their
request since it will be updated by the CCSS State Machine as appropriate.

The DEVICE_STATE mapping is configurable since the BLF being used on a
given phone type may vary.  The idea is to allow some level of
customization as to the phone's behavior.

As an example, you may want the BLF key to go solid once you have
requested a callback.  You may then want the LED to blink (typically
ringing) when either the callback is in process, which is a visual
indication that the incoming call is the desired callback.  You may want
it to blink when the callee is ready but you are busy, giving you a visual
indication that the target is available as you may want to get off the
line so that the callback can be successful.

Device state information is sent back via the ast_devstate_prov_add()
callback for any generic CCSS device as it traverses through the state
machine.  You simply provide a map between CC_STATE values and the
corresponding AST_DEVICE state values.

You could then generate hints against these states similar to what is
possible today with Custom Devstates or MeetMe states.  For example, you
may have an extension 3000 that is currently associated with device
SIP/3000.  You could then create a feature code for that extension that
may look something like:

exten => *823000,hint,ccss:sip/3000

You would then subscribe a BLF button to *823000 which would point to the
dialplan that handled CCSS requests/cancels using the available
DEVICE_STATE() information about ccss:sip/3000 to make the decision about
what to do.

(closes issue #18788)
Reported by: p_lindheimer
Patches:
      ccss.trunk.18788.patch uploaded by p lindheimer (license 558)
      Modified with final reviewboard comments.
Tested by: p_lindheimer, loloski

Review: https://reviewboard.asterisk.org/r/1105/


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@313744 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      ae2926b5
  19. Apr 13, 2011
    • Richard Mudgett's avatar
      Merged revisions 313588 via svnmerge from · c16d39ea
      Richard Mudgett authored 
      https://origsvn.digium.com/svn/asterisk/branches/1.8

................
  r313588 | rmudgett | 2011-04-13 11:31:50 -0500 (Wed, 13 Apr 2011) | 55 lines
  
  Merged revisions 313579 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
  
  ................
    r313579 | rmudgett | 2011-04-13 11:29:49 -0500 (Wed, 13 Apr 2011) | 48 lines
    
    Merged revisions 313545 via svnmerge from 
    https://origsvn.digium.com/svn/asterisk/branches/1.4
    
    ........
      r313545 | rmudgett | 2011-04-13 11:21:24 -0500 (Wed, 13 Apr 2011) | 41 lines
      
      Asterisk does not hangup a channel after endpoint hangs up.
      
      If the call that the dialplan started an AGI script for is hungup while
      the AGI script is in the middle of a command then the AGI script is not
      notified of the hangup.  There are many AGI Exec commands that this can
      happen with.  The reported applications have been: Background, Wait, Read,
      and Dial.  Also the AGI Get Data command.
      
      * Don't wait on the Asterisk channel after it has hung up.  The channel is
      likely to never need servicing again.
      
      * Restored the AGI script's ability to return the AGI_RESULT_HANGUP value
      in run_agi().  It previously only could return AGI_RESULT_SUCCESS or
      AGI_RESULT_FAILURE after the DeadAGI and AGI applications were merged.
      
      (closes issue #17954)
      Reported by: mn3250
      Patches:
            issue17954_v1.8.patch uploaded by rmudgett (license 664)
            issue17954_v1.6.2.patch uploaded by rmudgett (license 664)
            issue17954_v1.4.patch uploaded by rmudgett (license 664)
      Tested by: rmudgett
      JIRA SWP-2171
      
      (closes issue #18492)
      Reported by: devmod
      Tested by: rmudgett
      JIRA SWP-2761
      
      (closes issue #18935)
      Reported by: nvitaly
      Tested by: astmiv, rmudgett
      JIRA SWP-3216
      
      (closes issue #17393)
      Reported by: siby
      Tested by: rmudgett
      JIRA SWP-2727
      
      Review: https://reviewboard.asterisk.org/r/1165/
    ........
  ................
................


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@313606 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      c16d39ea
  21. Apr 11, 2011
  23. Apr 07, 2011
  25. Apr 05, 2011
  27. Apr 01, 2011
    • Richard Mudgett's avatar
      Merged revisions 312461 via svnmerge from · 75594e6e
      Richard Mudgett authored 
      https://origsvn.digium.com/svn/asterisk/branches/1.8

........
  r312461 | rmudgett | 2011-04-01 16:31:39 -0500 (Fri, 01 Apr 2011) | 25 lines
  
  CallCompletionRequest()/CallCompletionCancel() exit non-zero if fail.
  
  The CallCompletionRequest()/CallCompletionCancel() dialplan applications
  exit nonzero on normal failure conditions.  The nonzero exit causes the
  dialplan to hangup immediately.  The dialplan author has no opportunity to
  report success/failure to the user.
  
  * Made always return zero so the dialplan can continue.
  
  * Made set CC_REQUEST_RESULT/CC_REQUEST_REASON and
  CC_CANCEL_RESULT/CC_CANCEL_REASON channel variables respectively.  Also
  documented the values set.
  
  * Reduced the warning about no core instance in CallCompletionCancel() to
  a debug message.  It is a normal event and should not be output at the
  WARNING level.
  
  (closes issue #18763)
  Reported by: p_lindheimer
  Patches:
        ccss.patch uploaded by p lindheimer (license 558) Modified
  Tested by: p_lindheimer, rmudgett
  
  JIRA SWP-3042
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@312462 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      75594e6e
    • Jonathan Rose's avatar
      New Feature for chan_dahdi. 4 length pattern matching. · 846cfa0e
      Jonathan Rose authored 
      In chan_dahdi.conf, the user can now use length 4 patterns in addition to the usual length 2 patterns.  The s
ntax remains the same and the method used to track the pattern history will only change when using the length
 4 patterns.

(closes issue SWP-3250)
Code:
        jrose
        rmudgett


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@312384 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      846cfa0e
    • Tilghman Lesher's avatar
      Merged revisions 312286,312288 via svnmerge from · 3731fd9c
      Tilghman Lesher authored 
      https://origsvn.digium.com/svn/asterisk/branches/1.8

................
  r312286 | tilghman | 2011-04-01 05:44:33 -0500 (Fri, 01 Apr 2011) | 2 lines
  
  Reload must react correctly against a possibly changed table, so dropping the conditional reload flag.
................
  r312288 | tilghman | 2011-04-01 05:58:45 -0500 (Fri, 01 Apr 2011) | 21 lines
  
  Merged revisions 312287 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
  
  ................
    r312287 | tilghman | 2011-04-01 05:51:24 -0500 (Fri, 01 Apr 2011) | 14 lines
    
    Merged revisions 312285 via svnmerge from 
    https://origsvn.digium.com/svn/asterisk/branches/1.4
    
    ........
      r312285 | tilghman | 2011-04-01 05:36:42 -0500 (Fri, 01 Apr 2011) | 7 lines
      
      Found some leaking file descriptors while looking at ast_FD_SETSIZE dead code.
      
      (issue #18969)
       Reported by: oej
       Patches: 
             20110315__issue18969__14.diff.txt uploaded by tilghman (license 14)
    ........
  ................
................


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@312289 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      3731fd9c
  29. Mar 17, 2011
  31. Mar 16, 2011
  33. Mar 15, 2011
  35. Mar 14, 2011
  37. Mar 11, 2011
  39. Mar 10, 2011
  41. Mar 07, 2011
  43. Mar 05, 2011
  45. Mar 02, 2011
  47. Mar 01, 2011
  49. Feb 28, 2011
  51. Feb 24, 2011