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20 authors
  1. Sep 16, 2010
  3. Sep 15, 2010
    • Terry Wilson's avatar
      Merged revisions 287056 via svnmerge from · a51ce289
      Terry Wilson authored 
      https://origsvn.digium.com/svn/asterisk/branches/1.8

........
  r287056 | twilson | 2010-09-15 17:17:17 -0500 (Wed, 15 Sep 2010) | 10 lines
  
  Don't hang up a call on an SRTP unprotect failure
  
  Also make it more obvious when there is an issue en/decrypting.
  
  (closes issue #17563)
  Reported by: Alexcr
  Patches: 
        res_srtp.c.patch uploaded by sfritsch (license 1089)
  Tested by: twilson
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@287057 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      a51ce289
    • Jeff Peeler's avatar
      Merged revisions 287020 via svnmerge from · eee14db8
      Jeff Peeler authored 
      https://origsvn.digium.com/svn/asterisk/branches/1.8

........
  r287020 | jpeeler | 2010-09-15 15:58:39 -0500 (Wed, 15 Sep 2010) | 1 line
  
  fix uninintialized variable
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@287021 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      eee14db8
    • Richard Mudgett's avatar
      Merged revisions 287017 via svnmerge from · c1af9860
      Richard Mudgett authored 
      https://origsvn.digium.com/svn/asterisk/branches/1.8

................
  r287017 | rmudgett | 2010-09-15 15:53:38 -0500 (Wed, 15 Sep 2010) | 65 lines
  
  Merged revision 287014 from
  https://origsvn.digium.com/svn/asterisk/be/branches/C.3-bier
  
  ..........
    r287014 | rmudgett | 2010-09-15 15:32:24 -0500 (Wed, 15 Sep 2010) | 58 lines
  
    The handling of call transfer signaling for mISDN PTMP is not fully implemented.
  
    The handling of call transfer signaling for mISDN PTMP is not fully
    implemented.  The signaling of number updates with ISDN/DSS1 ECT
    supplementary services (ETS 300 369-1) comes along with a notification
    indicator IE and redirection number IE for PTMP.  The implementation in
    the current Asterisk mISDN channel unfortunately can handle these
    information elements only in a NOTIFY message.  These information elements
    are also signaled in a FACILTY message with a RequestSubaddress facility,
    when the subscriber is already in the active state (see 9.2.4 and 9.2.5 of
    ETS 300 369-1).
  
    **********
  
    abe_2526_ast.patch
  
    * Added support to handle the notification indicator IE and redirection
    number IE with the RequestSubaddress facility.
  
    * Made misdn_update_connected_line() send a NOTIFY message if Asterisk
    originated the call and it is not connected yet.
  
    * Made misdn_update_connected_line() send a FACILITY message if the call
    is already connected.
  
    This patch requires the presence of the associated mISDN patches to
    compile.  I had to enhance mISDN to allow the notification indicator IE
    and the redirection number IE to be used with a FACILITY message.  Earlier
    versions of the Digium enhanced mISDN are no longer going to work.
  
    **********
  
    abe_2526_misdn.patch
  
    * Made an incoming FACILITY message allow the presence of the notification
    indicator IE and the redirection number IE.
  
    **********
  
    abe_2526_misdnuser_v3.patch
  
    * Added support to send and receive a FACILITY message with the
    notification indicator IE and the redirection number IE.
  
    * Added the ability to send a NOTIFY message in PTMP/NT mode to all
    responding subcalls in Q.931 states 6, 7, 8, 9, and 25.
  
    **********
  
    Patches:
  	abe_2526_ast.patch uploaded by rmudgett (license 664)
  	abe_2526_misdn.patch uploaded by rmudgett (license 664)
  	abe_2526_misdnuser_v3.patch uploaded by rmudgett (license 664)
    Tested by: rmudgett and reporter
  
    JIRA SWP-2146
    JIRA ABE-2526
  ..........
................


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      c1af9860
    • Jeff Peeler's avatar
      Merged revisions 287015 via svnmerge from · f129ce3b
      Jeff Peeler authored 
      https://origsvn.digium.com/svn/asterisk/branches/1.8

................
  r287015 | jpeeler | 2010-09-15 15:32:52 -0500 (Wed, 15 Sep 2010) | 21 lines
  
  Merged revisions 286998 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
  
  ................
    r286998 | jpeeler | 2010-09-15 15:28:02 -0500 (Wed, 15 Sep 2010) | 14 lines
    
    Merged revisions 286941 via svnmerge from 
    https://origsvn.digium.com/svn/asterisk/branches/1.4
    
    ........
      r286941 | jpeeler | 2010-09-15 15:08:52 -0500 (Wed, 15 Sep 2010) | 7 lines
      
      Ensure mailbox is not filled to capacity before doing message forwarding.
      
      Specifically, before prompting to record a prepended message the capacity is
      checked first. If the mailbox is full the extension will be reprompted.
      
      ABE-2517
    ........
  ................
................


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      f129ce3b
    • Jeff Peeler's avatar
      Merged revisions 286931 via svnmerge from · 41b95ee8
      Jeff Peeler authored 
      https://origsvn.digium.com/svn/asterisk/branches/1.8

........
  r286931 | jpeeler | 2010-09-15 14:22:15 -0500 (Wed, 15 Sep 2010) | 16 lines
  
  Add parking extension for non-default parking lots.
  
  This is a new feature that allows for parking to custom parking lots to be
  accessed directly, rather than with channel variables or by changing the
  default parking lot. The extension is set with the parkext option just as the
  default parking lot is done. Also, the manager action has been updated to
  optionally allow a specified parking lot.
  
  (closes issue #14882)
  Reported by: vmikhnevych
  Patches: 
        patch_14882.txt uploaded by mnick (license 874)
        modified by me
  
  Review: https://reviewboard.asterisk.org/r/884/
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@286939 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      41b95ee8
    • Richard Mudgett's avatar
      Merged revisions 286904-286905 via svnmerge from · b3fa5ec3
      Richard Mudgett authored 
      https://origsvn.digium.com/svn/asterisk/branches/1.8

........
  r286904 | rmudgett | 2010-09-15 13:28:05 -0500 (Wed, 15 Sep 2010) | 12 lines
  
  Unable to originate calls using E&M over T1.
  
  When originating a call from Unit Under Test to Reference Unit using E&M
  RBS signaling mode, I get the following warning message: "Ring/Off-hook in
  strange state 3 on channel 1".
  
  Fixed the sig_analog outgoing flag.  It was never set when sig_analog was
  extracted from chan_dahdi.
  
  JIRA SWP-2191
  JIRA AST-408
........
  r286905 | rmudgett | 2010-09-15 13:29:21 -0500 (Wed, 15 Sep 2010) | 1 line
  
  Simplify some code in sig_analog.
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@286906 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      b3fa5ec3
    • Matthew Nicholson's avatar
      Merged revisions 286868 via svnmerge from · f9c7f53a
      Matthew Nicholson authored 
      https://origsvn.digium.com/svn/asterisk/branches/1.8

........
  r286868 | mnicholson | 2010-09-15 08:05:52 -0500 (Wed, 15 Sep 2010) | 16 lines
  
  Set tohost to the domain specified in the configuration file instead of the IP address of the host we are calling.
  
  This fixes a regression introduced in r274783.
  
  (closes issue #17960)
  Reported by: adriavidal
  Patches:
        sip-tohost-fix1.diff uploaded by mnicholson (license 96)
  Tested by: mich, mnicholson, adriavidal
  
  (closes issue #17676)
  Reported by: outcast
  Patches:
        sip-tohost-fix1.diff uploaded by mnicholson (license 96)
  Tested by: mnicholson
........


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      f9c7f53a
  5. Sep 14, 2010
  7. Sep 13, 2010
  9. Sep 11, 2010
  11. Sep 10, 2010
  13. Sep 09, 2010
  15. Sep 08, 2010
    • David Vossel's avatar
      Merged revisions 285568 via svnmerge from · 83bc091a
      David Vossel authored 
      https://origsvn.digium.com/svn/asterisk/branches/1.8

................
  r285568 | dvossel | 2010-09-08 17:14:19 -0500 (Wed, 08 Sep 2010) | 16 lines
  
  Merged revisions 285567 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
  
  ................
    r285567 | dvossel | 2010-09-08 17:11:28 -0500 (Wed, 08 Sep 2010) | 9 lines
    
    Merged revisions 285566 via svnmerge from 
    https://origsvn.digium.com/svn/asterisk/branches/1.4
    
    ........
      r285566 | dvossel | 2010-09-08 17:07:31 -0500 (Wed, 08 Sep 2010) | 2 lines
      
      In retrans_pkt, do not unlock pvt until the end of the function on a transmit failure.
    ........
  ................
................


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      83bc091a
    • David Vossel's avatar
      Merged revisions 285564 via svnmerge from · ede9032f
      David Vossel authored 
      https://origsvn.digium.com/svn/asterisk/branches/1.8

................
  r285564 | dvossel | 2010-09-08 16:48:37 -0500 (Wed, 08 Sep 2010) | 60 lines
  
  Merged revisions 285563 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
  
  ........
    r285563 | dvossel | 2010-09-08 16:47:29 -0500 (Wed, 08 Sep 2010) | 54 lines
    
    Fixes interoperability problems with session timer behavior in Asterisk.
    
    CHANGES:
    1. Never put "timer" in "Require" header.  This is not to our benefit
    and RFC 4028 section 7.1 even warns against it.  It is possible for one
    endpoint to perform session-timer refreshes while the other endpoint does
    not support them.  If in this case the end point performing the refreshing
    puts "timer" in the Require field during a refresh, the dialog will
    likely get terminated by the other end.
    
    2. Change the behavior of 'session-timer=accept' in sip.conf (which is
    the default behavior of Asterisk with no session timer configuration
    specified) to only run session-timers as result of an incoming INVITE
    request if the INVITE contains an "Session-Expires" header... Asterisk is
    currently treating having the "timer" option in the "Supported" header as
    a request for session timers by the UAC.  I do not agree with this.  Session
    timers should only be negotiated in "accept" mode when the incoming INVITE
    supplies a "Session-Expires" header, otherwise RFC 4028 says we should
    treat a request containing no "Session-Expires" header as a session with
    no expiration.
    
    Below I have outlined some situations and what Asterisk's behavior is.
    The table reflects the behavior changes implemented by this patch.
    
    SITUATIONS:
    -Asterisk as UAS
    1. Incoming INVITE: NO  "Session-Expires"
    2. Incoming INVITE: HAS "Session-Expires"
    
    -Asterisk as UAC
    3. Outgoing INVITE: NO  "Session-Expires". 200 Ok Response HAS "Session-Expires" header
    4. Outgoing INVITE: NO  "Session-Expires". 200 Ok Response NO  "Session-Expires" header
    5. Outgoing INVITE: HAS "Session-Expires".
    
    Active   - Asterisk will have an active refresh timer regardless if the other endpoint does.
    Inactive - Asterisk does not have an active refresh timer regardless if the other endpoint does.
    XXXXXXX  - Not possible for mode.
    ______________________________________
    |SITUATIONS | 'session-timer' MODES  |
    |___________|________________________|
    |           | originate  |  accept   |
    |-----------|------------|-----------|
    |1.         |   Active   | Inactive  |
    |2.         |   Active   |  Active   |
    |3.         | XXXXXXXX   | Active    |
    |4.         | XXXXXXXX   | Inactive  |
    |5.         |   Active   | XXXXXXXX  |
    --------------------------------------
    
    
    (closes issue #17005)
    Reported by: alexrecarey
  ........
................


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@285565 65c4cc65-6c06-0410-ace0-fbb531ad65f3
      ede9032f
    • Brett Bryant's avatar
      Merged revisions 285533 via svnmerge from · 0c63db04
      Brett Bryant authored 
      https://origsvn.digium.com/svn/asterisk/branches/1.8

................
  r285533 | bbryant | 2010-09-08 16:58:43 -0400 (Wed, 08 Sep 2010) | 15 lines
  
  Merged revisions 285532 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
  
  ........
    r285532 | bbryant | 2010-09-08 16:56:12 -0400 (Wed, 08 Sep 2010) | 8 lines
    
    Fixes a bug with MeetMe where after announcing the amount of time left in a conference, if music on hold was playing, it doesn't restart.
    
    (closes issue #17408)
    Reported by: sysreq
    Patches: 
          asterisk-issue-17408_fixed.patch uploaded by sysreq (license 1009)
    Tested by: sysreq
  ........
................


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      0c63db04