https://origsvn.digium.com/svn/asterisk/branches/1.8

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  r318351 | rmudgett | 2011-05-09 18:15:32 -0500 (Mon, 09 May 2011) | 6 lines
  
  Remove references to res_features and its export file.
  
  The contents of res/res_features.c was moved to into main/features.c
  awhile ago.  There is no longer any need for the res/Makefile to reference
  res_features or the res_features linker exports file to exist.
........


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  r318282 | rmudgett | 2011-05-09 14:07:01 -0500 (Mon, 09 May 2011) | 24 lines
  
  Hangup extension executed twice.
  
  When a user hangs up a call, in certain circumstances, the hangup
  extension can end up being executed twice:
  
  1) If a call is bridged and the 'h' extension executes the Hangup
  application, then the 'h' extension will be executed twice.
  
  2) If a call is bridged within a macro (Dial or Queue), it has its own 'h'
  extension, the main context also has an 'h' extension, and the macro 'h'
  extension executes the Hangup application, then both 'h' extensions will
  be executed.
  
  * Revert originally commited fix for #16106 and just set
  AST_FLAG_BRIDGE_HANGUP_RUN unconditionally in ast_bridge_call().  The
  bridge code just executed an 'h' extension so the main PBX loop does not
  need to execute one as well.
  
  (issue #16106)
  Reported by: ajohnson
  
  (issue #16548)
  Reported by: hajekd
........


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  r318233 | dvossel | 2011-05-09 12:09:55 -0500 (Mon, 09 May 2011) | 14 lines
  
  Merged revisions 318230 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
  
  ........
    r318230 | dvossel | 2011-05-09 11:51:45 -0500 (Mon, 09 May 2011) | 7 lines
    
    Fixes cases where sip_set_rtp_peer can return too early during media path reset.
    
    (closes issue #19225)
    Reported by: one47
    Patches:
          sip_set_rtp_peer.patch uploaded by one47 (license 23)
  ........
................


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................
  r318231 | rmudgett | 2011-05-09 11:57:18 -0500 (Mon, 09 May 2011) | 41 lines
  
  Don't get early media for ISDN on outgoing calls.
  
  It looks to be a long-standing misinterpretation of the progress indicator
  ie values:
  1 - Call is not end-to-end ISDN; further call progress information may be
  available in-band.
  8 - In-band information or an appropriate pattern is now available.
  
  Only value 8 is handled by chan_dahdi/sig_pri.  The 1 value is not handled
  as early media probably because the meaning of the second half of it's
  description was overlooked.
  
  * Test to see if either PRI_PROG_CALL_NOT_E2E_ISDN(1) or
  PRI_PROG_INBAND_AVAILABLE(8) bits are set to open the media path.
  
  (closes issue #18868)
  Reported by: isrl
  Patches:
        issue18868_19246_v1.8.patch uploaded by rmudgett (license 664)
  Tested by: satish_lx
  
  ..........
  
  No inband progress on PRI_EVENT_RINGING even if inband flag set.
  
  My ISDN-PRI provider sends an ALERTING with "Inband information or
  appropriate pattern now available", but Asterisk only generates and passes
  the RING to the SIP extension, not the inband message.  Unfortunately, the
  inband message is not a ringback tone but a prompt that says the number is
  not in service.  The SIP extension then hears two rings and the call is
  hungup which confuses the caller.
  
  * Post an AST_CONTROL_PROGRESS as well as opening the media path if inband
  audio is indicated with an ALERTING message.
  
  (closes issue #19246)
  Reported by: cristiandimache
  Patches:
        issue19246_v1.8.patch uploaded by rmudgett (license 664)
  Tested by: cristiandimache
................


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(closes issue #19238)
Reported by: byronclark
Patches: 
      increase_prepend_filename_length.patch uploaded by byronclark (license 1200)

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  r318148 | jrose | 2011-05-09 09:18:14 -0500 (Mon, 09 May 2011) | 4 lines
  
  Documenting an observed behavior of features in features.conf.  Since parkinglots use an
  integer for the parkinglot extensions, leading zeros specified in the configuration file
  are ignored.
........


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  r318142 | mnicholson | 2011-05-09 09:09:38 -0500 (Mon, 09 May 2011) | 9 lines
  
  Make indicate/control frames WRITE events on framehooks.  Also, if a framehook
  returns a non-control frame, don't forward it to the channel.
  
  (closes issue #19251)
  Reported by: irroot
  Patches:
        (modified) framehook_indicate.patch2 uploaded by irroot (license 52)
  Tested by: irroot
........


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@318143 65c4cc65-6c06-0410-ace0-fbb531ad65f3

When invoking the app parkedcall, the argument can now include '@parkinglot' after the
extension.

(closes issue #18777)
Reported by: cartama
Patches:
      0018777.diff uploaded by cartama (license 1157)

Review: https://reviewboard.asterisk.org/r/1209/



git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@318141 65c4cc65-6c06-0410-ace0-fbb531ad65f3

If a call is made to a line that already has a call and the device is
offhook (ie activeish call), the call is set to CALLWAIT rather than RINGIN. 


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  r318057 | russell | 2011-05-07 18:35:37 -0500 (Sat, 07 May 2011) | 8 lines
  
  res_config_curl: fix a crash with static realtime.
  
  (closes issue #18413)
  Reported by: jmls
  Patches:
        20101202__issue18413.diff.txt uploaded by tilghman (license 14)
  Tested by: jmls
........


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  r318055 | russell | 2011-05-07 18:24:18 -0500 (Sat, 07 May 2011) | 7 lines
  
  chan_iax2: Don't overwrite port found with an SRV lookup.
  
  (closes issue #17291)
  Reported by: jcovert
  Patches:
        chan_iax2.c.1.8.3-srvlookup-corrected.patch uploaded by jcovert (license 551)
........


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(closes issue #17901)
Reported by: salecha


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@318019 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Improve readability of code, eg, (sub->parent == d->activeline) becomes
(sub->line == d->activeline).


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@318018 65c4cc65-6c06-0410-ace0-fbb531ad65f3

Long time coming, finally moving the hookstate from line to device.
This may fix some issues where a device has multiple lines. Previously
we had to run through all lines on a device to see if it was actually
onhook or not.


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  r317969 | russell | 2011-05-06 16:49:01 -0500 (Fri, 06 May 2011) | 10 lines
  
  Use the right variable to print the time in a debug message.
  
  The original patch also increased some buffer sizes, but that was already
  done in this version.
  
  (closes issue #17034)
  Reported by: sysreq
  Patches:
        asterisk-issue-17034.patch uploaded by sysreq (license 1009)
........


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  r317967 | russell | 2011-05-06 16:38:54 -0500 (Fri, 06 May 2011) | 2 lines
  
  Fix some more "set but unused" compiler warnings.
........


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  r317918 | dvossel | 2011-05-06 16:06:55 -0500 (Fri, 06 May 2011) | 7 lines
  
  Fixes missing colon from To/From headers in RTCP manager events.
  
  (closes issue #18221)
  Reported by: clegall_proformatique
  Patches:
        18221_1.patch uploaded by ebroad (license 878)
........


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  r317917 | russell | 2011-05-06 16:06:33 -0500 (Fri, 06 May 2011) | 7 lines
  
  Fix calculation of free RAM to make minmemfree option work.
  
  (closes issue #17124)
  Reported by: loic
  Patches:
        pbx_c.diff uploaded by loic (license 1020)
........


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(closes issue #17036)
Reported by: precisenetworks
Patches:
      import-cdr-csv-mysql.pl uploaded by precisenetworks (license 1010)


git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@317916 65c4cc65-6c06-0410-ace0-fbb531ad65f3

(closes issue #16962)
Reported by: jlpedrosa
Patches:
      patch.diff uploaded by jlpedrosa (license 1002)


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  r317867 | russell | 2011-05-06 15:01:16 -0500 (Fri, 06 May 2011) | 10 lines
  
  chan_sip: Destroy variables on a sip_pvt before copying vars from the sip_peer.
  
  Don't duplicate variables on the sip_pvt.  Just reset the variable list each
  time.
  
  (closes issue #19202)
  Reported by: wdoekes
  Patches:
        issue19202_destroy_challenged_invite_chanvars.patch uploaded by wdoekes (license 717)
........


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  r317865 | russell | 2011-05-06 14:46:49 -0500 (Fri, 06 May 2011) | 11 lines
  
  chan_sip: fix a deadlock in check_rtp_timeout.
  
  Don't block doing silly deadlock avoidance.  Just return and try again later.
  The funciton gets called often enough that it's fine.  Also, this change was
  already made in trunk.
  
  (closes issue #18791)
  Reported by: irroot
  Patches:
        chan_sip.rtptimeout.patch uploaded by irroot (license 52)
........


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  r317861 | russell | 2011-05-06 14:35:00 -0500 (Fri, 06 May 2011) | 11 lines
  
  URI encode less characters in the RPID and Contact headers.
  
  If this change causes any problems, we will need to backport the more extensive
  uri encoding and decoding handling changes that are in trunk/1.10.
  
  (closes issue #18686)
  Reported by: wolfgang
  Patches:
        quick-and-dirty.patch uploaded by wdoekes (license 717)
  Tested by: wdoekes, devellow, wolfgang, mav3rick
........


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  r317858 | mnicholson | 2011-05-06 14:31:50 -0500 (Fri, 06 May 2011) | 6 lines
  
  pbx_lua autoservice fixes
  
  Don't start an autoservice in pbx_lua if pbx_lua already started one and don't
  stop one if we didn't start one.  Also start and stop the autoservice when
  transferring control from and to the pbx.
........

This change is already implemented in trunk.


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  r317837 | russell | 2011-05-06 14:24:11 -0500 (Fri, 06 May 2011) | 11 lines
  
  Fix a crash in the MySQL() application.
  
  This code was not handling channel datastores safely.  The channel
  must be locked.
  
  (closes issue #17964)
  Reported by: wuwu
  Patches:
        issue17964_addon_1.6.2_svn.patch uploaded by seanbright (license 71)
  Tested by: wuwu
........


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  r317805 | russell | 2011-05-06 14:14:39 -0500 (Fri, 06 May 2011) | 7 lines
  
  Add a new sipfriends.sql for MySQL that has more fields in it.
  
  (closes issue #16399)
  Reported by: pabelanger
  Patches:
        sipfriends.sql.v3 uploaded by pabelanger (license 224)
........


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automatically stopped when applications are executed, so this shouldn't cause
any problems.


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Make autoservice_start() and autoservice_stop() return nothing.  Also check if
the autoservice flag is set before starting or stopping the autoservice and
stop and start the autoservice when returning control to and getting control
from the pbx engine.


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This code will actually detect any dialplan jump from any application that
calls ast_explicit_goto().  This change is only being done in trunk as it may
change the way some dialplans execute.


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  r317670 | rmudgett | 2011-05-06 11:19:18 -0500 (Fri, 06 May 2011) | 22 lines
  
  Fix SIP connected line updates.
  
  This patch fixes a couple SIP connected line update problems:
  
  1) The connected line needs to be updated when the initial INVITE is sent
  if there is a peer callerid configured.  Previously, the connected line
  information did not get reported until the call was connected so SIP could
  not report connected line information in ringing or progress messages.
  
  2) The connected line should not be updated on initial connect if there is
  no connected line information.  Previously, all it did was wipe out any
  default preset CONNECTEDLINE information set by the dialplan with empty
  strings.
  
  (closes issue #18367)
  Reported by: GeorgeKonopacki
  Patches:
        issue18367_v1.8.patch uploaded by rmudgett (license 664)
  Tested by: rmudgett
  
  Review: https://reviewboard.asterisk.org/r/1199/
........


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  r317584 | twilson | 2011-05-06 01:18:53 -0700 (Fri, 06 May 2011) | 20 lines
  
  Merged revisions 317575 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.6.2
  
  ................
    r317575 | twilson | 2011-05-06 01:04:17 -0700 (Fri, 06 May 2011) | 13 lines
    
    Merged revisions 317574 via svnmerge from 
    https://origsvn.digium.com/svn/asterisk/branches/1.4
    
    ........
      r317574 | twilson | 2011-05-06 00:55:21 -0700 (Fri, 06 May 2011) | 6 lines
      
      Re-fix queue round-robin
      
      This part of the change for r315596 was incorrect. No bridge occurs
      when doing a roundrobin dial and no one answers, so this code shouldn't
      have been removed.
    ........
  ................
................


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  r317530 | russell | 2011-05-05 18:46:54 -0500 (Thu, 05 May 2011) | 10 lines
  
  If the configure script runs, force a rebuild of menuselect-tree.
  
  Some contents in the menuselect tree are dependent on configure script
  parameters, namely --enable-dev-mode.
  
  (closes issue #17219)
  Reported by: Nick_Lewis
  Patches:
        issue_17219.rev1.txt uploaded by russell (license 2)
........


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  r317486 | russell | 2011-05-05 18:15:53 -0500 (Thu, 05 May 2011) | 9 lines
  
  Fix some more realtime MySQL schema issues.
  
  (closes issue #18537)
  Reported by: denzs
  Patches:
        sipfriends.sql.svndiff uploaded by denzs (license 1182)
        queue_log.sql.svndiff uploaded by denzs (license 1182)
        meetme.sql.svndiff uploaded by denzs (license 1182)
........


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  r317484 | russell | 2011-05-05 18:12:35 -0500 (Thu, 05 May 2011) | 8 lines
  
  Fix some errors in sample MySQL realtime schema files.
  
  (closes issue #18915)
  Reported by: Dovid
  Patches:
        sipfriends.patch uploaded by Dovid (license 652)
        meetme.patch uploaded by Dovid (license 652)
........


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