[asterisk-commits] junky: branch junky/cli-tls r201684 - in /team/junky/cli-tls: ./ apps/ autoco...

SVN commits to the Asterisk project asterisk-commits at lists.digium.com
Thu Jun 18 12:16:47 CDT 2009


Author: junky
Date: Thu Jun 18 12:16:36 2009
New Revision: 201684

URL: http://svn.asterisk.org/svn-view/asterisk?view=rev&rev=201684
Log:
Merged revisions 199227,199298,199368,199370,199372,199374,199376,199409,199411,199413,199446,199479,199514,199547,199588,199630,199696,199743,199781,199818,199857,199923,199957-199958,200000,200038-200039,200108,200146,200190,200254,200290,200326,200361,200428,200430,200477,200514,200519,200584,200587,200620,200656,200689,200726,200762,200764,200799,200805,200841,200878,200942-200943,200946,200985,201056,201090,201135,201137,201139,201190,201223,201262,201331,201344,201381,201445,201453,201458,201462,201531,201534,201570,201583,201610,201678 via svnmerge from 
https://origsvn.digium.com/svn/asterisk/trunk

................
  r199227 | mmichelson | 2009-06-05 09:51:08 -0400 (Fri, 05 Jun 2009) | 14 lines
  
  Correct "dahdi show channels" output when specifying a group.
  
  Since a DAHDI channel may belong to multiple groups, we need to use
  a bitwise and instead of equivalence to determine whether to display
  the channel information.
  
  
  (closes issue #15248)
  Reported by: gentian
  Patches:
        15248.patch uploaded by mmichelson (license 60)
  Tested by: gentian
................
  r199298 | dvossel | 2009-06-05 17:21:22 -0400 (Fri, 05 Jun 2009) | 21 lines
  
  Merged revisions 199297 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r199297 | dvossel | 2009-06-05 16:19:56 -0500 (Fri, 05 Jun 2009) | 14 lines
    
    Fixes issue with hints giving unexpected results.
    
    Hints with two or more devices that include ONHOLD gave unexpected results.
    
    (closes issue #15057)
    Reported by: p_lindheimer
    Patches:
          onhold_trunk.diff uploaded by dvossel (license 671)
          pbx.c.1.4.patch uploaded by p (license 558)
          devicestate.c.trunk.patch uploaded by p (license 671)
    Tested by: p_lindheimer, dvossel
    
    Review: https://reviewboard.asterisk.org/r/254/
  ........
................
  r199368 | russell | 2009-06-06 17:38:54 -0400 (Sat, 06 Jun 2009) | 2 lines
  
  Switch from "echo -n" to printf.  On my mac, the -n was just getting printed out.
................
  r199370 | russell | 2009-06-06 17:40:56 -0400 (Sat, 06 Jun 2009) | 2 lines
  
  Constify a string and strip trailing whitespace.
................
  r199372 | russell | 2009-06-06 17:42:31 -0400 (Sat, 06 Jun 2009) | 1 line
  
  minor tweak
................
  r199374 | eliel | 2009-06-06 17:56:58 -0400 (Sat, 06 Jun 2009) | 10 lines
  
  Move function SYSINFO documentation to XML.
  
  Move function SYSINFO static documentation to the new AstXML form.
  
  (issue #15245)
  Reported by: eliel
  Patches:
        func_sysinfo_static_conversion.txt uploaded by lmadsen (license 10)
................
  r199376 | eliel | 2009-06-06 18:16:47 -0400 (Sat, 06 Jun 2009) | 12 lines
  
  Move function MINIVMACCOUNT and MINIVMCOUNTER static documentation to XML.
  
  Move function MINIVMACCOUNT and MINIVMCOUNTER statis documentation to the new
  AstXML form.
  
  (issue #15245)
  Reported by: eliel
  Patches:
        app_minivm_static_conversion.txt uploaded by lmadsen (license 10)
        (with minor changes by me)
................
  r199409 | eliel | 2009-06-06 18:27:48 -0400 (Sat, 06 Jun 2009) | 10 lines
  
  Move function MEETME_INFO documentation to XML.
  
  Move function MEETME_INFO static documentation to the new AstXML form.
  
  (issue #15245)
  Reported by: eliel
  Patches:
        app_meetme_static_conversion.txt uploaded by lmadsen (license 10)
................
  r199411 | eliel | 2009-06-06 18:45:42 -0400 (Sat, 06 Jun 2009) | 12 lines
  
  Move function PP_EACH_USER and PP_EACH_EXTENSION documentation to XML.
  
  Move function PP_EACH_USER and PP_EACH_EXTENSION documentation to the new
  AstXML form.
  
  (issue #15245)
  Reported by: eliel
  Patches:
        res_phoneprov_static_conversion.txt uploaded by lmadsen (license 10)
  	(with PP_EACH_USER add by me)
................
  r199413 | eliel | 2009-06-06 19:03:15 -0400 (Sat, 06 Jun 2009) | 11 lines
  
  Move music on hold related applications documentation to XML.
  
  Move MusicOnHold, SetMusicOnHold, StartMusicOnHold, StopMusicOnHold static
  documentation to the new AstXML form.
  
  (issue #15245)
  Reported by: eliel
  Patches:
        res_musiconhold_static_conversion.txt uploaded by lmadsen (license 10)
        (with some fixes and formatting by me)
................
  r199446 | eliel | 2009-06-06 19:28:38 -0400 (Sat, 06 Jun 2009) | 11 lines
  
  Move AGI command 'gosub' static documentation to XML.
  
  Move AGI command 'gosub' statis documentation to the new AstXML form.
  
  (issue #15245)
  Reported by: eliel
  Patches:
        app_stack_static_conversion.txt uploaded by lmadsen (license 10)
        (with minor changes by me)
................
  r199479 | russell | 2009-06-07 10:55:51 -0400 (Sun, 07 Jun 2009) | 2 lines
  
  Global var cleanup - constification and removing unused vars.
................
  r199514 | eliel | 2009-06-07 13:29:44 -0400 (Sun, 07 Jun 2009) | 10 lines
  
  Move application ExternalIVR static documentation to XML.
  
  Move application ExternalIVR static documentation to the new AstXML form.
  
  (issue #15245)
  Reported by: eliel
  Patches:
        app_externalivr.diff uploaded by eliel (license 64)
................
  r199547 | eliel | 2009-06-07 15:15:41 -0400 (Sun, 07 Jun 2009) | 10 lines
  
  Move OSP* applications static documentation to XML.
  
  Move OSP* applications static documentation to the new AstXML form.
  
  (closes issue #15245)
  Reported by: eliel
  Patches:
        app_osplookup_static_conversion.txt uploaded by lmadsen (license 10)
................
  r199588 | mmichelson | 2009-06-08 13:32:04 -0400 (Mon, 08 Jun 2009) | 9 lines
  
  Fix a deadlock that could occur when setting rtp stats on SIP calls.
  
  (closes issue #15143)
  Reported by: cristiandimache
  Patches:
        15143.patch uploaded by mmichelson (license 60)
  Tested by: cristiandimache
................
  r199630 | seanbright | 2009-06-08 15:33:09 -0400 (Mon, 08 Jun 2009) | 32 lines
  
  Merged revisions 199626,199628 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r199626 | seanbright | 2009-06-08 15:24:32 -0400 (Mon, 08 Jun 2009) | 21 lines
    
    Increase the size of our thread stack on 64 bit processors.
    
    We were setting the stack size for each thread to 240KB regardless of
    architecture, which meant that in some scenarios we actually had less available
    stack space on 64 bit processors (pointers use 8 bytes instead of 4).  So now we
    calculate the stack size we reserve based on the platform's __WORDSIZE, which
    gives us:
    
         32 bit -> 240KB
         64 bit -> 496KB
        128 bit -> 1008KB (that's right, we're ready for 128 bit processors)
    
    Patch typed by me but written by several members of #asterisk-dev, including
    Kevin, Tilghman, and Qwell.
    
    (closes issue #14932)
    Reported by: jpiszcz
    Patches:
          06052009_issue14932.patch uploaded by seanbright (license 71)
    Tested by: seanbright
  ........
    r199628 | seanbright | 2009-06-08 15:28:33 -0400 (Mon, 08 Jun 2009) | 2 lines
    
    Fix a typo in the stack size calculation just introduced.
  ........
................
  r199696 | tilghman | 2009-06-08 18:08:44 -0400 (Mon, 08 Jun 2009) | 1 line
  
  Add sigaction janitor
................
  r199743 | dvossel | 2009-06-09 12:22:04 -0400 (Tue, 09 Jun 2009) | 11 lines
  
  module load priority
  
  This patch adds the option to give a module a load priority. The value represents the order in which a module's load() function is initialized.  The lower the value, the higher the priority.  The value is only checked if the AST_MODFLAG_LOAD_ORDER flag is set.  If the AST_MODFLAG_LOAD_ORDER flag is not set, the value will never be read and the module will be given the lowest possible priority
  on load.  Since some modules are reliant on a timing interface, the timing modules have been given a high load priorty.
  
  (closes issue #15191)
  Reported by: alecdavis
  Tested by: dvossel
  
  Review: https://reviewboard.asterisk.org/r/262/
................
  r199781 | seanbright | 2009-06-09 14:08:53 -0400 (Tue, 09 Jun 2009) | 2 lines
  
  Fix all of the parallel build warnings issued when running make -j#.
................
  r199818 | dvossel | 2009-06-09 16:47:57 -0400 (Tue, 09 Jun 2009) | 11 lines
  
  CLI NOTIFY sending wrong transport type.
  
  SIP's cli NOTIFY command only used UDP rather than copying the transport type from the peer.
  
  (closes issue #15283)
  Reported by: jthurman
  Patches:
        sip-notify-tcp-svn199728.patch uploaded by jthurman (license 614)
  Tested by: jthurman, dvossel
................
  r199857 | seanbright | 2009-06-10 12:10:23 -0400 (Wed, 10 Jun 2009) | 9 lines
  
  Merged revisions 199856 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r199856 | seanbright | 2009-06-10 12:08:35 -0400 (Wed, 10 Jun 2009) | 2 lines
    
    __WORDSIZE is not available on all platforms, so use sizeof(void *) instead.
  ........
................
  r199923 | mmichelson | 2009-06-10 14:58:12 -0400 (Wed, 10 Jun 2009) | 8 lines
  
  Use ast_channel_unref to instead of ast_free on a newly created channel.
  
  Also I removed an unnecessary free of a cid_name. This will be freed properly
  in the channel destructor.
  
  Reported by mnicholson in #asterisk-dev.
................
  r199957 | dbrooks | 2009-06-10 16:00:45 -0400 (Wed, 10 Jun 2009) | 10 lines
  
  Fixes the argument order in definition of new_find_extension().
  
  In the definition of new_find_extension(), the arguments 'callerid' and
  'label' were swapped. The prototype declaration and all calls to the
  function are ordered 'callerid' then 'label', but the function itself
  was ordered 'label' then 'callerid'.
  
  (closes issue #15303)
  Reported by: JimDickenson
................
  r199958 | mmichelson | 2009-06-10 16:15:48 -0400 (Wed, 10 Jun 2009) | 6 lines
  
  Only try to use the invite_branch on outgoing INVITEs with auth credentials.
  
  I have added a comment to the code to help ease understanding of the logic here
  as well.
................
  r200000 | seanbright | 2009-06-10 16:40:41 -0400 (Wed, 10 Jun 2009) | 1 line
  
  Remove some trailing whitespace and steal revision 200000.
................
  r200038 | lmadsen | 2009-06-11 08:13:49 -0400 (Thu, 11 Jun 2009) | 14 lines
  
  Blocked revisions 200037 via svnmerge
  
  ........
    r200037 | lmadsen | 2009-06-11 08:12:06 -0400 (Thu, 11 Jun 2009) | 8 lines
    
    Fix path for .flavor and .version.
    
    (issue #14737)
    Reported by: davidw
    Patches:
          flavor.patch uploaded by davidw (license 780)
    Tested by: davidw
  ........
................
  r200039 | lmadsen | 2009-06-11 08:15:09 -0400 (Thu, 11 Jun 2009) | 8 lines
  
  Fix path for .flavor and .version
  
  (issue #14737)
  Reported by: davidw
  Patches:
        flavor.patch uploaded by davidw (license 780)
  Tested by: davidw
................
  r200108 | eliel | 2009-06-11 11:40:03 -0400 (Thu, 11 Jun 2009) | 9 lines
  
  Release the allocated channel decreasing the reference counter.
  
  When allocating the channel use ao2_ref(-1) to release it, instead of calling
  ast_free().
  Also avoid freeing structures inside that channel (on error) if they will be
  released by the channel destructor being called if the reference counter reachs
  0.
................
  r200146 | mmichelson | 2009-06-11 17:17:14 -0400 (Thu, 11 Jun 2009) | 5 lines
  
  Fix a crash due to a potentially NULL p->options.
  
  Thanks to mnicholson for pointing it out.
................
  r200190 | seanbright | 2009-06-11 18:21:32 -0400 (Thu, 11 Jun 2009) | 8 lines
  
  Blocked revisions 200185 via svnmerge
  
  ........
    r200185 | seanbright | 2009-06-11 18:20:31 -0400 (Thu, 11 Jun 2009) | 2 lines
    
    Backport fix for parallel build warnings from trunk r199781.
  ........
................
  r200254 | seanbright | 2009-06-11 22:20:19 -0400 (Thu, 11 Jun 2009) | 5 lines
  
  Call chgrp instead of chown when setting run directory group ownership.
  
  (issue #13153)
  Reported by: pabelanger
................
  r200290 | mmichelson | 2009-06-12 10:55:07 -0400 (Fri, 12 Jun 2009) | 3 lines
  
  Fix a potential crash from trying to access a NULL channel pointer.
................
  r200326 | mmichelson | 2009-06-12 11:37:30 -0400 (Fri, 12 Jun 2009) | 4 lines
  
  Fix some bad locking stemming from trying to forward a call to a non-existent
  extension from a queue.
................
  r200361 | mmichelson | 2009-06-12 15:07:51 -0400 (Fri, 12 Jun 2009) | 16 lines
  
  Merged revisions 200360 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r200360 | mmichelson | 2009-06-12 14:06:41 -0500 (Fri, 12 Jun 2009) | 10 lines
    
    Suppress a warning message and give a better return code when generating
    inband ringing after a call is answered.
    
    (closes issue #15158)
    Reported by: madkins
    Patches:
          15158.patch uploaded by mmichelson (license 60)
    Tested by: madkins
  ........
................
  r200428 | seanbright | 2009-06-12 15:42:26 -0400 (Fri, 12 Jun 2009) | 6 lines
  
  First shot at an upstart script for asterisk on Ubuntu.
  
  This works relatively well (assuming you are using /var/run/asterisk) as your
  run directory and upstart 0.3.9.  Needs to be generalized and eventually added
  to the 'make install' target for Ubuntu.
................
  r200430 | seanbright | 2009-06-12 15:46:25 -0400 (Fri, 12 Jun 2009) | 1 line
  
  Include basic installation and usage instructions for upstart script.
................
  r200477 | moy | 2009-06-14 02:13:48 -0400 (Sun, 14 Jun 2009) | 3 lines
  
  added openr2 to menuselect-deps.in, recent commit in menuselect made me realize this was never done but was working anyways
  also added support for skip category request feature of openr2 and updated chan_dahdi.conf.sample
................
  r200514 | mmichelson | 2009-06-15 11:22:11 -0400 (Mon, 15 Jun 2009) | 11 lines
  
  Merged revisions 200513 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r200513 | mmichelson | 2009-06-15 10:21:46 -0500 (Mon, 15 Jun 2009) | 5 lines
    
    Add INFO to our allowed methods so that endpoints know they may send it to us.
    
    AST-223
  ........
................
  r200519 | kpfleming | 2009-06-15 12:07:23 -0400 (Mon, 15 Jun 2009) | 11 lines
  
  Redesigned 'optional API' support.
  
  This patch provides a new implementation of the optional API support defined
  in asterisk/optional_api.h; this new version provides solves compatibility
  issues with the use of linker version scripts for suppressing global symbols.
  In addition, there is now a functional (and tested!) implementation for Mac OS/X,
  so module writers no longer need to use special tests before calling optional
  API functions. All future implementations must provide these same semantics,
  so that module writers can rely on them.
................
  r200584 | kpfleming | 2009-06-15 12:38:32 -0400 (Mon, 15 Jun 2009) | 9 lines
  
  Some minor structure size improvements in sip_pvt and sip_peer.
  
  Using the 'pahole' tool, it is now quite easy to see where structure fields
  could be organized differently to keep the compiler from having to add
  padding to satisfy alignment requirements. These changes reduced the sizes of
  sip_pvt and sip_peer by a few bytes each (on 64-bit platforms), and also fixed
  a spelling error in a field name.
................
  r200587 | kpfleming | 2009-06-15 13:06:34 -0400 (Mon, 15 Jun 2009) | 6 lines
  
  Convert a number of global module variables to 'static'.
  
  These modules all contained variables that are module-global but not system-global,
  but were not marked 'static'.
................
  r200620 | kpfleming | 2009-06-15 13:34:30 -0400 (Mon, 15 Jun 2009) | 5 lines
  
  More 'static' qualifiers on module global variables.
  
  The 'pglobal' tool is quite handy indeed :-)
................
  r200656 | kpfleming | 2009-06-15 15:10:10 -0400 (Mon, 15 Jun 2009) | 8 lines
  
  Last batch of 'static' qualifiers for module-level global variables.
  
  Fix up modules in the 'apps' directory, and also correct the bad example of
  enum definitions in include/asterisk/app.h, which many developers followed
  (thanks for reading the documentation!). In addition, add some basic usage
  examples of the 'pahole' and 'pglobal' tools to the coding guidelines.
................
  r200689 | kpfleming | 2009-06-15 16:42:38 -0400 (Mon, 15 Jun 2009) | 12 lines
  
  Accept T.38 re-INVITE responses with invalid SDP versions.
  
  This commit changes the 'incoming SDP version' check logic a bit more; when
  'ignoresdpversion' is *not* set for a peer, if we initiate a re-INVITE to
  switch to T.38, we'll always accept the peer's SDP response, even if they
  don't properly increment the SDP version number as they should. If this situation
  occurs, a warning message will be generated suggesting that the peer's
  configuration be changed to include the 'ignoresdpversion' configuration option
  (although ideally they'd fix their SIP implementation to be RFC compliant).
  
  AST-221
................
  r200726 | kpfleming | 2009-06-15 21:03:22 -0400 (Mon, 15 Jun 2009) | 6 lines
  
  Document the new automatic 'ignoresdpversion' behavior.
  
  Asterisk will now automatically ignore incorrect incoming SDP version numbers
  when necessary to complete a T.38 re-INVITE operation.
................
  r200762 | russell | 2009-06-15 21:26:20 -0400 (Mon, 15 Jun 2009) | 2 lines
  
  Add missing closure of verbatim environment.
................
  r200764 | kpfleming | 2009-06-15 21:28:08 -0400 (Mon, 15 Jun 2009) | 11 lines
  
  Ensure that configure-script testing for compiler attributes actually works.
  
  The configure script tests for compiler attributes didn't actually enable
  enough warnings or provide a proper test harness to determine whether the 
  compiler supports the attribute in question or not; this caused gcc 4.1 to
  report that it supports 'weakref', but it doesn't actually support it in the
  way that is needed for our optional API mechanism. The new configure script
  test will properly distinguish between full support and partial support
  for this attribute, among others.
................
  r200799 | moy | 2009-06-15 22:24:30 -0400 (Mon, 15 Jun 2009) | 2 lines
  
  keep backwards compatible chan_dahdi with older openr2 versions by not using the new skip category feature unless supported
................
  r200805 | russell | 2009-06-15 22:32:33 -0400 (Mon, 15 Jun 2009) | 6 lines
  
  Don't claim a char * is a mansession object.
  
  Since there was only 1 bucket, and no hash function was specified, the code
  actually worked perfectly fine.  However, in theory, this was invalid use of
  the OBJ_POINTER flag, so remove it so the code provides a better usage example.
................
  r200841 | eliel | 2009-06-16 08:32:00 -0400 (Tue, 16 Jun 2009) | 6 lines
  
  Show the interface name on error, if it is not found.
  
  If the smdiport specified is not found, show the interface name
  instead of '(null)'.
................
  r200878 | eliel | 2009-06-16 10:12:34 -0400 (Tue, 16 Jun 2009) | 12 lines
  
  Recorded merge of revisions 200875 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r200875 | eliel | 2009-06-16 09:25:51 -0400 (Tue, 16 Jun 2009) | 5 lines
    
    Show the interface name on error, if it is not found.
    
    If the smdiport specified is not found, show the interface name
    instead of '(null)'.
  ........
................
  r200942 | russell | 2009-06-16 11:26:57 -0400 (Tue, 16 Jun 2009) | 2 lines
  
  Add Sean Bright to CREDITS - Thanks, Sean!
................
  r200943 | mvanbaak | 2009-06-16 11:51:36 -0400 (Tue, 16 Jun 2009) | 9 lines
  
  add FILE_STORAGE to Voicemail Build Options
  
  Voicemail can only use one storage module at the moment.
  Because it's unclear that selecting one of the storage modules
  in menuselect will disable filesystem storage we now have
  a FILE_STORAGE option that conflicts with the other modules.
  
  (closes issue #15333)
................
  r200946 | dvossel | 2009-06-16 12:03:30 -0400 (Tue, 16 Jun 2009) | 32 lines
  
  SIP transport type issues
  
  What this patch addresses:
  1. ast_sip_ouraddrfor() by default binds to the UDP address/port
  reguardless if the sip->pvt is of type UDP or not.  Now when no
  remapping is required, ast_sip_ouraddrfor() checks the sip_pvt's
  transport type, attempting to set the address and port to the
  correct TCP/TLS bindings if necessary.
  2.  It is not necessary to send the port number in the Contact
  header unless the port is non-standard for the transport type.
  This patch fixes this and removes the todo note.
  3.  In sip_alloc(), the default dialog built always uses transport
  type UDP.  Now sip_alloc() looks at the sip_request (if present)
  and determines what transport type to use by default.
  4.  When changing the transport type of a sip_socket, the file
  descriptor must be set to -1 and in some cases the tcptls_session's
  ref count must be decremented and set to NULL.  I've encountered
  several issues associated with this process and have created a function,
  set_socket_transport(), to handle the setting of the socket type.
  
  
  (closes issue #13865)
  Reported by: st
  Patches:
        dont_add_port_if_tls.patch uploaded by Kristijan (license 753)
        13865.patch uploaded by mmichelson (license 60)
        tls_port_v5.patch uploaded by vrban (license 756)
        transport_issues.diff uploaded by dvossel (license 671)
  Tested by: mmichelson, Kristijan, vrban, jmacz, dvossel
  
  Review: https://reviewboard.asterisk.org/r/278/
................
  r200985 | kpfleming | 2009-06-16 12:32:36 -0400 (Tue, 16 Jun 2009) | 7 lines
  
  Fix problems with new compiler attribute checking in configure script.
  
  The last changes to ast_gcc_attribute.m4 caused some problems checking for
  various attributes, because the scope of the symbol the attribute is applied
  to can be important; this patch allows the scope to be specified for the check.
................
  r201056 | kpfleming | 2009-06-16 14:54:30 -0400 (Tue, 16 Jun 2009) | 18 lines
  
  Merged revisions 200991 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r200991 | kpfleming | 2009-06-16 12:05:38 -0500 (Tue, 16 Jun 2009) | 11 lines
    
    Improve support for media paths that can generate multiple frames at once.
    
    There are various media paths in Asterisk (codec translators and UDPTL, primarily)
    that can generate more than one frame to be generated when the application calling
    them expects only a single frame. This patch addresses a number of those cases,
    at least the primary ones to solve the known problems. In addition it removes the
    broken TRACE_FRAMES support, fixes a number of bugs in various frame-related API
    functions, and cleans up various code paths affected by these changes.
    
    https://reviewboard.asterisk.org/r/175/
  ........
................
  r201090 | kpfleming | 2009-06-16 15:27:12 -0400 (Tue, 16 Jun 2009) | 5 lines
  
  Another minor fix to compiler attribute checking.
  
  Defaulting to 'static' for the function scope was bad... so remove it.
................
  r201135 | kpfleming | 2009-06-16 16:50:41 -0400 (Tue, 16 Jun 2009) | 7 lines
  
  When compiling in an Emacs-spawned shell, always print directory names.
  
  This change ensures that Emacs can find the proper source files when parsing
  compiler error messages, since it uses the 'make' output including directory
  names to do it.
................
  r201137 | kpfleming | 2009-06-16 17:02:05 -0400 (Tue, 16 Jun 2009) | 6 lines
  
  Explicitly test for 'static weakref' support.
  
  Since we use 'static' weakref symbols, and not all GCC versions support them,
  test for that combination explicitly.
................
  r201139 | kpfleming | 2009-06-16 17:10:15 -0400 (Tue, 16 Jun 2009) | 10 lines
  
  Enable applications to enable/disable digit and tone detection.
  
  Some applications (notably app_fax) do not need digit detection nor FAX tone
  detection while they are running, and if Asterisk is using software DSPs to provide
  the detection, this consumes extra CPU cycles that could be better spent on the
  actual application. This patch allows applications to query and control the state
  of digit and tone detection on a channel, and modifies app_fax to disable them
  while the FAX operations are occurring (and re-enable digit detection afterwards).
................
  r201190 | seanbright | 2009-06-16 18:11:07 -0400 (Tue, 16 Jun 2009) | 2 lines
  
  Update my e-mail address (thanks for the props, russell :))
................
  r201223 | dvossel | 2009-06-16 18:29:30 -0400 (Tue, 16 Jun 2009) | 2 lines
  
  fix issue with build_contact introduced by the "SIP trasnport type issues" commit
................
  r201262 | kpfleming | 2009-06-17 08:04:17 -0400 (Wed, 17 Jun 2009) | 15 lines
  
  Merged revisions 201261 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r201261 | kpfleming | 2009-06-17 07:03:25 -0500 (Wed, 17 Jun 2009) | 9 lines
    
    Correct AST_LIST_APPEND_LIST behavior when list to be appended is empty.
    
    When the list to be appended is empty, and the list to be appended to is *not*,
    AST_LIST_APPEND_LIST would actually cause the target list to become broken,
    and no longer have a pointer to its last entry. This patch fixes the problem.
    
    (reported by Stanislaw Pitucha on the asterisk-dev mailing list)
  ........
................
  r201331 | dvossel | 2009-06-17 10:42:06 -0400 (Wed, 17 Jun 2009) | 7 lines
  
  update chan_iax to use 64bit feature flags.
  
  (closes issue #15335)
  Reported by: lmadsen
  
  Review: https://reviewboard.asterisk.org/r/284/
................
  r201344 | dvossel | 2009-06-17 11:20:26 -0400 (Wed, 17 Jun 2009) | 16 lines
  
  SIP registry ref count error
  
  During a sip reload, the list of sip_registry objects are
  supposed to be traversed, unlinked, and destroyed, but
  destruction never takes place due to a ref counting error.
  This causes a memory leak when registry items are removed
  from sip.conf and reloaded.  While the registries are removed
  from the global list, they are not removed from the scheduler.
  Because of this, SIP register attempts continue to be sent
  out for the item even though it may no longer be in the .conf.
  
  (closes issue #15295)
  Reported by: amorsen
  
  Review: https://reviewboard.asterisk.org/r/282/
................
  r201381 | dbrooks | 2009-06-17 15:15:07 -0400 (Wed, 17 Jun 2009) | 16 lines
  
  Merged revisions 201380 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r201380 | dbrooks | 2009-06-17 13:45:50 -0500 (Wed, 17 Jun 2009) | 9 lines
    
    Checks for NULL sip_pvt pointer in chan_sip.c->acf_channel_read()
    
    Zombie channels could be passed, and chan_sip.c wasn't checking for it.
    Could crash Asterisk. Now checking for NULL pointer.
    
    (closes issue #15330)
    Reported by: okrief
    Tested by: dbrooks
  ........
................
  r201445 | dvossel | 2009-06-17 15:45:35 -0400 (Wed, 17 Jun 2009) | 25 lines
  
  Merged revisions 201423 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r201423 | dvossel | 2009-06-17 14:28:12 -0500 (Wed, 17 Jun 2009) | 19 lines
    
    StopMixMonitor race condition (not giving up file immediately)
    
    StopMixMonitor only indicates to the MixMonitor thread to stop
    writing to the file.  It does not guarantee that the recording's
    file handle is available to the dialplan immediately after execution.
    This results in a race condition.  To resolve this, the filestream
    pointer is placed in a datastore on the channel. When StopMixMonitor
    is called, the datastore is retrieved from the channel and the
    filestream is closed immediately before returning to the dialplan.
    Documentation indicating the use of StopMixMonitor to free files
    has been updated as well.
    
    (closes issue #15259)
    Reported by: travisghansen
    Tested by: dvossel
    
    Review: https://reviewboard.asterisk.org/r/283/
  ........
................
  r201453 | dvossel | 2009-06-17 16:00:51 -0400 (Wed, 17 Jun 2009) | 3 lines
  
  ast_channel_datastore_alloc is no longer used. updating datastores.txt to reflect that.
................
  r201458 | mmichelson | 2009-06-17 16:04:12 -0400 (Wed, 17 Jun 2009) | 15 lines
  
  Merged revisions 201450 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r201450 | mmichelson | 2009-06-17 14:59:31 -0500 (Wed, 17 Jun 2009) | 9 lines
    
    Change the datastore traversal in ast_do_masquerade to use a safe list traversal.
    
    It is possible for datastore fixup functions to remove the datastore from the list
    and free it. In particular, the queue_transfer_fixup in app_queue does this. While
    I don't yet know of this causing any crashes, it certainly could.
    
    Found while discussing a separate issue with Brian Degenhardt.
  ........
................
  r201462 | mmichelson | 2009-06-17 16:10:01 -0400 (Wed, 17 Jun 2009) | 12 lines
  
  Fix problem with no audio due to ignoring the SDP.
  
  A recent change to our SDP version comparison made audio not function
  on some calls. This was because of a test wherein we were trying to
  see if an unsigned value was less than 0. This is a dumb comparison
  and arguably the compiler should have warned about it. Alas, though,
  it slipped past. Now it's fixed by changing the variable to be a
  signed type.
  
  Found by several developers. Tested by mnicholson and dbrooks.
................
  r201531 | tilghman | 2009-06-17 17:31:39 -0400 (Wed, 17 Jun 2009) | 7 lines
  
  Initialize additional variables, to prevent a possible crash.
  (closes issue #15186)
   Reported by: ajohnson
   Patches: 
         20090528__issue15186.diff.txt uploaded by tilghman (license 14)
   Tested by: ajohnson
................
  r201534 | dvossel | 2009-06-17 17:56:42 -0400 (Wed, 17 Jun 2009) | 11 lines
  
  Add rtsavesysname to chan_iax
  
  chan_sip has an option to save the sysname on rtupdate.  This patch copies that same logic to chan_iax.
  
  (closes issue #14837)
  Reported by: barthpbx
  Patches:
        iax2-rtsavesysname.patch uploaded by barthpbx (license 744)
        rt_iax.diff uploaded by dvossel (license 671)
................
  r201570 | dvossel | 2009-06-18 11:16:05 -0400 (Thu, 18 Jun 2009) | 11 lines
  
  parsing extension correctly from sip register lines
  
  If a transport type was specified, but no extension, parsing of the extension would return whatever was after the transport rather than defaulting to 's'.
  
  (closes issue #15111)
  Reported by: ffs
  Patches:
        chan_sip.c_register-parser.patch uploaded by ffs (license 730)
  Tested by: ffs, dvossel
................
  r201583 | mmichelson | 2009-06-18 11:20:17 -0400 (Thu, 18 Jun 2009) | 9 lines
  
  Trunk implementation of setting an alternate RTP source.
  
  This contains the interface by which we can let an rtp instance know
  that it might start receiving audio from a new source. This is similar
  in nature to revision 197588 of Asterisk 1.4.
  
  Review: https://reviewboard.asterisk.org/r/276
................
  r201610 | russell | 2009-06-18 11:27:10 -0400 (Thu, 18 Jun 2009) | 36 lines
  
  Merged revisions 201600 via svnmerge from 
  https://origsvn.digium.com/svn/asterisk/branches/1.4
  
  ........
    r201600 | russell | 2009-06-18 10:24:31 -0500 (Thu, 18 Jun 2009) | 29 lines
    
    Fix memory corruption and leakage related reloads of non files mode MoH classes.
    
    For Music on Hold classes that are not files mode, meaning that we are executing
    an application that will feed us audio data, we use a thread to monitor the
    external application and read audio from it.  This thread also makes use of the
    MoH class object.  In the MoH class destructor, we used pthread_cancel() to ask
    the thread to exit.  Unfortunately, the code did not wait to ensure that the
    thread actually went away.  What needed to be done is a pthread_join() to ensure
    that the thread fully cleans up before we proceed.  By adding this one line, we
    resolve two significant problems:
    
      1) Since the thread was never joined, it never fully goes away.  So, on every
         reload of non-files mode MoH, an unused thread was sticking around.
    
      2) There was a race condition here where the application monitoring thread
         could still try to access the MoH class, even though the thread executing
         the MoH reload has already destroyed it.
    
    (issue #15109)
    Reported by: jvandal
    
    (issue #15123)
    Reported by: axisinternet
    
    (issue #15195)
    Reported by: amorsen
    
    (issue AST-208)
  ........
................
  r201678 | dvossel | 2009-06-18 12:37:42 -0400 (Thu, 18 Jun 2009) | 11 lines
  
  fixes some memory leaks and redundant conditions
  
  (closes issue #15269)
  Reported by: contactmayankjain
  Patches:
        patch.txt uploaded by contactmayankjain (license 740)
        memory_leak_stuff.trunk.diff uploaded by dvossel (license 671)
  Tested by: contactmayankjain, dvossel
................

Added:
    team/junky/cli-tls/contrib/upstart/
      - copied from r201678, trunk/contrib/upstart/
    team/junky/cli-tls/contrib/upstart/asterisk.upstart-0.3.9
      - copied unchanged from r201678, trunk/contrib/upstart/asterisk.upstart-0.3.9
Modified:
    team/junky/cli-tls/   (props changed)
    team/junky/cli-tls/CHANGES
    team/junky/cli-tls/CREDITS
    team/junky/cli-tls/Makefile
    team/junky/cli-tls/apps/app_adsiprog.c
    team/junky/cli-tls/apps/app_alarmreceiver.c
    team/junky/cli-tls/apps/app_amd.c
    team/junky/cli-tls/apps/app_authenticate.c
    team/junky/cli-tls/apps/app_chanisavail.c
    team/junky/cli-tls/apps/app_channelredirect.c
    team/junky/cli-tls/apps/app_chanspy.c
    team/junky/cli-tls/apps/app_confbridge.c
    team/junky/cli-tls/apps/app_controlplayback.c
    team/junky/cli-tls/apps/app_dahdibarge.c
    team/junky/cli-tls/apps/app_dahdiras.c
    team/junky/cli-tls/apps/app_db.c
    team/junky/cli-tls/apps/app_dial.c
    team/junky/cli-tls/apps/app_dictate.c
    team/junky/cli-tls/apps/app_directed_pickup.c
    team/junky/cli-tls/apps/app_directory.c
    team/junky/cli-tls/apps/app_disa.c
    team/junky/cli-tls/apps/app_dumpchan.c
    team/junky/cli-tls/apps/app_echo.c
    team/junky/cli-tls/apps/app_exec.c
    team/junky/cli-tls/apps/app_externalivr.c
    team/junky/cli-tls/apps/app_fax.c
    team/junky/cli-tls/apps/app_jack.c
    team/junky/cli-tls/apps/app_macro.c
    team/junky/cli-tls/apps/app_meetme.c
    team/junky/cli-tls/apps/app_minivm.c
    team/junky/cli-tls/apps/app_mixmonitor.c
    team/junky/cli-tls/apps/app_osplookup.c
    team/junky/cli-tls/apps/app_page.c
    team/junky/cli-tls/apps/app_queue.c
    team/junky/cli-tls/apps/app_read.c
    team/junky/cli-tls/apps/app_readexten.c
    team/junky/cli-tls/apps/app_rpt.c
    team/junky/cli-tls/apps/app_skel.c
    team/junky/cli-tls/apps/app_sms.c
    team/junky/cli-tls/apps/app_stack.c
    team/junky/cli-tls/apps/app_url.c
    team/junky/cli-tls/apps/app_voicemail.c
    team/junky/cli-tls/autoconf/ast_gcc_attribute.m4
    team/junky/cli-tls/build_tools/make_version_c
    team/junky/cli-tls/build_tools/make_version_h
    team/junky/cli-tls/build_tools/menuselect-deps.in
    team/junky/cli-tls/cdr/cdr_manager.c
    team/junky/cli-tls/channels/chan_agent.c
    team/junky/cli-tls/channels/chan_dahdi.c
    team/junky/cli-tls/channels/chan_h323.c
    team/junky/cli-tls/channels/chan_iax2.c
    team/junky/cli-tls/channels/chan_local.c
    team/junky/cli-tls/channels/chan_misdn.c
    team/junky/cli-tls/channels/chan_sip.c
    team/junky/cli-tls/channels/chan_skinny.c
    team/junky/cli-tls/channels/h323/ast_h323.cxx
    team/junky/cli-tls/channels/misdn/isdn_lib.c
    team/junky/cli-tls/channels/xpmr/xpmr.c
    team/junky/cli-tls/codecs/gsm/src/gsm_destroy.c
    team/junky/cli-tls/configs/chan_dahdi.conf.sample
    team/junky/cli-tls/configs/iax.conf.sample
    team/junky/cli-tls/configure
    team/junky/cli-tls/configure.ac
    team/junky/cli-tls/contrib/init.d/rc.debian.asterisk
    team/junky/cli-tls/doc/CODING-GUIDELINES
    team/junky/cli-tls/doc/datastores.txt
    team/junky/cli-tls/doc/janitor-projects.txt
    team/junky/cli-tls/doc/tex/channelvariables.tex
    team/junky/cli-tls/formats/format_wav_gsm.c
    team/junky/cli-tls/funcs/func_cdr.c
    team/junky/cli-tls/funcs/func_channel.c
    team/junky/cli-tls/funcs/func_curl.c
    team/junky/cli-tls/funcs/func_cut.c
    team/junky/cli-tls/funcs/func_enum.c
    team/junky/cli-tls/funcs/func_lock.c
    team/junky/cli-tls/funcs/func_odbc.c
    team/junky/cli-tls/funcs/func_realtime.c
    team/junky/cli-tls/funcs/func_sysinfo.c
    team/junky/cli-tls/funcs/func_vmcount.c
    team/junky/cli-tls/include/asterisk/agi.h
    team/junky/cli-tls/include/asterisk/app.h
    team/junky/cli-tls/include/asterisk/autoconfig.h.in
    team/junky/cli-tls/include/asterisk/calendar.h
    team/junky/cli-tls/include/asterisk/channel.h
    team/junky/cli-tls/include/asterisk/compiler.h
    team/junky/cli-tls/include/asterisk/devicestate.h
    team/junky/cli-tls/include/asterisk/frame.h
    team/junky/cli-tls/include/asterisk/linkedlists.h
    team/junky/cli-tls/include/asterisk/module.h
    team/junky/cli-tls/include/asterisk/monitor.h
    team/junky/cli-tls/include/asterisk/optional_api.h
    team/junky/cli-tls/include/asterisk/rtp_engine.h
    team/junky/cli-tls/include/asterisk/smdi.h
    team/junky/cli-tls/include/asterisk/utils.h
    team/junky/cli-tls/main/Makefile
    team/junky/cli-tls/main/ast_expr2.c
    team/junky/cli-tls/main/ast_expr2f.c
    team/junky/cli-tls/main/asterisk.c
    team/junky/cli-tls/main/autoservice.c
    team/junky/cli-tls/main/channel.c
    team/junky/cli-tls/main/cli.c
    team/junky/cli-tls/main/db.c
    team/junky/cli-tls/main/devicestate.c
    team/junky/cli-tls/main/event.c
    team/junky/cli-tls/main/features.c
    team/junky/cli-tls/main/file.c
    team/junky/cli-tls/main/frame.c
    team/junky/cli-tls/main/http.c
    team/junky/cli-tls/main/image.c
    team/junky/cli-tls/main/loader.c
    team/junky/cli-tls/main/logger.c
    team/junky/cli-tls/main/manager.c
    team/junky/cli-tls/main/pbx.c
    team/junky/cli-tls/main/rtp_engine.c
    team/junky/cli-tls/main/slinfactory.c
    team/junky/cli-tls/main/xmldoc.c
    team/junky/cli-tls/pbx/pbx_config.c
    team/junky/cli-tls/pbx/pbx_dundi.c
    team/junky/cli-tls/pbx/pbx_lua.c
    team/junky/cli-tls/pbx/pbx_realtime.c
    team/junky/cli-tls/res/ael/ael_lex.c
    team/junky/cli-tls/res/res_agi.c
    team/junky/cli-tls/res/res_agi.exports
    team/junky/cli-tls/res/res_calendar.c
    team/junky/cli-tls/res/res_calendar_caldav.c
    team/junky/cli-tls/res/res_calendar_exchange.c
    team/junky/cli-tls/res/res_calendar_icalendar.c
    team/junky/cli-tls/res/res_config_ldap.c
    team/junky/cli-tls/res/res_config_pgsql.c
    team/junky/cli-tls/res/res_jabber.c
    team/junky/cli-tls/res/res_monitor.c
    team/junky/cli-tls/res/res_monitor.exports
    team/junky/cli-tls/res/res_musiconhold.c
    team/junky/cli-tls/res/res_phoneprov.c
    team/junky/cli-tls/res/res_rtp_asterisk.c
    team/junky/cli-tls/res/res_smdi.c
    team/junky/cli-tls/res/res_smdi.exports
    team/junky/cli-tls/res/res_snmp.c
    team/junky/cli-tls/res/res_timing_dahdi.c

[... 9998 lines stripped ...]



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