[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
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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
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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
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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/
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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.
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r199370 | russell | 2009-06-06 17:40:56 -0400 (Sat, 06 Jun 2009) | 2 lines
Constify a string and strip trailing whitespace.
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r199372 | russell | 2009-06-06 17:42:31 -0400 (Sat, 06 Jun 2009) | 1 line
minor tweak
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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)
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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)
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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)
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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)
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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)
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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)
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r199479 | russell | 2009-06-07 10:55:51 -0400 (Sun, 07 Jun 2009) | 2 lines
Global var cleanup - constification and removing unused vars.
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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)
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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)
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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
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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
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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
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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.
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r199696 | tilghman | 2009-06-08 18:08:44 -0400 (Mon, 08 Jun 2009) | 1 line
Add sigaction janitor
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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/
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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#.
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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
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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
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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.
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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.
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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
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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.
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r200000 | seanbright | 2009-06-10 16:40:41 -0400 (Wed, 10 Jun 2009) | 1 line
Remove some trailing whitespace and steal revision 200000.
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r200038 | lmadsen | 2009-06-11 08:13:49 -0400 (Thu, 11 Jun 2009) | 14 lines
Blocked revisions 200037 via svnmerge
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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
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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
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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.
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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.
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r200190 | seanbright | 2009-06-11 18:21:32 -0400 (Thu, 11 Jun 2009) | 8 lines
Blocked revisions 200185 via svnmerge
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r200185 | seanbright | 2009-06-11 18:20:31 -0400 (Thu, 11 Jun 2009) | 2 lines
Backport fix for parallel build warnings from trunk r199781.
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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
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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.
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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.
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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
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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
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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.
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r200430 | seanbright | 2009-06-12 15:46:25 -0400 (Fri, 12 Jun 2009) | 1 line
Include basic installation and usage instructions for upstart script.
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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
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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
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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
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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.
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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.
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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'.
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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 :-)
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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.
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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
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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.
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r200762 | russell | 2009-06-15 21:26:20 -0400 (Mon, 15 Jun 2009) | 2 lines
Add missing closure of verbatim environment.
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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.
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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
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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.
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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)'.
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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
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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)'.
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r200942 | russell | 2009-06-16 11:26:57 -0400 (Tue, 16 Jun 2009) | 2 lines
Add Sean Bright to CREDITS - Thanks, Sean!
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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)
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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/
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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.
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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
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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/
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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.
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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.
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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.
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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).
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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 :))
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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
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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
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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)
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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/
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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/
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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/
........
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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
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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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