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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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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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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 15:15:48 -0500 (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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r199923 | mmichelson | 2009-06-10 13:58:12 -0500 (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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r199857 | seanbright | 2009-06-10 12:10:23 -0400 (Wed, 10 Jun 2009) | 9 lines
Merged revisions 199856 via svnmerge from
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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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r199818 | dvossel | 2009-06-09 15:47:57 -0500 (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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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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r199743 | dvossel | 2009-06-09 11:22:04 -0500 (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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r199630 | seanbright | 2009-06-08 15:33:09 -0400 (Mon, 08 Jun 2009) | 32 lines
Merged revisions 199626,199628 via svnmerge from
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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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r199588 | mmichelson | 2009-06-08 12:32:04 -0500 (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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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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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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r199479 | russell | 2009-06-07 09:55:51 -0500 (Sun, 07 Jun 2009) | 2 lines
Global var cleanup - constification and removing unused vars.
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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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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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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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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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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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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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r199372 | russell | 2009-06-06 16:42:31 -0500 (Sat, 06 Jun 2009) | 1 line
minor tweak
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r199370 | russell | 2009-06-06 16:40:56 -0500 (Sat, 06 Jun 2009) | 2 lines
Constify a string and strip trailing whitespace.
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r199368 | russell | 2009-06-06 16:38:54 -0500 (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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r199298 | dvossel | 2009-06-05 16:21:22 -0500 (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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r199227 | mmichelson | 2009-06-05 08:51:08 -0500 (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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r199139 | dvossel | 2009-06-04 14:10:16 -0500 (Thu, 04 Jun 2009) | 9 lines
Merged revisions 199138 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r199138 | dvossel | 2009-06-04 14:00:15 -0500 (Thu, 04 Jun 2009) | 3 lines
Additional updates to AST-2009-001
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r199091 | eliel | 2009-06-04 12:29:50 -0400 (Thu, 04 Jun 2009) | 11 lines
Move static docs to the new AstXML form.
Move SMDI_MSG and SMDI_MSG_RETRIEVE functions statis documentation
to XML.
(issue #15245)
Reported by: eliel
Patches:
res_smdi_static_conversion.txt uploaded by lmadsen (license 10)
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r199051 | seanbright | 2009-06-04 10:31:24 -0400 (Thu, 04 Jun 2009) | 47 lines
Merged revisions 199022 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4
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r199022 | seanbright | 2009-06-04 10:14:57 -0400 (Thu, 04 Jun 2009) | 40 lines
Safely handle AMI connections/reload requests that occur during startup.
During asterisk startup, a lock on the list of modules is obtained by the
primary thread while each module is initialized. Issue 13778 pointed out a
problem with this approach, however. Because the AMI is loaded before other
modules, it is possible for a module reload to be issued by a connected client
(via Action: Command), causing a deadlock.
The resolution for 13778 was to move initialization of the manager to happen
after the other modules had already been lodaded. While this fixed this
particular issue, it caused a problem for users (like FreePBX) who call AMI
scripts via an #exec in a configuration file (See issue 15189).
The solution I have come up with is to defer any reload requests that come in
until after the server is fully booted. When a call comes in to
ast_module_reload (from wherever) before we are fully booted, the request is
added to a queue of pending requests. Once we are done booting up, we then
execute these deferred requests in turn.
Note that I have tried to make this a bit more intelligent in that it will not
queue up more than 1 request for the same module to be reloaded, and if a
general reload request comes in ('module reload') the queue is flushed and we
only issue a single deferred reload for the entire system.
As for how this will impact existing installations - Before 13778, a reload
issued before module initialization was completed would result in a deadlock.
After 13778, you simply couldn't connect to the manager during startup (which
causes problems with #exec-that-calls-AMI configuration files). I believe this
is a good general purpose solution that won't negatively impact existing
installations.
(closes issue #15189)
(closes issue #13778)
Reported by: p_lindheimer
Patches:
06032009_15189_deferred_reloads.diff uploaded by seanbright (license 71)
Tested by: p_lindheimer, seanbright
Review: https://reviewboard.asterisk.org/r/272/
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r198958 | seanbright | 2009-06-03 16:49:11 -0400 (Wed, 03 Jun 2009) | 17 lines
Blocked revisions 198957 via svnmerge
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r198957 | seanbright | 2009-06-03 16:39:10 -0400 (Wed, 03 Jun 2009) | 11 lines
Fix a possible crash in pbx_spool.
We were trying to reference members of a struct that had previously been freed.
This patch makes sure that we free the struct after it has been removed from
the spooler queue.
(closes issue #15072)
Reported by: garlew
Patches:
spool.diff uploaded by garlew (license 376)
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r198954 | dvossel | 2009-06-03 15:30:10 -0500 (Wed, 03 Jun 2009) | 3 lines
ast_call_forward() todo notes and originate flag copy.
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r198856 | dvossel | 2009-06-02 16:17:49 -0500 (Tue, 02 Jun 2009) | 10 lines
Generic call forward api, ast_call_forward()
The function ast_call_forward() forwards a call to an extension specified in an ast_channel's call_forward string. After an ast_channel is called, if the channel's call_forward string is set this function can be used to forward the call to a new channel and terminate the original one. I have included this api call in both channel.c's ast_request_and_dial() and feature.c's feature_request_and_dial(). App_dial and app_queue already contain call forward logic specific for their application and options.
(closes issue #13630)
Reported by: festr
Review: https://reviewboard.asterisk.org/r/271/
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r198824 | dvossel | 2009-06-02 12:55:35 -0500 (Tue, 02 Jun 2009) | 8 lines
fixes issue with channels not going down after transfer
Iax2 currently does not support native bridging if the timeoutms value is set. We check for that in iax2_bridge, but then set timeoutms to 0 by default. If the timeoutms is not provided it is set to -1. By setting timeoutms to 0 it is processed causing a bridging retry loop.
(closes issue #15216)
Reported by: oxymoron
Tested by: dvossel
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r198791 | file | 2009-06-02 10:48:06 -0300 (Tue, 02 Jun 2009) | 5 lines
Correct documentation for the register line, specifically where the domain should be specified.
(closes issue #14367)
Reported by: Nick_Lewis
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r198762 | file | 2009-06-02 10:12:59 -0300 (Tue, 02 Jun 2009) | 5 lines
Fix a bug where we were passing in address information that should remain unmodified to a function that may modify it.
(closes issue #15243)
Reported by: pj
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r198729 | russell | 2009-06-01 16:03:18 -0500 (Mon, 01 Jun 2009) | 2 lines
Tell the IAX2 parser about more control frame types.
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r198727 | mmichelson | 2009-06-01 15:57:31 -0500 (Mon, 01 Jun 2009) | 13 lines
Add the ability to execute connected line interception macros.
When connected line updates are received or generated in the middle
of an application call, it is now possible to execute a macro to
manipulate the connected line data. This way, phone numbers may be
manipulated to be more presentable to users, names may be changed
for...whatever reason, or whatever else needs to be done may be.
Review: https://reviewboard.asterisk.org/r/256
AST-165
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r198725 | tilghman | 2009-06-01 15:33:50 -0500 (Mon, 01 Jun 2009) | 8 lines
Add INCrement and DECrement functions
(closes issue #15025)
Reported by: greenfieldtech
Patches:
func_math.c.patch_v4 uploaded by greenfieldtech (license 369)
slightly modified by me
Tested by: greenfieldtech, lmadsen
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r198670 | russell | 2009-06-01 15:17:50 -0500 (Mon, 01 Jun 2009) | 2 lines
Minor whitespace fix.
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r198661 | eliel | 2009-06-01 15:37:30 -0400 (Mon, 01 Jun 2009) | 6 lines
Moved more static documentation to the new AstXML form.
Moved more static docs to XML (pplications and manager actions):
Monitor, StopMonitor, ChangeMonitor, PauseMonitor, UnpauseMonitor.
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r198626 | tilghman | 2009-06-01 13:40:35 -0500 (Mon, 01 Jun 2009) | 2 lines
Add information for new meetme realtime fields
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r198597 | eliel | 2009-06-01 13:53:38 -0400 (Mon, 01 Jun 2009) | 3 lines
Do not add say.o in a separate line.
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r198565 | eliel | 2009-06-01 12:09:42 -0400 (Mon, 01 Jun 2009) | 2 lines
Move JabberSend manager action from static docs to the AstXML form.
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r198561 | eliel | 2009-06-01 11:38:48 -0400 (Mon, 01 Jun 2009) | 2 lines
Move static documentation of E|Dead|AGI() application and manager action to XML.
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r198529 | mmichelson | 2009-06-01 09:42:57 -0500 (Mon, 01 Jun 2009) | 5 lines
Remove extra lock from local_indicate in connected line case.
Oh, and this fixes a deadlock I was seeing.
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r198530 | mmichelson | 2009-06-01 09:45:43 -0500 (Mon, 01 Jun 2009) | 3 lines
Remove extra lock from app_queue.
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r198500 | mmichelson | 2009-06-01 09:02:05 -0500 (Mon, 01 Jun 2009) | 6 lines
Remove documentation for the 'exten' argument to the AGENT function.
Since AgentCallbackLogin has been removed, this should not be documented
any more.
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r198498 | file | 2009-06-01 10:31:27 -0300 (Mon, 01 Jun 2009) | 5 lines
Fix a bug where the Event and Content-Type headers were added twice to outgoing SIP NOTIFY messages.
(closes issue #15239)
Reported by: pj
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