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${ noResults }
18123 Commits (4f94236de5dd19aead1d171ff79e22885ee76475)
| Author | SHA1 | Message | Date |
|---|---|---|---|
Eliel C. Sardanons
|
4f94236de5 |
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) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@199411 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
Eliel C. Sardanons
|
d8e2ef0f30 |
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) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@199409 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
Eliel C. Sardanons
|
7cced65370 |
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) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@199376 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
Eliel C. Sardanons
|
65afefff9c |
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) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@199374 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
Russell Bryant
|
a05f90e709 |
minor tweak
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@199372 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
Russell Bryant
|
338e22b866 |
Constify a string and strip trailing whitespace.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@199370 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
Russell Bryant
|
27bd9d5f18 |
Switch from "echo -n" to printf. On my mac, the -n was just getting printed out.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@199368 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
David Vossel
|
d1d9beadc9 |
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/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@199298 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
Mark Michelson
|
d068c2f5e6 |
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 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@199227 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
David Vossel
|
e018606c7e |
Merged revisions 199138 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r199138 | dvossel | 2009-06-04 14:00:15 -0500 (Thu, 04 Jun 2009) | 3 lines Additional updates to AST-2009-001 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@199139 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
Eliel C. Sardanons
|
9ce385bd72 |
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) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@199091 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
Sean Bright
|
befad10893 |
Merged revisions 199022 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ 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/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@199051 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
Sean Bright
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51a8d4a01d |
Blocked revisions 198957 via svnmerge
........ 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) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@198958 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
David Vossel
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c42344b319 |
ast_call_forward() todo notes and originate flag copy.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@198954 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
David Vossel
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0c47c8f448 |
Blocked revisions 198891 via svnmerge
........ r198891 | dvossel | 2009-06-03 10:49:46 -0500 (Wed, 03 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 res_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/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@198892 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
David Vossel
|
3830c415c7 |
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/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@198856 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
David Vossel
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61bc1854d5 |
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 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@198824 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
Joshua Colp
|
5fcf193d7b |
Correct documentation for the register line, specifically where the domain should be specified.
(closes issue #14367) Reported by: Nick_Lewis git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@198791 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
Joshua Colp
|
4900d37d71 |
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 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@198762 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
Russell Bryant
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a13d62aa49 |
Tell the IAX2 parser about more control frame types.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@198729 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
Mark Michelson
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298d745fb4 |
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 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@198727 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
Tilghman Lesher
|
0fb1700522 |
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 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@198725 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
Russell Bryant
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8da5e991ee |
Minor whitespace fix.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@198670 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
Tilghman Lesher
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440e2e07e3 |
Blocked revisions 198665 via svnmerge
........ r198665 | tilghman | 2009-06-01 15:07:04 -0500 (Mon, 01 Jun 2009) | 7 lines If using the old deprecated format, a reload would cause the class to disappear. (closes issue #14759) Reported by: lidocaineus Patches: 20090518__issue14759.diff.txt uploaded by tilghman (license 14) Tested by: lmadsen ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@198666 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
Eliel C. Sardanons
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fb73ee6187 |
Moved more static documentation to the new AstXML form.
Moved more static docs to XML (pplications and manager actions): Monitor, StopMonitor, ChangeMonitor, PauseMonitor, UnpauseMonitor. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@198661 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
Tilghman Lesher
|
469fb6b79a |
Add information for new meetme realtime fields
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@198626 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
Eliel C. Sardanons
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5d8d1545d8 |
Do not add say.o in a separate line.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@198597 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
Eliel C. Sardanons
|
8d464b7211 |
Move JabberSend manager action from static docs to the AstXML form.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@198565 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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Eliel C. Sardanons
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1b59a1cd7d |
Move static documentation of E|Dead|AGI() application and manager action to XML.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@198561 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
David Vossel
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ab73b6e556 |
Fixed an issue in the threadstorage cli functions resulting from the constification of struct ast_cli_args in r196072.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@198558 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
Mark Michelson
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4c7c13d574 |
Remove extra lock from app_queue.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@198530 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
Mark Michelson
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0bde0b9ed2 |
Remove extra lock from local_indicate in connected line case.
Oh, and this fixes a deadlock I was seeing. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@198529 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
Mark Michelson
|
3166b6dac9 |
Add missing unlock of local pvt.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@198511 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
Mark Michelson
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0550a878f4 |
Remove documentation for the 'exten' argument to the AGENT function.
Since AgentCallbackLogin has been removed, this should not be documented any more. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@198500 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
Joshua Colp
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6e1bd8aad7 |
Fix a bug where the Event and Content-Type headers were added twice to outgoing SIP NOTIFY messages.
(closes issue #15239) Reported by: pj git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@198498 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
Tilghman Lesher
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ba6f16d55f |
Fix documentation for FIELDQTY.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@198470 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
Eliel C. Sardanons
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7ef2d7bca7 |
Filter the say.o object, it is being added later.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@198442 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
Russell Bryant
|
1630c85861 |
Constification and remove some unused code.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@198438 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
Eliel C. Sardanons
|
0c99bc31cb |
Avoid a crash when res_timing_dahdi is unloaded but wasn't properly loaded.
if dahdi_test_timer() fails, timing_funcs_handle remains NULL causing a crash when calling ast_unregister_timing_interface() with a NULL pointer. (closes issue #15234) Reported by: eliel Patches: timing_dahdi1.diff uploaded by eliel (license 64) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@198437 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
Russell Bryant
|
8580871fd4 |
Constify the ast_frame arg to ast_queue_frame().
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@198434 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
Sean Bright
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3353710e16 |
Properly terminate the receive buffer before sending to iksemel.
aji_io_recv takes the maximum number of bytes to read (instead of the total buffer size), so we have to subtract 1 from our buffer size. Without this, when we receive packets that are larger than our buffer, iksemel will choke and things get wonky. (closes issue #15232) Reported by: lp0 Patches: 05302009_res_jabber.c.patch uploaded by seanbright (license 71) Tested by: seanbright, lp0 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@198375 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
Sean Bright
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90c3db40ed |
Merged revisions 198370 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r198370 | seanbright | 2009-05-30 15:36:20 -0400 (Sat, 30 May 2009) | 12 lines Properly terminate AMI JabberSend response messages. The response message (either Error or Success) needs an extra trailing \r\n after the fields to inform the client that the message is complete. (closes issue #14876) Reported by: srt Patches: 05302009_1.4_res_jabber.c.diff uploaded by seanbright (license 71) asterisk_14876.patch uploaded by srt (license 378) trunk-14876-2.diff uploaded by phsultan (license 73) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@198371 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
Russell Bryant
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1ee78437e4 |
Merged revisions 198311 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r198311 | russell | 2009-05-29 22:42:46 -0500 (Fri, 29 May 2009) | 5 lines Fix a crash that occurred when MWI SMDI messages expired. (closes issue #14561) Reported by: cmoss28 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@198312 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
Sean Bright
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9241877c10 |
Merged revisions 198251 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r198251 | seanbright | 2009-05-29 22:46:41 -0400 (Fri, 29 May 2009) | 8 lines Treat an empty FORWARD_CONTEXT the same way we treat a missing one. (closes issue #15056) Reported by: p_lindheimer Patches: 05292009_bug15056.diff uploaded by seanbright (license 71) Tested by: p_lindheimer ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@198285 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
Joshua Colp
|
5c8626e315 |
When removing all packets from a dialog we also need to free the data if present.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@198248 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
Eliel C. Sardanons
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453a2f7331 |
Remove not used code in the Agent channel.
This code was there because of the AgentCallbackLogin() application. ->loginchan[] member was only used by AgentCallbackLogin(). Agent where dumped to astdb if they where logged in using AgentCallbacklogin() so they are not being dumper anymore. Review: https://reviewboard.asterisk.org/r/267/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@198217 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
Russell Bryant
|
58766cd2cf |
Suggesting that only a single timing module be loaded is no longer necessary.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@198186 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
Russell Bryant
|
04beecc859 |
Improve handling of trying to ACK too many timer expirations.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@198183 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
Terry Wilson
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c317d8f444 |
Add a couple of TODO items so I don't forget
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@198182 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
Russell Bryant
|
1fab70a1c6 |
Resolve issues with choppy sound when using res_timing_pthread.
The situation that caused this problem was when continuous mode was being
turned on and off while a rate was set for a timing interface. A very easy
way to replicate this bug was to do a Playback() from behind a Local channel.
In this scenario, a rate gets set on the channel for doing file playback.
At the same time, continuous mode gets turned on and off about every 20 ms
as frames get queued on to the PBX side channel from the other side of the
Local channel.
Essentially, this module treated continuous mode and a set rate as mutually
exclusive states for the timer to be in. When I dug deep enough, I observed
the following pattern:
1) Set timer to tick every 20 ms.
2) Wait almost 20 ms ...
3) Continuous mode gets turned on for a queued up frame
4) Continuous mode gets turned off
5) The timer goes back to its tick per 20 ms. state but starts counting
at 0 ms.
6) Goto step 2.
Sometimes, res_timing_pthread would make it 20 ms and produce a timer tick,
but not most of the time. This is what produced the choppy sound (or sometimes
no sound at all).
Now, the module treats continuous mode and a set rate as completely independent
timer modes. They can be enabled and disabled independently of each other and
things work as expected.
(closes issue #14412)
Reported by: dome
Patches:
issue14412.diff.txt uploaded by russell (license 2)
issue14412-1.6.1.0.diff.txt uploaded by russell (license 2)
Tested by: DennisD, russell
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@198146 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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17 years ago |