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2009-06-15More 'static' qualifiers on module global variables.Kevin P. Fleming
The 'pglobal' tool is quite handy indeed :-) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@200620 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-06-15Convert a number of global module variables to 'static'.Kevin P. Fleming
These modules all contained variables that are module-global but not system-global, but were not marked 'static'. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@200587 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-06-15Some minor structure size improvements in sip_pvt and sip_peer.Kevin P. Fleming
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. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@200584 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-06-15Redesigned 'optional API' support.Kevin P. Fleming
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. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@200519 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-06-15Merged revisions 200513 via svnmerge from Mark Michelson
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 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@200514 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-06-14added openr2 to menuselect-deps.in, recent commit in menuselect made me ↵Moises Silva
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 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@200477 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-06-12Include basic installation and usage instructions for upstart script.Sean Bright
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@200430 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-06-12First shot at an upstart script for asterisk on Ubuntu.Sean Bright
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. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@200428 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-06-12Merged revisions 200360 via svnmerge from Mark Michelson
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 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@200361 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-06-12Fix some bad locking stemming from trying to forward a call to a non-existentMark Michelson
extension from a queue. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@200326 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-06-12Fix a potential crash from trying to access a NULL channel pointer.Mark Michelson
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@200290 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-06-12Call chgrp instead of chown when setting run directory group ownership.Sean Bright
(issue #13153) Reported by: pabelanger git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@200254 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-06-11Blocked revisions 200185 via svnmergeSean Bright
........ r200185 | seanbright | 2009-06-11 18:20:31 -0400 (Thu, 11 Jun 2009) | 2 lines Backport fix for parallel build warnings from trunk r199781. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@200190 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-06-11Fix a crash due to a potentially NULL p->options.Mark Michelson
Thanks to mnicholson for pointing it out. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@200146 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-06-11Release the allocated channel decreasing the reference counter.Eliel C. Sardanons
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. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@200108 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-06-11Fix path for .flavor and .versionLeif Madsen
(issue #14737) Reported by: davidw Patches: flavor.patch uploaded by davidw (license 780) Tested by: davidw git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@200039 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-06-11Blocked revisions 200037 via svnmergeLeif Madsen
........ 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 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@200038 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-06-10Remove some trailing whitespace and steal revision 200000.Sean Bright
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@200000 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-06-10Only try to use the invite_branch on outgoing INVITEs with auth credentials.Mark Michelson
I have added a comment to the code to help ease understanding of the logic here as well. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@199958 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-06-10Fixes the argument order in definition of new_find_extension().David Brooks
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 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@199957 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-06-10Use ast_channel_unref to instead of ast_free on a newly created channel.Mark Michelson
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. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@199923 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-06-10Merged revisions 199856 via svnmerge from Sean Bright
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. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@199857 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-06-09CLI NOTIFY sending wrong transport type.David Vossel
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 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@199818 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-06-09Fix all of the parallel build warnings issued when running make -j#.Sean Bright
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@199781 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-06-09module load priorityDavid Vossel
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/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@199743 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-06-08Add sigaction janitorTilghman Lesher
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@199696 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-06-08Merged revisions 199626,199628 via svnmerge from Sean Bright
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. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@199630 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-06-08Fix a deadlock that could occur when setting rtp stats on SIP calls.Mark Michelson
(closes issue #15143) Reported by: cristiandimache Patches: 15143.patch uploaded by mmichelson (license 60) Tested by: cristiandimache git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@199588 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-06-07Move OSP* applications static documentation to XML.Eliel C. Sardanons
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) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@199547 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-06-07Move application ExternalIVR static documentation to XML.Eliel C. Sardanons
Move application ExternalIVR static documentation to the new AstXML form. (issue #15245) Reported by: eliel Patches: app_externalivr.diff uploaded by eliel (license 64) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@199514 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-06-07Global var cleanup - constification and removing unused vars.Russell Bryant
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@199479 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-06-06Move AGI command 'gosub' static documentation to XML.Eliel C. Sardanons
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) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@199446 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-06-06Move music on hold related applications documentation to XML.Eliel C. Sardanons
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) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@199413 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-06-06Move function PP_EACH_USER and PP_EACH_EXTENSION documentation to XML.Eliel C. Sardanons
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
2009-06-06Move function MEETME_INFO documentation to XML.Eliel C. Sardanons
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
2009-06-06Move function MINIVMACCOUNT and MINIVMCOUNTER static documentation to XML.Eliel C. Sardanons
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
2009-06-06Move function SYSINFO documentation to XML.Eliel C. Sardanons
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
2009-06-06minor tweakRussell Bryant
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@199372 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-06-06Constify a string and strip trailing whitespace.Russell Bryant
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@199370 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-06-06Switch from "echo -n" to printf. On my mac, the -n was just getting printed ↵Russell Bryant
out. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@199368 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-06-05Merged revisions 199297 via svnmerge from David Vossel
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
2009-06-05Correct "dahdi show channels" output when specifying a group.Mark Michelson
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
2009-06-04Merged revisions 199138 via svnmerge from David Vossel
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
2009-06-04Move static docs to the new AstXML form.Eliel C. Sardanons
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
2009-06-04Merged revisions 199022 via svnmerge from Sean Bright
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
2009-06-03Blocked revisions 198957 via svnmergeSean Bright
........ 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
2009-06-03ast_call_forward() todo notes and originate flag copy.David Vossel
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@198954 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-06-03Blocked revisions 198891 via svnmergeDavid Vossel
........ 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
2009-06-02Generic call forward api, ast_call_forward()David Vossel
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
2009-06-02fixes issue with channels not going down after transferDavid Vossel
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
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