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2011-07-14Merged revisions 328247 via svnmerge from lmadsen1-0/+1
https://origsvn.digium.com/svn/asterisk/branches/1.10 ................ r328247 | lmadsen | 2011-07-14 16:25:31 -0400 (Thu, 14 Jul 2011) | 14 lines Merged revisions 328209 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r328209 | lmadsen | 2011-07-14 16:13:06 -0400 (Thu, 14 Jul 2011) | 6 lines Introduce <support_level> tags in MODULEINFO. This change introduces MODULEINFO into many modules in Asterisk in order to show the community support level for those modules. This is used by changes committed to menuselect by Russell Bryant recently (r917 in menuselect). More information about the support level types and what they mean is available on the wiki at https://wiki.asterisk.org/wiki/display/AST/Asterisk+Module+Support+States ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@328259 f38db490-d61c-443f-a65b-d21fe96a405b
2011-05-03Merged revisions 316265 via svnmerge from russell1-7/+1
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r316265 | russell | 2011-05-03 14:55:49 -0500 (Tue, 03 May 2011) | 5 lines Fix a bunch of compiler warnings generated by gcc 4.6.0. Most of these are -Wunused-but-set-variable, but there were a few others mixed in here, as well. ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@316293 f38db490-d61c-443f-a65b-d21fe96a405b
2011-04-22Merged revisions 314959 via svnmerge from mnicholson1-62/+92
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r314959 | mnicholson | 2011-04-22 16:20:08 -0500 (Fri, 22 Apr 2011) | 24 lines Merged revisions 314958 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r314958 | mnicholson | 2011-04-22 15:49:45 -0500 (Fri, 22 Apr 2011) | 17 lines Merged revisions 311203,314908 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r311203 | mnicholson | 2011-03-17 14:14:37 -0500 (Thu, 17 Mar 2011) | 4 lines Don't hold the pvt lock while streaming a file. ABE-2756 ........ r314908 | mnicholson | 2011-04-22 15:01:48 -0500 (Fri, 22 Apr 2011) | 4 lines Prevent the login thread and the app threads from using the asterisk channel at the same time. ABE-2756 ........ ................ ................ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@314960 f38db490-d61c-443f-a65b-d21fe96a405b
2011-02-03Asterisk media architecture conversion - no more format bitfieldsdvossel1-35/+47
This patch is the foundation of an entire new way of looking at media in Asterisk. The code present in this patch is everything required to complete phase1 of my Media Architecture proposal. For more information about this project visit the link below. https://wiki.asterisk.org/wiki/display/AST/Media+Architecture+Proposal The primary function of this patch is to convert all the usages of format bitfields in Asterisk to use the new format and format_cap APIs. Functionally no change in behavior should be present in this patch. Thanks to twilson and russell for all the time they spent reviewing these changes. Review: https://reviewboard.asterisk.org/r/1083/ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@306010 f38db490-d61c-443f-a65b-d21fe96a405b
2011-01-04Merged revisions 300521 via svnmerge from lmadsen1-1/+1
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r300521 | lmadsen | 2011-01-04 15:53:27 -0600 (Tue, 04 Jan 2011) | 17 lines Merged revisions 300520 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r300520 | lmadsen | 2011-01-04 15:52:41 -0600 (Tue, 04 Jan 2011) | 9 lines Fix backwards and broken XML documentation. (closes issue #18547) Reported by: jcovert Patches: xmldoc.c.patch uploaded by jcovert (license 551) chan_iax2.c.doc.patch uploaded by jcovert (license 551) chan_sip.c.patch uploaded by jcovert (license 551) chan_agent.c.patch uploaded by jcovert (license 551) ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@300522 f38db490-d61c-443f-a65b-d21fe96a405b
2010-09-02Merged revisions 284610 via svnmerge from tilghman1-0/+2
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r284610 | tilghman | 2010-09-02 00:20:59 -0500 (Thu, 02 Sep 2010) | 10 lines When optional_api is non-optional, force dependent modules to be loaded. (closes issue #17707) Reported by: ira Patches: 20100819__issue17707__asterisk1.8.diff.txt uploaded by tilghman (license 14) Tested by: tilghman Review: https://reviewboard.asterisk.org/r/876/ ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@284628 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-20Add load priority order, such that preload becomes unnecessary in most casestilghman1-1/+2
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@278132 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-14ast_callerid restructuringrmudgett1-5/+8
The purpose of this patch is to eliminate struct ast_callerid since it has turned into a miscellaneous collection of various party information. Eliminate struct ast_callerid and replace it with the following struct organization: struct ast_party_name { char *str; int char_set; int presentation; unsigned char valid; }; struct ast_party_number { char *str; int plan; int presentation; unsigned char valid; }; struct ast_party_subaddress { char *str; int type; unsigned char odd_even_indicator; unsigned char valid; }; struct ast_party_id { struct ast_party_name name; struct ast_party_number number; struct ast_party_subaddress subaddress; char *tag; }; struct ast_party_dialed { struct { char *str; int plan; } number; struct ast_party_subaddress subaddress; int transit_network_select; }; struct ast_party_caller { struct ast_party_id id; char *ani; int ani2; }; The new organization adds some new information as well. * The party name and number now have their own presentation value that can be manipulated independently. ISDN supplies the presentation value for the name and number at different times with the possibility that they could be different. * The party name and number now have a valid flag. Before this change the name or number string could be empty if the presentation were restricted. Most channel drivers assume that the name or number is then simply not available instead of indicating that the name or number was restricted. * The party name now has a character set value. SIP and Q.SIG have the ability to indicate what character set a name string is using so it could be presented properly. * The dialed party now has a numbering plan value that could be useful to have available. The various channel drivers will need to be updated to support the new core features as needed. They have simply been converted to supply current functionality at this time. The following items of note were either corrected or enhanced: * The CONNECTEDLINE() and REDIRECTING() dialplan functions were consolidated into func_callerid.c to share party id handling code. * CALLERPRES() is now deprecated because the name and number have their own presentation values. * Fixed app_alarmreceiver.c write_metadata(). The workstring[] could contain garbage. It also can only contain the caller id number so using ast_callerid_parse() on it is silly. There was also a typo in the CALLERNAME if test. * Fixed app_rpt.c using ast_callerid_parse() on the channel's caller id number string. ast_callerid_parse() alters the given buffer which in this case is the channel's caller id number string. Then using ast_shrink_phone_number() could alter it even more. * Fixed caller ID name and number memory leak in chan_usbradio.c. * Fixed uninitialized char arrays cid_num[] and cid_name[] in sig_analog.c. * Protected access to a caller channel with lock in chan_sip.c. * Clarified intent of code in app_meetme.c sla_ring_station() and dial_trunk(). Also made save all caller ID data instead of just the name and number strings. * Simplified cdr.c set_one_cid(). It hand coded the ast_callerid_merge() function. * Corrected some weirdness with app_privacy.c's use of caller presentation. Review: https://reviewboard.asterisk.org/r/702/ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@276347 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-08Implement AstData API data providers as part of the GSOC 2010 project,eliel1-0/+88
midterm evaluation. Review: https://reviewboard.asterisk.org/r/757/ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@274727 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-03Merged revisions 273793 via svnmerge from tilghman1-1/+6
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r273793 | tilghman | 2010-07-02 16:36:39 -0500 (Fri, 02 Jul 2010) | 9 lines Have the DEADLOCK_AVOIDANCE macro warn when an unlock fails, to help catch potentially large software bugs. (closes issue #17407) Reported by: pdf Patches: 20100527__issue17407.diff.txt uploaded by tilghman (license 14) Review: https://reviewboard.asterisk.org/r/751/ ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@273830 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-16Fix the actual place that was pointed out, for previous commit.qwell1-0/+2
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@270983 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-16Merged revisions 270980 via svnmerge from qwell1-0/+2
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r270980 | qwell | 2010-06-16 16:10:09 -0500 (Wed, 16 Jun 2010) | 4 lines Need to lock the agent chan before access its internal bits. Pointed out by russellb on asterisk-dev mailing list. ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@270981 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-14Add option to get untruncated channel name from AGENT function.qwell1-0/+7
The "channel" option would chop the channel name at the last '-', which made it useless for something like a channel transfer from the dialplan. The "fullchannel" option will return the channel name as-is. ABE-2218 git-svn-id: http://svn.digium.com/svn/asterisk/trunk@270260 f38db490-d61c-443f-a65b-d21fe96a405b
2010-01-19small correction from 241314jpeeler1-2/+0
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@241315 f38db490-d61c-443f-a65b-d21fe96a405b
2010-01-19Merged revisions 241227 via svnmerge from jpeeler1-1/+19
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r241227 | jpeeler | 2010-01-19 11:22:18 -0600 (Tue, 19 Jan 2010) | 13 lines Fix deadlock in agent_read by removing call to agent_logoff. One must always lock the agents list lock before the agent private. agent_read locks the private immediately, so locking the agents list lock is not an option (which is what agent_logoff requires). Because agent_read already has access to the agent private all that is necessary is to do the required hanging up that agent_logoff performed. (closes issue #16321) Reported by: valon24 Patches: bug16321.patch uploaded by jpeeler (license 325) ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@241314 f38db490-d61c-443f-a65b-d21fe96a405b
2009-11-04Expand codec bitfield from 32 bits to 64 bits.tilghman1-13/+14
Reviewboard: https://reviewboard.asterisk.org/r/416/ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@227580 f38db490-d61c-443f-a65b-d21fe96a405b
2009-08-17Correct spelling of AGENTACCEPTDTMF in chan_agent.seanbright1-1/+1
(closes issue #15668) Reported by: davidw git-svn-id: http://svn.digium.com/svn/asterisk/trunk@212581 f38db490-d61c-443f-a65b-d21fe96a405b
2009-08-10AST-2009-005tilghman1-4/+4
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@211539 f38db490-d61c-443f-a65b-d21fe96a405b
2009-06-26Merge the new Channel Event Logging (CEL) subsystem.russell1-9/+9
CEL is the new system for logging channel events. This was inspired after facing many problems trying to represent what is possible to happen to a call in Asterisk using CDR records. For more information on CEL, see the built in HTML or PDF documentation generated from the files in doc/tex/. Many thanks to Steve Murphy (murf) and Brian Degenhardt (bmd) for their hard work developing this code. Also, thanks to Matt Nicholson (mnicholson) and Sean Bright (seanbright) for their assistance in the final push to get this code ready for Asterisk trunk. Review: https://reviewboard.asterisk.org/r/239/ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@203638 f38db490-d61c-443f-a65b-d21fe96a405b
2009-06-15Convert a number of global module variables to 'static'.kpfleming1-1/+1
These modules all contained variables that are module-global but not system-global, but were not marked 'static'. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@200587 f38db490-d61c-443f-a65b-d21fe96a405b
2009-06-01Remove documentation for the 'exten' argument to the AGENT function.mmichelson1-3/+0
Since AgentCallbackLogin has been removed, this should not be documented any more. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@198500 f38db490-d61c-443f-a65b-d21fe96a405b
2009-05-30Remove not used code in the Agent channel.eliel1-303/+36
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: http://svn.digium.com/svn/asterisk/trunk@198217 f38db490-d61c-443f-a65b-d21fe96a405b
2009-05-22Implement a new element in AstXML for AMI actions documentation.eliel1-12/+30
A new xml element was created to manage the AMI actions documentation, using AstXML. To register a manager action using XML documentation it is now possible using ast_manager_register_xml(). The CLI command 'manager show command' can be used to show the parsed documentation. Example manager xml documentation: <manager name="ami action name" language="en_US"> <synopsis> AMI action synopsis. </synopsis> <syntax> <xi:include xpointer="xpointer(...)" /> <-- for ActionID <parameter name="header1" required="true"> <para>Description</para> </parameter> ... </syntax> <description> <para>AMI action description</para> </description> <see-also> ... </see-also> </manager> git-svn-id: http://svn.digium.com/svn/asterisk/trunk@196308 f38db490-d61c-443f-a65b-d21fe96a405b
2009-05-21Const-ify the world (or at least a good part of it)kpfleming1-3/+3
This patch adds 'const' tags to a number of Asterisk APIs where they are appropriate (where the API already demanded that the function argument not be modified, but the compiler was not informed of that fact). The list includes: - CLI command handlers - CLI command handler arguments - AGI command handlers - AGI command handler arguments - Dialplan application handler arguments - Speech engine API function arguments In addition, various file-scope and function-scope constant arrays got 'const' and/or 'static' qualifiers where they were missing. Review: https://reviewboard.asterisk.org/r/251/ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@196072 f38db490-d61c-443f-a65b-d21fe96a405b
2009-04-24Convert the ast_channel data structure over to the astobj2 framework.russell1-4/+5
There is a lot that could be said about this, but the patch is a big improvement for performance, stability, code maintainability, and ease of future code development. The channel list is no longer an unsorted linked list. The main container for channels is an astobj2 hash table. All of the code related to searching for channels or iterating active channels has been rewritten. Let n be the number of active channels. Iterating the channel list has gone from O(n^2) to O(n). Searching for a channel by name went from O(n) to O(1). Searching for a channel by extension is still O(n), but uses a new method for doing so, which is more efficient. The ast_channel object is now a reference counted object. The benefits here are plentiful. Some benefits directly related to issues in the previous code include: 1) When threads other than the channel thread owning a channel wanted access to a channel, it had to hold the lock on it to ensure that it didn't go away. This is no longer a requirement. Holding a reference is sufficient. 2) There are places that now require less dealing with channel locks. 3) There are places where channel locks are held for much shorter periods of time. 4) There are places where dealing with more than one channel at a time becomes _MUCH_ easier. ChanSpy is a great example of this. Writing code in the future that deals with multiple channels will be much easier. Some additional information regarding channel locking and reference count handling can be found in channel.h, where a new section has been added that discusses some of the rules associated with it. Mark Michelson also assisted with the development of this patch. He did the conversion of ChanSpy and introduced a new API, ast_autochan, which makes it much easier to deal with holding on to a channel pointer for an extended period of time and having it get automatically updated if the channel gets masqueraded. Mark was also a huge help in the code review process. Thanks to David Vossel for his assistance with this branch, as well. David did the conversion of the DAHDIScan application by making it become a wrapper for ChanSpy internally. The changes come from the svn/asterisk/team/russell/ast_channel_ao2 branch. Review: http://reviewboard.digium.com/r/203/ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@190423 f38db490-d61c-443f-a65b-d21fe96a405b
2009-04-18Merged revisions 189203 via svnmerge from dvossel1-51/+63
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r189203 | dvossel | 2009-04-17 20:27:19 -0500 (Fri, 17 Apr 2009) | 12 lines Fixed autologoff in agents.conf not working when agent logs in via AgentLogin app An agent logs in by calling an extension that calls the AgentLogin app. In agents.conf ackcall=always is set, so when they get a call they have the choice to either acknowledge it or ignore it. autologoff=10 is set as well, so if the agent ignores the call over 10sec one may assume that the agent should be logged out (and in this case hungup on as well), but this was not happening. (closes issue #14091) Reported by: evandro Patches: autologoff.diff uploaded by dvossel (license 671) Review: http://reviewboard.digium.com/r/225/ ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@189204 f38db490-d61c-443f-a65b-d21fe96a405b
2009-04-03This commit introduces COLP/CONP and Redirecting party information into ↵mmichelson1-2/+1
Asterisk. The channel drivers which have been most heavily tested with these enhancements are chan_sip and chan_misdn. Further work is being done to add Q.SIG support and will be introduced in a later commit. chan_skinny has code added to it here, but according to user pj, the support on chan_skinny is not working as of now. This will be fixed in a later commit. A special thanks goes out to bugtracker user gareth for getting the ball rolling and providing the initial support for this work. Without his initial work on this, this would not have been nearly as painless as it was. This functionality has been tested by Digium's product quality department, as well as a customer site running thousands of calls every day. In addition, many many many many bugtracker users have tested this, too. (closes issue #8824) Reported by: gareth Review: http://reviewboard.digium.com/r/201 git-svn-id: http://svn.digium.com/svn/asterisk/trunk@186525 f38db490-d61c-443f-a65b-d21fe96a405b
2009-04-02Merge in the RTP engine API.file1-1/+0
This API provides a generic way for multiple RTP stacks to be integrated into Asterisk. Right now there is only one present, res_rtp_asterisk, which is the existing Asterisk RTP stack. Functionality wise this commit performs the same as previously. API documentation can be viewed in the rtp_engine.h header file. Review: http://reviewboard.digium.com/r/209/ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@186078 f38db490-d61c-443f-a65b-d21fe96a405b
2009-01-27Merged revisions 171689 via svnmerge from mmichelson1-64/+0
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r171689 | mmichelson | 2009-01-27 15:55:08 -0600 (Tue, 27 Jan 2009) | 39 lines Fix devicestate problems for "always-on" agent channels A revision to chan_agent attempted to "inherit" the device state of the underlying channel in order to report the device state of an agent channel more accurately. The problem with the logic here is that it makes no sense to use this for always-on agents. If the agent is logged in, then to the underlying channel, the agent will always appear to be "in use," no matter if the agent is on a call or not. The reason is that to the underlying channel, the channel is currently in use on a call to the AgentLogin application. The most common cause that I found for this issue to occur was for a SIP channel to be the underlying channel type for an Agent channel. If the SIP phone re-registers, then the registration will cause the device state core to query the device state of the SIP channel. Since the SIP channel is in use, the Agent channel would also inherit this status. Once the agent channel was set to "in use" there was no way that the device state could change on that channel unless the agent logged out. The solution for this problem is a bit different in 1.4 than it is in the other branches. In 1.4, there will be a one-line fix to make sure that only callback agents will inherit device state from their underlying channel type. For the other branches of Asterisk, since callback support has been removed, there is also no need for device state inheritance in chan_agent, so I will simply be removing it from the code. In addition, the 1.4 source is getting a new comment to help the next person who edits chan_agent.c. I'm adding a comment that a agent_pvt's loginchan field may be used to determine if the agent is a callback agent or not. (closes issue #14173) Reported by: nathan Patches: 14173.patch uploaded by putnopvut (license 60) Tested by: nathan, aramirez ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@171691 f38db490-d61c-443f-a65b-d21fe96a405b
2009-01-12Merged revisions 168507 via svnmerge from jpeeler1-19/+28
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r168507 | jpeeler | 2009-01-12 14:26:22 -0600 (Mon, 12 Jan 2009) | 9 lines (closes issue #12269) Reported by: IgorG Tested by: denisgalvao This gits rid of the notion of an owning_app allowing the request and hangup to be initiated by different threads. Originating from an active agent channel requires this. The implementation primarily changes __login_exec to wait on a condition variable rather than a lock. Review: http://reviewboard.digium.com/r/35/ ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@168508 f38db490-d61c-443f-a65b-d21fe96a405b
2008-12-05Janitor, use ARRAY_LEN() when possible.eliel1-2/+2
(closes issue #13990) Reported by: eliel Patches: array_len.diff uploaded by eliel (license 64) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@161218 f38db490-d61c-443f-a65b-d21fe96a405b
2008-11-29incorporates r159808 from branches/1.4:kpfleming1-1/+1
------------------------------------------------------------------------ r159808 | kpfleming | 2008-11-29 10:58:29 -0600 (Sat, 29 Nov 2008) | 7 lines update dev-mode compiler flags to match the ones used by default on Ubuntu Intrepid, so all developers will see the same warnings and errors since this branch already had some printf format attributes, enable checking for them and tag functions that didn't have them format attributes in a consistent way ------------------------------------------------------------------------ in addition: move some format attributes from main/utils.c to the header files they belong in, and fix up references to the relevant functions based on new compiler warnings git-svn-id: http://svn.digium.com/svn/asterisk/trunk@159818 f38db490-d61c-443f-a65b-d21fe96a405b
2008-11-26Don't allow for configuration options to overwrite optionsmmichelson1-5/+27
set via channel variables on a reload. (closes issue #13921) Reported by: davidw Patches: 13921.patch uploaded by putnopvut (license 60) Tested by: davidw git-svn-id: http://svn.digium.com/svn/asterisk/trunk@159437 f38db490-d61c-443f-a65b-d21fe96a405b
2008-11-10Merged revisions 155861 via svnmerge from mmichelson1-3/+9
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r155861 | mmichelson | 2008-11-10 15:07:39 -0600 (Mon, 10 Nov 2008) | 14 lines Channel drivers assume that when their indicate callback is invoked, that the channel on which the callback was called is locked. This patch corrects an instance in chan_agent where a channel's indicate callback is called directly without first locking the channel. This was leading to some observed locking issues in chan_local, but considering that all channel drivers operate under the same expectations, the generic fix in chan_agent is the right way to go. AST-126 ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@155863 f38db490-d61c-443f-a65b-d21fe96a405b
2008-11-05Add AgentLogin(), AgentMonitorOutgoing() applications andeliel1-40/+103
AGENT() function XML documentation. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@154837 f38db490-d61c-443f-a65b-d21fe96a405b
2008-10-09Reverting format addition for nowtilghman1-5/+2
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@148071 f38db490-d61c-443f-a65b-d21fe96a405b
2008-10-09Fudges for wav16, just like wav49tilghman1-2/+5
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@148070 f38db490-d61c-443f-a65b-d21fe96a405b
2008-09-19We should only unsubscribe to the device state eventmmichelson1-1/+3
subscription if we have previously subscribed. Otherwise a segfault will occur. (closes issue #13476) Reported by: jonnt Patches: 13476.patch uploaded by putnopvut (license 60) Tested by: jonnt git-svn-id: http://svn.digium.com/svn/asterisk/trunk@143609 f38db490-d61c-443f-a65b-d21fe96a405b
2008-09-12Create a new config file status, CONFIG_STATUS_FILEINVALID for differentiatingtilghman1-2/+15
when a file is invalid from when a file is missing. This is most important when we have two configuration files. Consider the following example: Old system: sip.conf users.conf Old result New result ======== ========== ========== ========== Missing Missing SIP doesn't load SIP doesn't load Missing OK SIP doesn't load SIP doesn't load Missing Invalid SIP doesn't load SIP doesn't load OK Missing SIP loads SIP loads OK OK SIP loads SIP loads OK Invalid SIP loads incompletely SIP doesn't load Invalid Missing SIP doesn't load SIP doesn't load Invalid OK SIP doesn't load SIP doesn't load Invalid Invalid SIP doesn't load SIP doesn't load So in the case when users.conf doesn't load because there's a typo that disrupts the syntax, we may only partially load users, instead of failing with an error, which may cause some calls not to get processed. Worse yet, the old system would do this with no indication that anything was even wrong. (closes issue #10690) Reported by: dtyoo Patches: 20080716__bug10690.diff.txt uploaded by Corydon76 (license 14) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@142992 f38db490-d61c-443f-a65b-d21fe96a405b
2008-09-05Merged revisions 141366 via svnmerge from mmichelson1-1/+1
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r141366 | mmichelson | 2008-09-05 16:10:32 -0500 (Fri, 05 Sep 2008) | 7 lines Agent's should not try to call a channel's indicate callback if the channel has been hung up. It will likely crash otherwise ABE-1159 ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@141367 f38db490-d61c-443f-a65b-d21fe96a405b
2008-08-19Merged revisions 138942 via svnmerge from mmichelson1-0/+6
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r138942 | mmichelson | 2008-08-19 18:17:17 -0500 (Tue, 19 Aug 2008) | 11 lines Reset agent_pvt variables back to the values in agents.conf (from what the corresponding channel variables were set to) when the agent logs out. (closes issue #13098) Reported by: davidw Patches: 20080731__issue13098_agent_ackcall_not_reset.diff uploaded by bbryant (license 36) Tested by: davidw ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@138943 f38db490-d61c-443f-a65b-d21fe96a405b
2008-08-09Merge more changes from the resolve-shadow-warnings branch (henceforth knownseanbright1-16/+16
as RSW since i am too lazy to keep typing it all out). This time a few of the channels. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@136888 f38db490-d61c-443f-a65b-d21fe96a405b
2008-07-25Deprecate *_device_state_* APIs in favor of *_devstate_* APIstilghman1-10/+10
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@133860 f38db490-d61c-443f-a65b-d21fe96a405b
2008-07-25Merged revisions 133649 via svnmerge from tilghman1-8/+15
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r133649 | tilghman | 2008-07-25 12:19:39 -0500 (Fri, 25 Jul 2008) | 8 lines Fix some errant device states by making the devicestate API more strict in terms of the device argument (only without the unique identifier appended). (closes issue #12771) Reported by: davidw Patches: 20080717__bug12771.diff.txt uploaded by Corydon76 (license 14) Tested by: davidw, jvandal, murf ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@133665 f38db490-d61c-443f-a65b-d21fe96a405b
2008-07-24I made this change from DEVICE_STATE to DEVICE_STATE_CHANGE, but I had it ↵russell1-1/+1
backwards, this is the right event to subscribe to ... git-svn-id: http://svn.digium.com/svn/asterisk/trunk@133486 f38db490-d61c-443f-a65b-d21fe96a405b
2008-07-17Instead of attempting to pass through AST_EVENT_DEVICE_STATE, use ↵russell1-1/+2
DEVICE_STATE_CHANGE instead. DEVICE_STATE is a state change on one server, and DEVICE_STATE_CHANGE is the "real" state of that device across all servers sharing state. This would have only been a problem with distributed device state. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@131643 f38db490-d61c-443f-a65b-d21fe96a405b
2008-07-16Add missing terminator to ast_event_subscribe to fix a crash.russell1-1/+1
(from rev 131206 in the 1.6.0 branch) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@131207 f38db490-d61c-443f-a65b-d21fe96a405b
2008-07-13Unlock list before returningtilghman1-0/+1
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@130444 f38db490-d61c-443f-a65b-d21fe96a405b
2008-07-11Merged revisions 130102 via svnmerge from tilghman1-1/+59
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r130102 | tilghman | 2008-07-11 11:50:42 -0500 (Fri, 11 Jul 2008) | 9 lines Pass the devicestate from an underlying channel up through the Agent channel. This should make the Agent always report the correct device state, even when the underlying channel is used for other purposes. (closes issue #12773) Reported by: davidw Patches: 20080710__bug12773.diff.txt uploaded by Corydon76 (license 14) Tested by: davidw ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@130126 f38db490-d61c-443f-a65b-d21fe96a405b
2008-07-02Merged revisions 127560 via svnmerge from mmichelson1-0/+9
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r127560 | mmichelson | 2008-07-02 15:47:38 -0500 (Wed, 02 Jul 2008) | 3 lines Fix thread-safety of some of the pbx_builtin_getvar_helper calls ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@127562 f38db490-d61c-443f-a65b-d21fe96a405b