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2010-07-20Add load priority order, such that preload becomes unnecessary in most casesTilghman Lesher
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@278132 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-07-14ast_callerid restructuringRichard Mudgett
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: https://origsvn.digium.com/svn/asterisk/trunk@276347 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-07-09Kill some startup warnings and errors and make some messages more helpful in ↵Tilghman Lesher
tracking down the source. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@275105 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-07-02Fix various typos reported by LintianTzafrir Cohen
(Also fix the typos in the comments) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@273641 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-06-16MSG_OOB flag on HANGUP packet removed.Paul Belanger
Per Tilghman's request on IRC (#asterisk-bugs). (closes issue #17506) Reported by: brycebaril Tested by: pabelanger, tilghman git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@270936 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-04-27Fix compile on systems without HAVE_NULLSAFE_PRINTF defined.Jason Parker
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@259617 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-04-03Consolidate ast_channel.cid.cid_rdnis into ast_channel.redirecting.from.number.Richard Mudgett
SWP-1229 ABE-2161 * Ensure chan_local.c:local_call() will not leak cid.cid_dnid when copying. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@256104 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-03-25handle_speechset has 4 arguments.Leif Madsen
Update code to reflect that handle_speechset has 4 arguments. (closes issue #17093) Reported by: gpatri Patches: res_agi.patch uploaded by gpatri (license 1014) Tested by: pabelanger, mmichelson git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@254446 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-02-10Solaris doesn't like outputting a NULL to a %s in format strings.Tilghman Lesher
Detect all platforms that don't like that, either, and ensure that when documentation is missing, we pass a non-NULL pointer when outputting the corresponding documentation. (closes issue #16689) Reported by: bklang Patches: 20100209__issue16689__with_tests.diff.txt uploaded by tilghman (license 14) Review: https://reviewboard.asterisk.org/r/497/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@246030 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-01-19Create iterative method for querying SRV results, and use that for finding ↵Tilghman Lesher
AGI servers. (closes issue #14775) Reported by: _brent_ Patches: 20091215__issue14775.diff.txt uploaded by tilghman (license 14) hagi-5.patch uploaded by brent (license 388) Tested by: _brent_ Reviewboard: https://reviewboard.asterisk.org/r/378/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@241188 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-01-08Initialize variables that we attempt to free later.Tilghman Lesher
(closes issue #16302) Reported by: yahsyn Patches: 20091124__issue16302.diff.txt uploaded by tilghman (license 14) Tested by: yahsyn git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@238754 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-01-04Merged revisions 237405 via svnmerge from Tilghman Lesher
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r237405 | tilghman | 2010-01-04 12:19:00 -0600 (Mon, 04 Jan 2010) | 16 lines Add a flag to disable the Background behavior, for AGI users. This is in a section of code that relates to two other issues, namely issue #14011 and issue #14940), one of which was the behavior of Background when called with a context argument that matched the current context. This fix broke FreePBX, however, in a post-Dial situation. Needless to say, this is an extremely difficult collision of several different issues. While the use of an exception flag is ugly, fixing all of the issues linked is rather difficult (although if someone would like to propose a better solution, we're happy to entertain that suggestion). (closes issue #16434) Reported by: rickead2000 Patches: 20091217__issue16434.diff.txt uploaded by tilghman (license 14) 20091222__issue16434__1.6.1.diff.txt uploaded by tilghman (license 14) Tested by: rickead2000 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@237406 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2010-01-04Fix timeout for AGI command speech recognize.Jeff Peeler
(closes issue #16297) Reported by: semond git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@237323 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-12-23Merged revisions 236184 via svnmerge from Tilghman Lesher
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r236184 | tilghman | 2009-12-22 20:55:24 -0600 (Tue, 22 Dec 2009) | 4 lines If EXEC only gets a single argument, don't crash when the second is used. (closes issue #16504) Reported by: bklang ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@236186 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-12-15Merged revisions 235052 via svnmerge from Tilghman Lesher
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r235052 | tilghman | 2009-12-15 09:29:24 -0600 (Tue, 15 Dec 2009) | 4 lines Mandatory argument checking (closes issue #16446) Reported by: nicchap ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@235053 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-11-04Expand codec bitfield from 32 bits to 64 bits.Tilghman Lesher
Reviewboard: https://reviewboard.asterisk.org/r/416/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@227580 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-08-10AST-2009-005Tilghman Lesher
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@211539 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-06-26Merge the new Channel Event Logging (CEL) subsystem.Russell Bryant
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: https://origsvn.digium.com/svn/asterisk/trunk@203638 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
2009-06-01Move static documentation of E|Dead|AGI() application and manager action to XML.Eliel C. Sardanons
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@198561 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-05-24Move AGI static documentation to the new AstXML form.Eliel C. Sardanons
Move AGI commands documentation to XML docs: 'set priority' 'set variable' 'stream file' 'control stream file' 'tdd mode' 'verbose' 'wait for digit' 'speech create' 'speech set' 'speech destroy' 'speech load grammar' 'speech unload grammar' 'speech activate grammar' 'speech deactivate grammar' 'speech recognize' git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@196585 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-05-23Move static AGI commands documentation to XML.Eliel C. Sardanons
Move AGI commands ('say datetime', 'send image', 'send text', 'set autohangup', 'set callerid', 'set context', 'set extension') documentation to the AstXML form. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@196554 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-05-22Moved static documentation to the AstXML form.Eliel C. Sardanons
Moved AGI commands static documentation to XML docs ('say alpha', 'say digits', 'say number', 'say phonetic', 'say date' and 'say time'). git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@196344 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-05-22Implement a new element in AstXML for AMI actions documentation.Eliel C. Sardanons
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: https://origsvn.digium.com/svn/asterisk/trunk@196308 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-05-22Fix res_agi compilation after the const-ify the world merge.Sean Bright
Since we are dealing with a 'const char * const' now, we have to create a temporary copy of the string to work on rather than the original. Fix inspired by reporter. Reviewed by everyone-and-their-mother in #asterisk-dev. (closes issue #15184) Reported by: andrew git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@196270 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-05-21Const-ify the world (or at least a good part of it)Kevin P. Fleming
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: https://origsvn.digium.com/svn/asterisk/trunk@196072 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-05-18Move AGI documentation from static to the XML form.Eliel C. Sardanons
Move the AGI commands 'receive text', 'receive char' and 'record' static documentation to XML docs. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@195365 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-05-04Restore 'asyncagi break' command to 1.6.1 and higher.Tilghman Lesher
(closes issue #14985) Reported by: nikkk Patches: 20090428__bug14985.diff.txt uploaded by tilghman (license 14) 20090429__bug14985__1.6.1.diff.txt uploaded by tilghman (license 14) Tested by: nikkk git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@192171 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-04-29Merge str_substitution branch.Tilghman Lesher
This branch adds additional methods to dialplan functions, whereby the result buffers are now dynamic buffers, which can be expanded to the size of any result. No longer are variable substitutions limited to 4095 bytes of data. In addition, the common case of needing buffers much smaller than that will enable substitution to only take up the amount of memory actually needed. The existing variable substitution routines are still available, but users of those API calls should transition to using the dynamic-buffer APIs. Reviewboard: http://reviewboard.digium.com/r/174/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@191140 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-04-24Convert the ast_channel data structure over to the astobj2 framework.Russell Bryant
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: https://origsvn.digium.com/svn/asterisk/trunk@190423 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-03-27Fix speech structure leak in the AGI speech recognition integration.Joshua Colp
The AGI dialplan applications did not destroy the speech structure automatically if it was not destroyed by the running AGI script. They will now do this. (issue LUMENVOX-15) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@184673 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-03-18Merged revisions 182810 via svnmerge from Russell Bryant
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r182810 | russell | 2009-03-17 21:09:13 -0500 (Tue, 17 Mar 2009) | 44 lines Fix cases where the internal poll() was not being used when it needed to be. We have seen a number of problems caused by poll() not working properly on Mac OSX. If you search around, you'll find a number of references to using select() instead of poll() to work around these issues. In Asterisk, we've had poll.c which implements poll() using select() internally. However, we were still getting reports of problems. vadim investigated a bit and realized that at least on his system, even though we were compiling in poll.o, the system poll() was still being used. So, the primary purpose of this patch is to ensure that we're using the internal poll() when we want it to be used. The changes are: 1) Remove logic for when internal poll should be used from the Makefile. Instead, put it in the configure script. The logic in the configure script is the same as it was in the Makefile. Ideally, we would have a functionality test for the problem, but that's not actually possible, since we would have to be able to run an application on the _target_ system to test poll() behavior. 2) Always include poll.o in the build, but it will be empty if AST_POLL_COMPAT is not defined. 3) Change uses of poll() throughout the source tree to ast_poll(). I feel that it is good practice to give the API call a new name when we are changing its behavior and not using the system version directly in all cases. So, normally, ast_poll() is just redefined to poll(). On systems where AST_POLL_COMPAT is defined, ast_poll() is redefined to ast_internal_poll(). 4) Change poll() in main/poll.c to be ast_internal_poll(). It's worth noting that any code that still uses poll() directly will work fine (if they worked fine before). So, for example, out of tree modules that are using poll() will not stop working or anything. However, for modules to work properly on Mac OSX, ast_poll() needs to be used. (closes issue #13404) Reported by: agalbraith Tested by: russell, vadim http://reviewboard.digium.com/r/198/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@182847 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-01-24Add a todo to finish the XML docs in this moduleRussell Bryant
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@170902 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-01-12Merged revisions 168516 via svnmerge from Jeff Peeler
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r168516 | jpeeler | 2009-01-12 15:42:34 -0600 (Mon, 12 Jan 2009) | 5 lines (closes issue #13881) Reported by: hoowa Update the app CDR field for AGI commands that are not executing an application via "exec". ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@168517 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-01-09When using ast_str with a non-ast_str-enabled API, we need to update the bufferTilghman Lesher
or otherwise, we cannot use ast_str_strlen(). git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@168090 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-01-08Merged revisions 167840 via svnmerge from Tilghman Lesher
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r167840 | tilghman | 2009-01-08 16:08:56 -0600 (Thu, 08 Jan 2009) | 6 lines Don't truncate database results at 255 chars. (closes issue #14069) Reported by: evandro Patches: 20081214__bug14069.diff.txt uploaded by Corydon76 (license 14) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@167894 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-12-22Always use the value of the AGISIGHUP when running an AGI.Mark Michelson
Prior to this patch, the value of AGISIGUP was not always honored when set on a channel. (closes issue #13711) Reported by: fmueller Patches: 13711.patch uploaded by putnopvut (license 60) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@166470 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-12-22Remove AST_PBX_KEEPALIVE usage from res_agi.Russell Bryant
This patch removes the usage of AST_PBX_KEEPALIVE from res_agi. The only usage was for the AGI command, "asyncagi break". This patch removes this feature. Normally, a feature would not be removed like this. However, this code is broken and usage of it will result in a memory leak. Usage of this feature will make the AGI code return a result of AST_PBX_KEEPALIVE. The PBX handler assumes that another thread has assumed ownership of the channel. The channel thread will exit without destroying the channel. Unfortunately, _no_ thread has ownership of the channel at this point. There are a couple of serious problems here: 1) The only way to recover the caller is to issue a channel redirect. This will work, but this will be done with a masquerade, and the old ast_channel structure will be lost. 2) Until the channel redirect happens, there is no code servicing the channel. That means nothing is reading audio or handling events coming from the channel. This is very bad. The recommended way to get this same "break" functionality is to issue the redirect while the channel is still being handled by the AGI code. That way, there will be no memory leak, and there will be no period of time that the channel is not being serviced. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@166258 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-12-13Merge ast_str_opaque branch (discontinue usage of ast_str internals)Tilghman Lesher
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@163991 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-12-11Merged revisions 163088 via svnmerge from Tilghman Lesher
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r163088 | tilghman | 2008-12-11 10:51:27 -0600 (Thu, 11 Dec 2008) | 6 lines Don't wait forever, if there's a specified recording timeout. (closes issue #13885) Reported by: bamby Patches: res_agi.c.patch uploaded by bamby (license 430) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@163089 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-12-05Janitor, use ARRAY_LEN() when possible.Eliel C. Sardanons
(closes issue #13990) Reported by: eliel Patches: array_len.diff uploaded by eliel (license 64) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@161218 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-12-04Added XML documentation for the following AGI commands:Eliel C. Sardanons
- get option - get variable - hangup - noop git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@160896 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-11-26improve handling of API calls provided by loaded modules through use of some ↵Kevin P. Fleming
GCC features; this makes app_stack's usage of AGI APIs even cleaner, and will allow it to work 'as expected' either with or without res_agi being loaded reviewed at http://reviewboard.digium.com/r/62 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@159631 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-11-24Added EVENT_FLAG_AGI and used it for manager calls in res_agi.cMatthew Nicholson
(closes issue #13873) Reported by: fnordian Patches: ami_agievent.patch uploaded by fnordian (license 110) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@158876 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-11-19correct small bug introduced during API conversionKevin P. Fleming
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@157743 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-11-19make some corrections to the ast_agi_register_multiple(), ↵Kevin P. Fleming
ast_agi_unregister_multiple() and ast_agi_fdprintf() API calls to be consistent with API guidelines also, move UPGRADE.txt to UPGRADE-1.6.txt and make the new UPGRADE.txt contain information about upgrading between Asterisk 1.6 releases git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@157706 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-11-12Add XML documentation for AGI commands:Eliel C. Sardanons
- database deltree - database get - exec - get data - get full variable git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@156125 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-11-12This commit does two things:Michiel van Baak
- Add CLI aliases module to asterisk. - Remove all deprecated CLI commands from the code Initial work done by file. Junk-Y and lmadsen did a lot of work and testing to get the list of deprecated commands into the configuration file. Deprecated CLI commands are now handled by this new module, see cli_aliases.conf for more info about that. ok russellb@ via reviewboard (closes issue #13735) Reported by: mvanbaak git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@156120 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-11-12- Add 'database del', 'database put' and 'set music' AGI commands XML ↵Eliel C. Sardanons
documentation. - Add to the DTD the possibility to put a parameter inside an <enum>. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@156087 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2008-11-12Implement AGI XML documentation parsing functions.Eliel C. Sardanons
A new <agi> element is used to describe the XML documentation. We have the usual synopsis,syntax,description and seealso for AGI commands. The CLI 'agi show commands' command was changed to show all the documentation se ctions. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@156051 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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