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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
2009-06-25Add support for multicast RTP paging.Joshua Colp
(closes issue #11797) Reported by: macbrody Review: https://reviewboard.asterisk.org/r/270/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@203227 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-06-19Fix 2 typos and add support for wide character types.Tilghman Lesher
Reported by Benny Amorsen via the asterisk-users mailing list. http://lists.digium.com/pipermail/asterisk-users/2009-June/233622.html git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@201904 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-06-18fixes some memory leaks and redundant conditionsDavid Vossel
(closes issue #15269) Reported by: contactmayankjain Patches: patch.txt uploaded by contactmayankjain (license 740) memory_leak_stuff.trunk.diff uploaded by dvossel (license 671) Tested by: contactmayankjain, dvossel git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@201678 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-06-18Merged revisions 201600 via svnmerge from Russell Bryant
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r201600 | russell | 2009-06-18 10:24:31 -0500 (Thu, 18 Jun 2009) | 29 lines Fix memory corruption and leakage related reloads of non files mode MoH classes. For Music on Hold classes that are not files mode, meaning that we are executing an application that will feed us audio data, we use a thread to monitor the external application and read audio from it. This thread also makes use of the MoH class object. In the MoH class destructor, we used pthread_cancel() to ask the thread to exit. Unfortunately, the code did not wait to ensure that the thread actually went away. What needed to be done is a pthread_join() to ensure that the thread fully cleans up before we proceed. By adding this one line, we resolve two significant problems: 1) Since the thread was never joined, it never fully goes away. So, on every reload of non-files mode MoH, an unused thread was sticking around. 2) There was a race condition here where the application monitoring thread could still try to access the MoH class, even though the thread executing the MoH reload has already destroyed it. (issue #15109) Reported by: jvandal (issue #15123) Reported by: axisinternet (issue #15195) Reported by: amorsen (issue AST-208) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@201610 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-06-18Trunk implementation of setting an alternate RTP source.Mark Michelson
This contains the interface by which we can let an rtp instance know that it might start receiving audio from a new source. This is similar in nature to revision 197588 of Asterisk 1.4. Review: https://reviewboard.asterisk.org/r/276 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@201583 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-06-16Show the interface name on error, if it is not found.Eliel C. Sardanons
If the smdiport specified is not found, show the interface name instead of '(null)'. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@200841 65c4cc65-6c06-0410-ace0-fbb531ad65f3
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
2009-06-15Redesigned '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
2009-06-09module 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
2009-06-06Move 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
2009-06-04Move 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
2009-06-01Moved more static documentation to the new AstXML form.Eliel C. Sardanons
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
2009-06-01Move JabberSend manager action from static docs to the AstXML form.Eliel C. Sardanons
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@198565 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
2009-05-31Avoid a crash when res_timing_dahdi is unloaded but wasn't properly loaded.Eliel C. Sardanons
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
2009-05-30Properly terminate the receive buffer before sending to iksemel.Sean Bright
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
2009-05-30Merged revisions 198370 via svnmerge from Sean Bright
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
2009-05-30Merged revisions 198311 via svnmerge from Russell Bryant
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
2009-05-29Improve handling of trying to ACK too many timer expirations.Russell Bryant
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@198183 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-05-29Add a couple of TODO items so I don't forgetTerry Wilson
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@198182 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-05-29Resolve issues with choppy sound when using res_timing_pthread.Russell Bryant
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
2009-05-29Trim trailing whitespace so that I can work on this bug without it bothering ↵Russell Bryant
me. :-) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@197960 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-05-28Add Calendaring support for AsteriskTerry Wilson
This commit add Calendaring support to Asterisk for iCalendar, CalDAV, and MS Exchange calendars. Exchange support has only been tested on Exchange Server 2k3 and does not support forms-based authentication at this time (patches *very* welcome). Exchange support is also currently missing the ability to return a list of a meting's attendees (again, patches are very, very welcome). Features include: Querying a calendar for events over a specific time range Checking a calendar's busy status via the dialplan Writing calendar events via the dialplan (CalDAV and Exchange only) Handling calendar event notifications through the dialplan (closes issue #14771) Tested by: lmadsen, twilson, Shivaprakash Review: https://reviewboard.asterisk.org/r/58 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@197738 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-05-26Merged revisions 196826 via svnmerge from Russell Bryant
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r196826 | russell | 2009-05-26 13:14:36 -0500 (Tue, 26 May 2009) | 9 lines Resolve a file handle leak. The frames here should have always been freed. However, out of luck, there was never any memory leaked. However, after file streams became reference counted, this code would leak the file stream for the file being read. (closes issue #15181) Reported by: jkroon ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@196843 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-05-26Add new ast_complete_applications function so that we can use it with theSean Bright
'channel originate ... application <app>' CLI command. (And yeah, I cleaned up some whitespace in res_clioriginate.c... big whoop, wanna fight about it!?) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@196758 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
2009-05-22Fix build under dev mode and remove some casts that are no longer necessary asSean Bright
a result of the const-ify the world patch. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@196227 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
2009-05-18Recorded merge of revisions 195366 via svnmerge from Tilghman Lesher
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r195366 | tilghman | 2009-05-18 15:24:13 -0500 (Mon, 18 May 2009) | 8 lines Add a similar dependency on SMDI for voicemail as already exists for ADSI. (closes issue #14846) Reported by: pj Patches: 20090413__bug14846__1.4.diff.txt uploaded by tilghman (license 14) 20090507__issue14846__1.6.0.diff.txt uploaded by tilghman (license 14) 20090507__issue14846__1.6.1.diff.txt uploaded by tilghman (license 14) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@195370 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
2009-05-13Merged revisions 194208 via svnmerge from Joshua Colp
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r194208 | file | 2009-05-13 10:38:01 -0300 (Wed, 13 May 2009) | 11 lines Fix RFC2833 issues with DTMF getting duplicated and with duration wrapping over. (closes issue #14815) Reported by: geoff2010 Patches: v1-14815.patch uploaded by dimas (license 88) Tested by: geoff2010, file, dimas, ZX81, moliveras (closes issue #14460) Reported by: moliveras Tested by: moliveras ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@194209 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-05-12add 'const' qualifiers in various places where they should have beenKevin P. Fleming
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@193832 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-05-11Fix some timer state corruption.Russell Bryant
In res_timer_timerfd, handle the case that set_rate gets called while a timer is still in continuous mode. In this case, we want to remember the configured rate, but not actually set it until continuous mode has been disabled. Thanks to dvossel for finding and helping to debug the problem. (closes issue #15080) Reported by: dvossel Tested by: dvossel git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@193718 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-05-06Make the code that prevents an infinite loop from happening into a case ↵Joshua Colp
insensitive check. (thanks eliel) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@192736 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-05-06Fix an infinite loop with tab completion of CLI aliases that reference ↵Joshua Colp
themselves. (closes issue #15020) Reported by: junky git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@192700 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
2009-04-29fix typosJeff Peeler
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@191213 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
2009-04-27Merged revisions 190661-190662 via svnmerge from Russell Bryant
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r190661 | russell | 2009-04-27 14:00:54 -0500 (Mon, 27 Apr 2009) | 9 lines Resolve a crash in res_smdi when used with chan_dahdi. When chan_dahdi goes to get an SMDI message, it provides no search criteria. It just grabs the next message that arrives. This code was written with the SMDI dialplan functions in mind, since that is now the preferred method of using SMDI. However, this broke support of it being used from chan_dahdi. (closes AST-212) ........ r190662 | russell | 2009-04-27 14:03:59 -0500 (Mon, 27 Apr 2009) | 2 lines Fix a typo from 190661. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@190663 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
2009-04-23Support HTTP digest authentication for the http manager interface.Tilghman Lesher
(closes issue #10961) Reported by: ys Patches: digest_auth_r148468_v5.diff uploaded by ys (license 281) SVN branch http://svn.digium.com/svn/asterisk/team/group/manager_http_auth Tested by: ys, twilson, tilghman Review: http://reviewboard.digium.com/r/223/ Reviewed by: tilghman,russellb,mmichelson git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@190349 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-04-20Merged revisions 189462 via svnmerge from Sean Bright
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r189462 | seanbright | 2009-04-20 16:58:39 -0400 (Mon, 20 Apr 2009) | 13 lines Properly handle @s within hints in AEL. AEL was not handling the case of a device hint containing an @ symbol, which caused parking hints (e.g. hint(park:exten@context)) to error out the parser. This patch makes AEL treat the @ the same way it treats colon and ampersand now, meaning the characters are included in verbatim. (closes issue #14941) Reported by: bpgoldsb Patches: bug14941.patch uploaded by seanbright (license 71) Tested by: bpgoldsb ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@189464 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-04-14Fix an incorrect clock rate when sending T140 text.Joshua Colp
(closes issue #14029) Reported by: epicac git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@188413 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2009-04-13Fix another crash related to cached realtime music on hold.Mark Michelson
This was another off-by-one problem caused by moh_register. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@188102 65c4cc65-6c06-0410-ace0-fbb531ad65f3
generated by cgit v1.2.3 (git 2.39.1) at 2024-07-05 16:49:21 +0000