asterisk

Commit Graph

Author	SHA1	Message	Date
Joshua Colp	e85296e244	Add support for allowing an RTP engine to decide on whether it is possible for specific formats to be transcoded for an RTP instance. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@201902 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	dce6a54a4a	Trunk implementation of setting an alternate RTP source. 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	17 years ago
Kevin P. Fleming	962bd7ab26	Merged revisions 201261 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r201261 \| kpfleming \| 2009-06-17 07:03:25 -0500 (Wed, 17 Jun 2009) \| 9 lines Correct AST_LIST_APPEND_LIST behavior when list to be appended is empty. When the list to be appended is empty, and the list to be appended to is not, AST_LIST_APPEND_LIST would actually cause the target list to become broken, and no longer have a pointer to its last entry. This patch fixes the problem. (reported by Stanislaw Pitucha on the asterisk-dev mailing list) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@201262 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Kevin P. Fleming	f1dc620467	Enable applications to enable/disable digit and tone detection. Some applications (notably app_fax) do not need digit detection nor FAX tone detection while they are running, and if Asterisk is using software DSPs to provide the detection, this consumes extra CPU cycles that could be better spent on the actual application. This patch allows applications to query and control the state of digit and tone detection on a channel, and modifies app_fax to disable them while the FAX operations are occurring (and re-enable digit detection afterwards). git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@201139 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Kevin P. Fleming	4c0265664e	Merged revisions 200991 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r200991 \| kpfleming \| 2009-06-16 12:05:38 -0500 (Tue, 16 Jun 2009) \| 11 lines Improve support for media paths that can generate multiple frames at once. There are various media paths in Asterisk (codec translators and UDPTL, primarily) that can generate more than one frame to be generated when the application calling them expects only a single frame. This patch addresses a number of those cases, at least the primary ones to solve the known problems. In addition it removes the broken TRACE_FRAMES support, fixes a number of bugs in various frame-related API functions, and cleans up various code paths affected by these changes. https://reviewboard.asterisk.org/r/175/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@201056 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Kevin P. Fleming	aaeec3b40f	Last batch of 'static' qualifiers for module-level global variables. Fix up modules in the 'apps' directory, and also correct the bad example of enum definitions in include/asterisk/app.h, which many developers followed (thanks for reading the documentation!). In addition, add some basic usage examples of the 'pahole' and 'pglobal' tools to the coding guidelines. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@200656 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Kevin P. Fleming	82fb56886e	More 'static' qualifiers on module global variables. The 'pglobal' tool is quite handy indeed :-) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@200620 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Kevin P. Fleming	6c5987811c	Redesigned 'optional API' support. 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	17 years ago
Sean Bright	2cec55f038	Merged revisions 199856 via svnmerge from 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	17 years ago
David Vossel	d532cbcd8a	module load priority 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	17 years ago
Sean Bright	8ceb6c5d20	Merged revisions 199626,199628 via svnmerge from 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	17 years ago
David Vossel	d1d9beadc9	Merged revisions 199297 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r199297 \| dvossel \| 2009-06-05 16:19:56 -0500 (Fri, 05 Jun 2009) \| 14 lines Fixes issue with hints giving unexpected results. Hints with two or more devices that include ONHOLD gave unexpected results. (closes issue #15057) Reported by: p_lindheimer Patches: onhold_trunk.diff uploaded by dvossel (license 671) pbx.c.1.4.patch uploaded by p (license 558) devicestate.c.trunk.patch uploaded by p (license 671) Tested by: p_lindheimer, dvossel Review: https://reviewboard.asterisk.org/r/254/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@199298 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Sean Bright	befad10893	Merged revisions 199022 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r199022 \| seanbright \| 2009-06-04 10:14:57 -0400 (Thu, 04 Jun 2009) \| 40 lines Safely handle AMI connections/reload requests that occur during startup. During asterisk startup, a lock on the list of modules is obtained by the primary thread while each module is initialized. Issue 13778 pointed out a problem with this approach, however. Because the AMI is loaded before other modules, it is possible for a module reload to be issued by a connected client (via Action: Command), causing a deadlock. The resolution for 13778 was to move initialization of the manager to happen after the other modules had already been lodaded. While this fixed this particular issue, it caused a problem for users (like FreePBX) who call AMI scripts via an #exec in a configuration file (See issue 15189). The solution I have come up with is to defer any reload requests that come in until after the server is fully booted. When a call comes in to ast_module_reload (from wherever) before we are fully booted, the request is added to a queue of pending requests. Once we are done booting up, we then execute these deferred requests in turn. Note that I have tried to make this a bit more intelligent in that it will not queue up more than 1 request for the same module to be reloaded, and if a general reload request comes in ('module reload') the queue is flushed and we only issue a single deferred reload for the entire system. As for how this will impact existing installations - Before 13778, a reload issued before module initialization was completed would result in a deadlock. After 13778, you simply couldn't connect to the manager during startup (which causes problems with #exec-that-calls-AMI configuration files). I believe this is a good general purpose solution that won't negatively impact existing installations. (closes issue #15189) (closes issue #13778) Reported by: p_lindheimer Patches: 06032009_15189_deferred_reloads.diff uploaded by seanbright (license 71) Tested by: p_lindheimer, seanbright Review: https://reviewboard.asterisk.org/r/272/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@199051 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
David Vossel	3830c415c7	Generic call forward api, ast_call_forward() The function ast_call_forward() forwards a call to an extension specified in an ast_channel's call_forward string. After an ast_channel is called, if the channel's call_forward string is set this function can be used to forward the call to a new channel and terminate the original one. I have included this api call in both channel.c's ast_request_and_dial() and feature.c's feature_request_and_dial(). App_dial and app_queue already contain call forward logic specific for their application and options. (closes issue #13630) Reported by: festr Review: https://reviewboard.asterisk.org/r/271/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@198856 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	298d745fb4	Add the ability to execute connected line interception macros. When connected line updates are received or generated in the middle of an application call, it is now possible to execute a macro to manipulate the connected line data. This way, phone numbers may be manipulated to be more presentable to users, names may be changed for...whatever reason, or whatever else needs to be done may be. Review: https://reviewboard.asterisk.org/r/256 AST-165 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@198727 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	8da5e991ee	Minor whitespace fix. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@198670 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	8580871fd4	Constify the ast_frame arg to ast_queue_frame(). git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@198434 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Matthew Nicholson	c8b0c41ed8	Merged revisions 198068 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r198068 \| mnicholson \| 2009-05-29 13:53:01 -0500 (Fri, 29 May 2009) \| 15 lines Use AST_CDR_NOANSWER instead of AST_CDR_NULL as the default CDR disposition. This change also involves the addition of an AST_CDR_FLAG_ORIGINATED flag that is used on originated channels to distinguish: them from dialed channels. (closes issue #12946) Reported by: meral Patches: null-cdr2.diff uploaded by mnicholson (license 96) Tested by: mnicholson, dbrooks (closes issue #15122) Reported by: sum Tested by: sum ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@198072 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Sean Bright	c772d5a0e6	Update references to downloads.digium.com to its new URL. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@197861 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Sean Bright	f51bb019bb	Update references to bugs.digium.com and reviewboard.digium.com to the new URLs. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@197824 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Terry Wilson	71a3a2ebf6	Add Calendaring support for Asterisk 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	17 years ago
Mark Michelson	a7fd763ecc	Merged revisions 197537 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r197537 \| mmichelson \| 2009-05-28 09:49:13 -0500 (Thu, 28 May 2009) \| 21 lines Add flags to chanspy audiohook so that audio stays in sync. There are two flags being added to the chanspy audiohook here. One is the pre-existing AST_AUDIOHOOK_TRIGGER_SYNC flag. With this set, we ensure that the read and write slinfactories on the audiohook do not skew beyond a certain tolerance. In addition, there is a new audiohook flag added here, AST_AUDIOHOOK_SMALL_QUEUE. With this flag set, we do not allow for a slinfactory to build up a substantial amount of audio before flushing it. For this particular issue, this means that the person spying on the call will hear the conversations in real time with very little delay in the audio. (closes issue #13745) Reported by: geoffs Patches: 13745.patch uploaded by mmichelson (license 60) Tested by: snblitz ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@197543 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Kevin P. Fleming	1a02e34ccf	Ensure that this header includes xmldoc.h, since it depends on it. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@197335 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	0e62eddb93	Update configure script to check for OSP toolkit 3.5.0. (closes issue #14988) Reported by: tzafrir Patches: configure.ac.diff uploaded by homesick (license 91) new_ast_check_osptk.m4 uploaded by homesick (license 91) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@196946 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Sean Bright	3abe8a963e	Add new ast_complete_applications function so that we can use it with the '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	17 years ago
Kevin P. Fleming	57eedf97d0	Correct example for CLI autocompletion (generation) Reported by Atis on #asterisk-dev git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@196488 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Eliel C. Sardanons	2c882626a0	Implement a new element in AstXML for AMI actions documentation. 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	17 years ago
Kevin P. Fleming	e6b2e9a750	Const-ify the world (or at least a good part of it) 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	17 years ago
Matthew Nicholson	d02ad6b5f7	Merged revisions 195881 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r195881 \| mnicholson \| 2009-05-21 10:25:50 -0500 (Thu, 21 May 2009) \| 13 lines This commit prevents cdr records with AST_CDR_FLAG_ANSLOCKED and AST_CDR_FLAG_LOCKED from being updated in certain cases. This is accomplished by adding two functions to update the answer time and disposition of calls that checks for the proper lock flags. These functions are used in the ast_bridge_call() function so that ForkCDR(A) calls are respected. This patch also modifies the way ast_bridge_call() chooses the cdr record to base the bridged_cdr on. Previously the first unlocked cdr record would be chosen, now instead the first cdr record is chosen and forked cdr records are moved to the bridge_cdr. This allows the original cdr record and any forked cdr records to be properly updated with answer and end times. (closes issue #13797) Reported by: sh0t Tested by: sh0t (closes issue #14744) Reported by: deepesh ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@195882 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	bdcafc1ab4	Recorded merge of revisions 195366 via svnmerge from 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	17 years ago
Kevin P. Fleming	d1e0b11343	Add ability for modules to dynamically register logger levels This patch adds the ability for modules to dynamically create logger levels for their own use; these are named levels just like the built-in levels, and can be directed to any destination that the logger can send any level to, by including their names in logger.conf. Review: https://reviewboard.asterisk.org/r/244/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@194610 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	9c16774cc2	Minor documentation update for ast_event_queue(). git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@193461 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Kevin P. Fleming	2746f589b7	Add a more efficient way of allocating structures that use stringfields This commit adds an API call that can be used to allocate a structure along with this stringfield storage in a single allocation. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@192362 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Kevin P. Fleming	1f49e675bb	Correct some flaws in the memory accounting code for stringfields and ao2 objects Under some conditions, the memory allocation for stringfields and ao2 objects would not have supplied valid file/function names for MALLOC_DEBUG tracking, so this commit corrects that. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@192357 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Kevin P. Fleming	ec5116f80c	Properly account for memory allocated for channels and datastores As in previous commits, when channels are allocated (with ast_channel_alloc) or datastores are allocated (with ast_datastore_alloc) properly account for the memory being owned by the caller, instead of the allocator function itself. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@192318 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Kevin P. Fleming	d8182202ef	Ensure that string pools allocated to hold stringfields are properly accounted in MALLOC_DEBUG mode This commit modifies the stringfield pool allocator to remember the 'owner' of the stringfield manager the pool is being allocated for, and ensures that pools allocated in the future when fields are populated are owned by that file/function. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@192279 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	e0aba74fa9	Restore 'asyncagi break' command to 1.6.1 and higher. (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	17 years ago
Kevin P. Fleming	1726475a54	Ensure that astobj2 memory allocations are properly accounted for when MALLOC_DEBUG is used This commit ensures that all astobj2 allocated objects are properly accounted for in MALLOC_DEBUG mode by passing down the file/function/line information from the module/function that actually called the astobj2 allocation function. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@192059 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Kevin P. Fleming	73743b77b0	Add 'bitflags'-style information elements to event framework This patch add a new payload type for information elements, a set of bit flags. The payload is transported as a 32-bit unsigned integer but when matching is performed between events and subscribers, the matching is done by using a bitwise AND instead of numeric value comparison. Review: http://reviewboard.asterisk.org/r/242/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@191919 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Kevin P. Fleming	a3af213506	Remove rarely-used event_log/LOG_EVENT support In discussions today at the Europe Asterisk Developer Meet-Up, we determined that the event_log was used in only 9 places in the entire tree, and really was not needed at all. The users have been converted to use LOG_NOTICE, or the messages have been removed since other messages were already in place that provided the same information. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@191785 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Kevin P. Fleming	d9d2779008	Add buffer and echo canceller control to CHANNEL() dialplan function for DAHDI channels Adds ability for CHANNEL() dialplan function, when used on DAHDI channels, to temporarily change the number of buffers and/or the buffer policy, and also to enable, disable, or switch the echo canceller between FAX/data and voice modes. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@191411 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	91dde03ba8	Detect eaccess (or euidaccess) before using it. Reported by Andrew Lindh via the -dev list. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@191367 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
David Vossel	a6adc84e69	SIP option to specify outbound TLS/SSL client protocol. chan_sip allows for outbound TLS connections, but does not allow the user to specify what protocol to use (default was SSLv2, and still is if this new option is not specified). This patch lets the user pick the SSL/TLS client method for outbound connections in sip. (closes issue #14770) Reported by: TheOldSaint (closes issue #14768) Reported by: TheOldSaint Review: http://reviewboard.digium.com/r/240/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@191177 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	a866a75900	Merge str_substitution branch. 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	17 years ago
David Vossel	ca138fc807	Consistent SSL/TLS options across conf files ast_tls_read_conf() is a new api call for handling SSL/TLS options across all conf files. Before this change, SSL/TLS options were not consistent. http.conf and manager.conf required the 'ssl' prefix while sip.conf used options with the 'tls' prefix. While the options had different names in different conf files, they all did the exact same thing. Now, instead of mixing 'ssl' or 'tls' prefixes to do the same thing depending on what conf file you're in, all SSL/TLS options use the 'tls' prefix. For example. 'sslenable' in http.conf and manager.conf is now 'tlsenable' which matches what already existed in sip.conf. Since this has the potential to break backwards compatibility, previous options containing the 'ssl' prefix still work, but they are no longer documented in the sample.conf files. The change is noted in the CHANGES file though. Review: http://reviewboard.digium.com/r/237/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@191028 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	2c1ffef923	Resolve Solaris build issues and add some API documentation. (issue #14981) Reported by: snuffy git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@190989 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Kevin P. Fleming	a9657a86bf	Merged revisions 190721 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r190721 \| kpfleming \| 2009-04-27 14:29:46 -0500 (Mon, 27 Apr 2009) \| 7 lines Fix 'inconsistent line endings' when autoconf 2.63 is used Attempt to make configure script regeneration 'safe' using autoconf 2.63, which embeds a bare CR into the script, thus making Subversion complain about inconsistent line endings This commit changes the MIME type of the configure script to be 'binary' thus making Subversion no longer inspect line endings, and as a bonus 'svn diff' will no longer try to generate diff output for it, which is not generally useful anyway. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@190725 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
David Vossel	8f0b88c8c8	TLS/SSL private key option Adds option to specify a private key .pem file when configuring TLS or SSL in AMI, HTTP, and SIP. Before this, the certificate file was used for both the public and private key. It is possible for this file to hold both, but most configurations allow for a separate private key file to be specified. Clarified in .conf files how these options are to be used. The current conf files do not explain how the private key is handled at all, so without knowledge of Asterisk's TLS implementation, it would be hard to know for sure what was going on or how to set it up. Review: http://reviewboard.digium.com/r/234/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@190545 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Richard Mudgett	014aa91b84	Update comment. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@190516 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	f052718a80	Add \since tag for new API calls. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@190484 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago

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1948 Commits (eccfeaf9b47b86d9440bd79f16d790ca050dbaba)