111 Commits (cafc1158961f28ad54ebcdc5e042bf8094cbc620)

Author	SHA1	Message	Date
Matthew Jordan	cafc115896	A great big renaming patch ... This patch renames the bridging* files to bridge*. This may seem pedantic and silly, but it fits better in line with current Asterisk naming conventions: * channel is not "channeling" * monitor is not "monitoring" etc. A bridge is an object. It is a first class citizen in Asterisk. "Bridging" is the act of using a bridge on a set of channels - and the API that fulfills that role is more than just the action. (closes issue ASTERISK-22130) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@395378 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	d0a55fa52d	Refactor RTCP events over to Stasis; associate with channels ... This patch does the following: * It merges Jaco Kroon's patch from ASTERISK-20754, which provides channel information in the RTCP events. Because Stasis provides a cache, Jaco's patch was modified to pass the channel uniqueid to the RTP layer as opposed to a pointer to the channel. This has the following benefits: (1) It keeps the RTP engine 'clean' of references back to channels (2) It prevents circular dependencies and other potential ref counting issues * The RTP engine now allows any RTP implementation to raise RTCP messages. Potentially, other implementations (such as res_rtp_multicast) could also raise RTCP information. The engine provides structs to represent RTCP headers and RTCP SR/RR reports. * Some general refactoring in res_rtp_asterisk was done to try and tame the RTCP code. It isn't perfect - that's *way* beyond the scope of this work - but it does feel marginally better. * A few random bugs were fixed in the RTCP statistics. (Example: performing an assignment of a = a is probably not correct) * We now raise RTCP events for each SR/RR sent/received. Previously we wouldn't raise an event when we sent a RR report. Note that this work will be of use to others who want to monitor call quality or build modules that report call quality statistics. Since the events are now moving across the Stasis message bus, this is far easier to accomplish. It is also a first step (though by no means the last step) towards getting Olle's pinefrog work incorporated. Again: note that the patch by Jaco Kroon was modified slightly for this work; however, he did all of the hard work in finding the right places to set the channel in the RTP engine across the channel drivers. Much thanks goes to Jaco for his hard work here. Review: https://reviewboard.asterisk.org/r/2603/ (closes issue ASTERISK-20574) Reported by: Jaco Kroon patches: asterisk-rtcp-channel.patch uploaded by jkroon (License 5671) (closes issue ASTERISK-21471) Reported by: Matt Jordan git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@393740 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Igor Goncharovskiy	f7624718f8	Fix issue with inability to cancell call transfer made by on-sceen menus. ...	12 years ago
Mark Michelson	e3a89a0a18	Change chan_unistim to use core transfer API. ... Review: https://reviewboard.asterisk.org/r/2553 (closes issue ASTERISK-21527) Reported by Matt Jordan git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@392436 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Matthew Jordan	6258bbe7bd	Update Asterisk's CDRs for the new bridging framework ... This patch is the initial push to update Asterisk's CDR engine for the new bridging framework. This patch guts the existing CDR engine and builds the new on top of messages coming across Stasis. As changes in channel state and bridge state are detected, CDRs are built and dispatched accordingly. This fundamentally changes CDRs in a few ways. (1) CDRs are now *very* reflective of the actual state of channels and bridges. This means CDRs track well with what an actual channel is doing - which is useful in transfer scenarios (which were previously difficult to pin down). It does, however, mean that CDRs cannot be 'fooled'. Previous behavior in Asterisk allowed for CDR applications, channels, and other properties to be spoofed in parts of the code - this no longer works. (2) CDRs have defined behavior in multi-party scenarios. This behavior will not be what everyone wants, but it is a defined behavior and as such, it is predictable. (3) The CDR manipulation functions and applications have been overhauled. Major changes have been made to ResetCDR and ForkCDR in particular. Many of the options for these two applications no longer made any sense with the new framework and the (slightly) more immutable nature of CDRs. There are a plethora of other changes. For a full description of CDR behavior, see the CDR specification on the Asterisk wiki. (closes issue ASTERISK-21196) Review: https://reviewboard.asterisk.org/r/2486/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@391947 65c4cc65-6c06-0410-ace0-fbb531ad65f3	12 years ago
Igor Goncharovskiy	2053fc3159	Fix issue with no sound in both way in case of previous call to chan_unistim phone was canceled. ...	12 years ago
Mark Michelson	2dc8a06006	Refactor the features configuration scheme. ... Features configuration is handled in its own API in features_config.h and features_config.c. This way, features configuration is accessible to anything that needs it. In addition, features configuration has been altered to be more channel-oriented. Most callers of features API code will be supplying a channel so that the individual channel's settings will be acquired rather than the global setting. Missing from this commit is XML documentation for the features configuration. That will be handled in a separate commit. Review: https://reviewboard.asterisk.org/r/2578/ (issue ASTERISK-21542) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@390751 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Jason Parker	154fbf8cae	Split Hold event into Hold/Unhold, and move it into core. ... (closes issue ASTERISK-21487) Review: https://reviewboard.asterisk.org/r/2565/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@389746 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Matthew Jordan	06be8463b6	Migrate a large number of AMI events over to Stasis-Core ... This patch moves a number of AMI events over to the Stasis-Core message bus. This includes: * ChanSpyStart/Stop * MonitorStart/Stop * MusicOnHoldStart/Stop * FullyBooted/Reload * All Voicemail/MWI related events In addition, it adds some Stasis-Core and AMI support for generic AMI messages, refactors the message router in AMI to use a single router with topic forwarding for the topics that AMI cares about, and refactors MWI message types and topics to be more name compliant. Review: https://reviewboard.asterisk.org/r/2532 (closes issue ASTERISK-21462) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@389733 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Igor Goncharovskiy	1fb6f365ec	Fix several problems caused by multiple line usage with i2004 phones. ...	13 years ago
Richard Mudgett	3d63833bd6	Merge in the bridge_construction branch to make the system use the Bridging API. ... Breaks many things until they can be reworked. A partial list: chan_agent chan_dahdi, chan_misdn, chan_iax2 native bridging app_queue COLP updates DTMF attended transfers Protocol attended transfers git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@389378 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Richard Mudgett	5fec9b8d1f	Remove some unnecessary calls to ast_bridged_channel() in chan_unistim.c ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@387185 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Kinsey Moore	1a2a4578d2	Convert MWI state message type to the new stasis naming convention ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@384219 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Kinsey Moore	99aa02d17f	Transition MWI to Stasis-core ... Remove MWI's dependency on the event system by moving it to Stasis-core. This also introduces forwarding topic pools in Stasis-core which aggregate many dynamically allocated topics into a single primary topic. Review: https://reviewboard.asterisk.org/r/2368/ (closes issue ASTERISK-21097) Patch-by: Kinsey Moore git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@383284 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Igor Goncharovskiy	ef64b29f8b	Fix core dump on CLI usage ... Fix issue with 'unistim show info' CLI command when device connected not configured ........ Merged revisions 382827 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@382828 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Igor Goncharovskiy	469ca1c71d	Fix several unreleased mutex locks that cause problem with processing calls ...	13 years ago
Igor Goncharovskiy	8c99bcc5a3	Add firmware information to CLI devices listing ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@377595 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Igor Goncharovskiy	98539ffb32	Fix codec mismatch ... Fix code to send in both rx and tx open stream messages correct codecs. Found that on phase 0/1 phones wrong codecs cause to no audio in some situations. (issue ASTERISK-20183) ........ Merged revisions 377591 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 377592 from http://svn.asterisk.org/svn/asterisk/branches/10 ........ Merged revisions 377593 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@377594 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Igor Goncharovskiy	1042d43160	Remove trailing whitespaces in number from incoming redial list. ... Reported by: Igor Olhovskiy ........ Merged revisions 377577 from http://svn.asterisk.org/svn/asterisk/branches/11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@377579 65c4cc65-6c06-0410-ace0-fbb531ad65f3	13 years ago
Igor Goncharovskiy	e41a591dfc	Fix underscreen buttons warnings apeared while transfer process ...	13 years ago
Igor Goncharovskiy	8eaba809ab	Remove code, that operate with cdr in attempt_transfer(). That was removed somewhere between 1.2 and 1.4 and acidentaly put back in chan_unistim. ...	13 years ago
Igor Goncharovskiy	9278b5e51e	Added option 'interdigit_timer' to unistim.conf to make able controll hardcoded dial timeout constant. ...	13 years ago
Igor Goncharovskiy	f9c3585d73	Deactivate timer for dialing entered number on hook switch hang up. ...	13 years ago
Igor Goncharovskiy	95ac8f4743	Add French translation for chan_unistim phones on-screen menus. ...	13 years ago
Igor Goncharovskiy	4ca35e0907	Fix MWI update so LED display correct voicemail state after phone usage. Also fixes few warnings. ...	14 years ago
Kinsey Moore	571445ab9c	Convert AST_FLAG_ANSWERED_ELSEWHERE usage to AST_CAUSE_ANSWERED_ELSEWHERE ... This was essentially duplicated functionality where normal channels used AST_CAUSE_ANSWERED_ELSEWHERE while local channels and queues used AST_FLAG_ANSWERED_ELSEWHERE. This removes the flag and converts that usage into AST_CAUSE_ANSWERED_ELSEWHER usage. Review: https://reviewboard.asterisk.org/r/1944 (closes issue ASTERISK-19865) Patch-by: Birger Harzenetter git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@368519 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Kinsey Moore	b5a6de76fc	Commit framework for HANGUPCAUSE (replacement for SIP_CAUSE) ... This is the starting point for the Asterisk 11: Who Hung Up work and provides a framework which will allow channel drivers to report the types of hangup cause information available in SIP_CAUSE without incurring the overhead of the MASTER_CHANNEL dialplan function. The initial implementation only includes cause generation for chan_sip and does not include cause code translation utilities. This change deprecates SIP_CAUSE and replaces its method of reporting cause codes with the new framework. This change also deprecates the 'storesipcause' option in sip.conf. Review: https://reviewboard.asterisk.org/r/1822/ (Closes issue SWP-4221) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@366408 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Matthew Jordan	f78290068a	Fix a variety of potential buffer overflows ... * chan_mobile: Fixed an overrun where the cind_state buffer (an integer array of size 16) would be overrun due to improper bounds checking. At worst, the buffer can be overrun by a total of 48 bytes (assuming 4-byte integers), which would still leave it within the allocated memory of struct hfp. This would corrupt other elements in that struct but not necessarily cause any further issues. * app_sms: The array imsg is of size 250, while the array (ud) that the data is copied into is of size 160. If the size of the inbound message is greater then 160, up to 90 bytes could be overrun in ud. This would corrupt the user data header (array udh) adjacent to ud. * chan_unistim: A number of invalid memmoves are corrected. These would move data (which may or may not be valid) into the ends of these buffers. * asterisk: ast_console_toggle_loglevel does not check that the console log level being set is less then or equal to the allowed log levels of 32. * format_pref: In ast_codec_pref_prepend, if any occurrence of the specified codec is not found, the value used to index into the array pref->order would be one greater then the maximum size of the array. * jitterbuf: If the element being placed into the jitter buffer lands in the last available slot in the jitter history buffer, the insertion sort attempts to move the last entry in the buffer into one slot past the maximum length of the buffer. Note that this occurred for both the min and max jitter history buffers. * tdd: If a read from fsk_serial returns a character that is greater then 32, an attempt to read past one of the statically defined arrays containing the values that character maps to would occur. * localtime: struct ast_time and tm are not the same size - ast_time is larger, although it contains the elements of tm within it in the same layout. Hence, when using memcpy to copy the contents of tm into ast_time, the size of tm should be used, as opposed to the size of ast_time. * extconf: this treats ast_timing's minmask array as if it had a length of 48, when it has defined the size of the array as 24. pbx.h defines minmask as having a size of 48. (issue ASTERISK-19668) Reported by: Matt Jordan ........ Merged revisions 362485 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 362496 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@362497 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Walter Doekes	fc63e07135	Avoid cppcheck warnings; removing unused vars and a bit of cleanup. ... Patch by: junky Review: https://reviewboard.asterisk.org/r/1743/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@362307 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Kinsey Moore	a485f44022	Add missing newlines to CLI logging ... ........ Merged revisions 361471 from http://svn.asterisk.org/svn/asterisk/branches/1.8 ........ Merged revisions 361472 from http://svn.asterisk.org/svn/asterisk/branches/10 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@361476 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Terry Wilson	786f5898d1	Finalize ast_channel opaquification ... Review: https://reviewboard.asterisk.org/r/1786/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@358907 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Igor Goncharovskiy	c369a4416b	Massive changes in chan_unistim channel driver. Include many fixes in channel driver operation and add additional functionality: ...	14 years ago
Terry Wilson	0e5c761c28	Opaquify ast_channel typedefs, fd arrays, and softhangup flag ... Review: https://reviewboard.asterisk.org/r/1784/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@357721 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Terry Wilson	a9d607a357	Opaquify ast_channel structs and lists ... Review: https://reviewboard.asterisk.org/r/1773/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@357542 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Sean Bright	51c24c88a1	Prefer ast_set_qos() over ast_netsock_set_qos() ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@356882 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Terry Wilson	ebaf59a656	Opaquification for ast_format structs in struct ast_channel ... Review: https://reviewboard.asterisk.org/r/1770/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@356573 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Terry Wilson	57f42bd74f	ast_channel opaquification of pointers and integral types ... Review: https://reviewboard.asterisk.org/r/1753/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@356042 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Terry Wilson	34c55e8e7c	Opaquify char * and char[] in ast_channel ... Review: https://reviewboard.asterisk.org/r/1733/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@354968 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Richard Mudgett	23bc964e1c	Constify some more channel driver technology callback parameters. ... Review: https://reviewboard.asterisk.org/r/1707/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@353685 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Terry Wilson	99cae5b750	Opaquify channel stringfields ... Continue channel opaque-ification by wrapping all of the stringfields. Eventually, we will restrict what can actually set these variables, but the purpose for now is to hide the implementation and keep people from adding code that directly accesses the channel structure. Semantic changes will follow afterward. Review: https://reviewboard.asterisk.org/r/1661/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@352348 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Terry Wilson	04da92c379	Replace direct access to channel name with accessor functions ... There are many benefits to making the ast_channel an opaque handle, from increasing maintainability to presenting ways to kill masquerades. This patch kicks things off by taking things a field at a time, renaming the field to '__do_not_use_${fieldname}' and then writing setters/getters and converting the existing code to using them. When all fields are done, we can move ast_channel to a C file from channel.h and lop off the '__do_not_use_'. This patch sets up main/channel_interal_api.c to be the only file that actually accesses the ast_channel's fields directly. The intent would be for any API functions in channel.c to use the accessor functions. No more monkeying around with channel internals. We should use our own APIs. The interesting changes in this patch are the addition of channel_internal_api.c, the moving of the AST_DATA stuff from channel.c to channel_internal_api.c (note: the AST_DATA stuff will have to be reworked to use accessor functions when ast_channel is really opaque), and some re-working of the way channel iterators/callbacks are handled so as to avoid creating fake ast_channels on the stack to pass in matching data by directly accessing fields (since "name" is a stringfield and the fake channel doesn't init the stringfields, you can't use the ast_channel_name_set() function). I went with ast_channel_name(chan) for a getter, and ast_channel_name_set(chan, name) for a setter. The majority of the grunt-work for this change was done by writing a semantic patch using Coccinelle ( http://coccinelle.lip6.fr/ ). Review: https://reviewboard.asterisk.org/r/1655/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@350223 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Tilghman Lesher	56b21b4683	Remove the few places where we try to ast_verbose() without a newline. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@346655 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Igor Goncharovskiy	7e5ce2ac49	Merged revisions 339942 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/10 ................ r339942 | igorg | 2011-10-09 08:18:02 +0700 (Вск, 09 Окт 2011) | 12 lines Merged revisions 339938 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r339938 | igorg | 2011-10-09 08:16:09 +0700 (Вск, 09 Окт 2011) | 6 lines Fix compilation issue, caused by missed session structure (closes issue ASTERISK-18694) Reported by: alex70 ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@339947 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Igor Goncharovskiy	326c3a39d5	Merged revisions 339885 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/10 ................ r339885 | igorg | 2011-10-08 22:46:27 +0700 (Сбт, 08 Окт 2011) | 13 lines Merged revisions 339884 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r339884 | igorg | 2011-10-08 22:45:20 +0700 (Сбт, 08 Окт 2011) | 7 lines Fix segfault in Unistim channel (closes issue ASTERISK-18638) Reported by: jonnt ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@339886 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Igor Goncharovskiy	a01b34f488	Merged revisions 339831 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/10 ................ r339831 | igorg | 2011-10-08 22:01:35 +0700 (Сбт, 08 Окт 2011) | 14 lines Merged revisions 339830 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r339830 | igorg | 2011-10-08 21:56:35 +0700 (Сбт, 08 Окт 2011) | 8 lines Fix char array cast as short array in send_client() function (for ARM platform) (closes issue ASTERISK-17314) Reported by: jjoshua ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@339832 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Matthew Jordan	8b5ba33fe0	Merged revisions 335078 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/10 ................ r335078 | mjordan | 2011-09-09 11:27:01 -0500 (Fri, 09 Sep 2011) | 29 lines Merged revisions 335064 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r335064 | mjordan | 2011-09-09 11:09:09 -0500 (Fri, 09 Sep 2011) | 23 lines Updated SIP 484 handling; added Incomplete control frame When a SIP phone uses the dial application and receives a 484 Address Incomplete response, if overlapped dialing is enabled for SIP, then the 484 Address Incomplete is forwarded back to the SIP phone and the HANGUPCAUSE channel variable is set to 28. Previously, the Incomplete application dialplan logic was automatically triggered; now, explicit dialplan usage of the application is required. Additionally, this patch adds a new AST_CONTOL_FRAME type called AST_CONTROL_INCOMPLETE. If a channel driver receives this control frame, it is an indication that the dialplan expects more digits back from the device. If the device supports overlap dialing it should attempt to notify the device that the dialplan is waiting for more digits; otherwise, it can handle the frame in a manner appropriate to the channel driver. (closes issue ASTERISK-17288) Reported by: Mikael Carlsson Tested by: Matthew Jordan Review: https://reviewboard.asterisk.org/r/1416/ ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@335079 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Leif Madsen	a525edea59	Merged revisions 328247 via svnmerge from ... 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: https://origsvn.digium.com/svn/asterisk/trunk@328259 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Russell Bryant	0938974902	Merged revisions 317478 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r317478 | russell | 2011-05-05 17:53:45 -0500 (Thu, 05 May 2011) | 12 lines Fix some consistency issues with jitterbuffer config. Store the defaults noted in the sample config files in the jitterbuffer config data structure. This makes the CLI commands that output these settings show the right thing. Also only show the settings that are relevant in the settings CLI commands, based on which jitterbuffer is selected and whether it's enabled. (closes issue #19083) Reported by: rgagnon Patches: issue-19083-trunk-r313139.diff uploaded by rgagnon (license 1202) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@317479 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Russell Bryant	37aa52fd78	Merged revisions 316265 via svnmerge from ... 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: https://origsvn.digium.com/svn/asterisk/trunk@316293 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Tzafrir Cohen	2b56cf085c	Merged revisions 314779 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r314779 | tzafrir | 2011-04-22 16:59:43 +0300 (ו', 22 אפר 2011) | 2 lines Fix a few typos (shown by Lintian) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@314824 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago