asterisk

Commit Graph

Author	SHA1	Message	Date
Tilghman Lesher	2302618bb7	Merged revisions 285373 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r285373 \| tilghman \| 2010-09-07 16:14:03 -0500 (Tue, 07 Sep 2010) \| 7 lines Add CHANNEL(checkhangup) to check whether a channel is in the process of being hanged up. (closes issue #17652) Reported by: kobaz Patches: func_channel.patch uploaded by kobaz (license 834) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@285374 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Terry Wilson	857814f435	Add SRTP support for Asterisk After 5 years in mantis and over a year on reviewboard, SRTP support is finally being comitted. This includes generic CHANNEL dialplan functions that work for getting the status of whether a call has secure media or signaling as defined by the underlying channel technology and for setting whether or not a new channel being bridged to a calling channel should have secure signaling or media. See doc/tex/secure-calls.tex for examples. Original patch by mikma, updated for trunk and revised by me. (closes issue #5413) Reported by: mikma Tested by: twilson, notthematrix, hemanshurpatel Review: https://reviewboard.asterisk.org/r/191/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@268894 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Mark Michelson	7acfebf2b8	Adjust XML for func_channel to indicate that rtpdest can take a "text" argument. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@250730 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Russell Bryant	34317fb0d3	Fix the ability to specify an OSP token for an outbound IAX2 call. When this patch was originally submitted, the code allowed for the token to be set via a channel variable. I decided that a cleaner approach would be to integrate it into the CHANNEL() function. Unfortunately, that is not a suitable approach. It's not possible to get the value set on the channel soon enough using that method. So, go back to the simple channel variable method. (closes issue #16711) Reported by: homesick Patches: iax-svn.diff uploaded by homesick (license 91) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@243482 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Kevin P. Fleming	cdd1f9e296	Finish implementaton of astobj2 OBJ_MULTIPLE, and convert ast_channel_iterator to use it. This patch finishes the implementation of OBJ_MULTIPLE in astobj2 (the case where multiple results need to be returned; OBJ_NODATA mode already was supported). In addition, it converts ast_channel_iterators (only the targeted versions, not the ones that iterate over all channels) to use this method. During this work, I removed the 'ao2_flags' arguments to the ast_channel_iterator constructor functions; there were no uses of that argument yet, there is only one possible flag to pass, and it made the iterators less 'opaque'. If at some point in the future someone really needs an ast_channel_iterator that does not lock the container, we can provide constructor(s) for that purpose. Review: https://reviewboard.asterisk.org/r/379/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@225244 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	2cfddf8cb6	Add MASTER_CHANNEL() dialplan function, as well as a useful usage. (closes issue #13140) Reported by: cpina Patches: 20090807__issue13140.diff.txt uploaded by tilghman (license 14) Tested by: lmadsen Change-type: feature git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@215301 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	642bec4d6f	AST-2009-005 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@211539 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Sean Bright	719917fe59	Support setting and receiving Reverse Charging Indication over ISDN PRI. This is a continuation of revision 885 to LibPRI (Capture and expose the Reverse Charging Indication IE on ISDN PRI) which added the ability to get/set Reverse Charging Indication in LibPRI. This patch adds the ability to specify RCI on the outbound leg of a PRI call from within Asterisk, by prefixing the dialed number with a capital 'C' like: ...,Dial(DAHDI/g1/C4445556666) And to read it off an inbound channel: exten => s,1,Set(RCI=${CHANNEL(reversecharge)}) Thanks again to rmudgett for the thorough review. (closes issue #13760) Reported by: mrgabu Review: https://reviewboard.asterisk.org/r/303/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@204749 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Russell Bryant	0264eef115	Merge the new Channel Event Logging (CEL) subsystem. 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	16 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	16 years ago
Leif Madsen	9408242796	Recorded merge of revisions 193544 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r193544 \| lmadsen \| 2009-05-11 13:35:17 -0400 (Mon, 11 May 2009) \| 7 lines Document CHANNEL(transfercapability) in CLI documentation. (issue #15073) Reported by: pkempgen Patches: 20090511__issue15073.diff.txt uploaded by tilghman (license 14) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@193545 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Russell Bryant	cba19c8a67	Convert the ast_channel data structure over to the astobj2 framework. 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	16 years ago
Russell Bryant	16fc1993ef	Add support for the "name" option in the CHANNEL() function. Review: http://reviewboard.digium.com/r/199/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@182762 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Russell Bryant	4ec301360c	Merge a large set of updates to the Asterisk indications API. This patch includes a number of changes to the indications API. The primary motivation for this work was to improve stability. The object management in this API was significantly flawed, and a number of trivial situations could cause crashes. The changes included are: 1) Remove the module res_indications. This included the critical functionality that actually loaded the indications configuration. I have seen many people have Asterisk problems because they accidentally did not have an indications.conf present and loaded. Now, this code is in the core, and Asterisk will fail to start without indications configuration. There was one part of res_indications, the dialplan applications, which did belong in a module, and have been moved to a new module, app_playtones. 2) Object management has been significantly changed. Tone zones are now managed using astobj2, and it is no longer possible to crash Asterisk by issuing a reload that destroys tone zones while they are in use. 3) The API documentation has been filled out. 4) The API has been updated to follow our naming conventions. 5) Various bits of code throughout the tree have been updated to account for the API update. 6) Configuration parsing has been mostly re-written. 7) "Code cleanup" The code is from svn/asterisk/team/russell/indications/. Review: http://reviewboard.digium.com/r/149/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@176627 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Russell Bryant	ef6ad2b53c	Merged revisions 168561 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r168561 \| russell \| 2009-01-13 13:13:05 -0600 (Tue, 13 Jan 2009) \| 2 lines Revert unnecessary indications API change from rev 122314 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@168562 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	5b168ee34b	Merge changes from team/group/appdocsxml This commit introduces the first phase of an effort to manage documentation of the interfaces in Asterisk in an XML format. Currently, a new format is available for applications and dialplan functions. A good number of conversions to the new format are also included. For more information, see the following message to asterisk-dev: http://lists.digium.com/pipermail/asterisk-dev/2008-October/034968.html git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@153365 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	b2ef18dab4	Add some more IAX2-specific information about the channel to the CHANNEL() function and begin the transition from SIPCHANINFO() to just using CHANNEL(). (closes issue #12856) Reported by: mostyn Patches: iax_and_sip_channel_info.patch uploaded by mostyn (license 398) (with some additional cleanup by me) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@122802 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Brett Bryant	c1451b5537	This patch adds more detailed statistics for RTP channels, and provides an API call to access it, including maximums, minimums, standard deviatinos, and normal deviations. Currently this is implemented for chan_sip, but could be added to the func_channel_read callbacks for the CHANNEL function for any channel that uses RTP. (closes issue #10590) Reported by: gasparz Patches: chan_sip_c.diff uploaded by gasparz (license 219) rtp_c.diff uploaded by gasparz (license 219) rtp_h.diff uploaded by gasparz (license 219) audioqos-trunk.diff uploaded by snuffy (license 35) rtpqos-trunk-r119891.diff uploaded by sergee (license 138) Tested by: jsmith, gasparz, snuffy, marsosa, chappell, sergee git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@120635 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	2f0abd23d2	Add a function, CHANNELS(), which retrieves a list of all active channels. (closes issue #11330) Reported by: rain Patches: func_channel-channel_list_function.diff uploaded by rain (license 327) (with some additional changes by me, mostly to meet coding guidelines) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@120230 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Jeff Peeler	41fd7a6a21	(closes issue #6113 ) Reported by: oej Tested by: jpeeler This patch implements multiple parking lots for parked calls. The default parkinglot is used by default, however setting the channel variable PARKINGLOT in the dialplan will allow use of any other configured parkinglot. See configs/features.conf.sample for more details on setting up another non-default parkinglot. Also, one can (currently) set the default parkinglot to use in the driver configuration file via the parkinglot option. Patch initially written by oej, brought up to date and finalized by mvanbaak, and then stabilized and converted to astobj2 by me. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@114487 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	26755e3882	Context tracing for channels (closes issue #11268) Reported by: moy Patches: chantrace-datastored-encapsulated-rev94934.patch uploaded by moy (license 222) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@103754 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Luigi Rizzo	9335ace850	another bunch of include removals (errno.h and asterisk/logger.h) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89425 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Luigi Rizzo	fdb7f7ba3d	Start untangling header inclusion in a way that does not affect build times - tested, there is no measureable difference before and after this commit. In this change: use asterisk/compat.h to include a small set of system headers: inttypes.h, unistd.h, stddef.h, stddint.h, sys/types.h, stdarg.h, stdlib.h, alloca.h, stdio.h Where available, the inclusion is conditional on HAVE_FOO_H as determined by autoconf. Normally, source files should not include any of the above system headers, and instead use either "asterisk.h" or "asterisk/compat.h" which does it better. For the time being I have left alone second-level directories (main/db1-ast, etc.). git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89333 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Tilghman Lesher	20bbd09de3	Mostly cleanup of documentation to substitute the pipe with the comma, but a few other formatting cleanups, too. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@77808 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Joshua Colp	1961b57705	Add rtpdest option to SIP CHANNEL() dialplan function to return the IP address and port that RTP (be it audio/video/text) is going to. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@71988 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Russell Bryant	b94378b5e6	Merge changes from team/russell/iax2_osp This set of changes adds OSP support to chan_iax2. However, I have modified the patch a bit from what was submitted. You now use the CHANNEL() function to get and set the OSP token for IAX2. (issue #8531, reported by and original patch by homesick, patch updated by me) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@61702 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Russell Bryant	7c884d76ea	Merged revisions 59256 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r59256 \| russell \| 2007-03-27 11:20:53 -0500 (Tue, 27 Mar 2007) \| 4 lines Convert the RTPQOS function to just be additional parameter of the CHANNEL function. This way, it will be possible for other RTP based channel drivers to expose this information in the future. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@59257 65c4cc65-6c06-0410-ace0-fbb531ad65f3	18 years ago
Olle Johansson	0ddb38ddee	Doxygen update git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@51920 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Kevin P. Fleming	37182c873e	finish const-ifying ast_func_read() git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@49741 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Luigi Rizzo	09f75aa6dc	rename the structs struct tone_zone_sound and struct tone_zone defined in indications.h to ind_tone_zone_sound and ind_tone_zone, to avoid conflicts with the structs with the same names defined in tonezone.h Hope i haven't missed any instance. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@48958 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Paul Cadach	fc58bec502	Merged revisions 44809 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r44809 \| pcadach \| 2006-10-10 23:44:54 +0700 (Втр, 10 Окт 2006) \| 1 line CHANNEL() function sometime mix parameter and value ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@47718 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Kevin P. Fleming	0a27d8bfe5	merge new_loader_completion branch, including (at least): - restructured build tree and makefiles to eliminate recursion problems - support for embedded modules - support for static builds - simpler cross-compilation support - simpler module/loader interface (no exported symbols) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@40722 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Kevin P. Fleming	472c1ca282	simplify autoconfig include mechanism (make tholo happy he can use lint again :-) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@32846 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Olle Johansson	7a60aff3b1	Make tonezone writeable in CHANNEL() (from my old func_tonezone.c) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@32018 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Russell Bryant	d2872c914d	- mark some applications deprecated that already have replacements - add BLACKLIST and mark LookupBlacklist deprecated - add transfercapability support to CHANNEL and mark SetTransferCapability deprecated (issue #7225, Corydon) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@30359 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Russell Bryant	932c5c7fd7	use the channel lock wrappers (issue #7120 , Mithraen) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@26528 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
BJ Weschke	a7b1476058	Provide the ability to adjust txgain/rxgain on a channel level via the CHANNEL() function git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@24621 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Luigi Rizzo	e43bc6634d	This rather large commit changes the way modules are loaded. As partly documented in loader.c and include/asterisk/module.h, modules are now expected to return all of their methods and flags into a structure 'mod_data', and are normally loaded with RTLD_NOW \| RTLD_LOCAL, so symbols are resolved immediately and conflicts should be less likely. Only in a small number of cases (res_*, typically) modules are loaded RTLD_GLOBAL, so they can export symbols. The core of the change is only the two files loader.c and include/asterisk/module.h, all the rest is simply adaptation of the existing modules to the new API, a rather mechanical (but believe me, time and finger-consuming!) process whose detail you can figure out by svn diff'ing any single module. Expect some minor compilation issue after this change, please report it on mantis http://bugs.digium.com/view.php?id=6968 so we collect all the feedback in one place. I am just sorry that this change missed SVN version number 20000! git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@20003 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Kevin P. Fleming	f10f427d49	since the module API is changing, it's a good time to const-ify the description() and key() return values git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@18552 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Tilghman Lesher	0af770edd1	Bug 6670 - Additional parameters to the CHANNEL func git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@14870 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Kevin P. Fleming	bb7325ade8	set keywords property correctly git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@10627 65c4cc65-6c06-0410-ace0-fbb531ad65f3	19 years ago
Kevin P. Fleming	96bf384e7d	major dialplan functions update deprecate LANGUAGE() and MUSICCLASS(), in favor of CHANNEL() git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@9674 65c4cc65-6c06-0410-ace0-fbb531ad65f3	20 years ago

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92 Commits (97a653518dc6dbdc927521e92b51dedaffe149bc)