555 Commits (5edeada22a25fdc19d7b3ef54a4d15c1d11f443f)

Author	SHA1	Message	Date
Andrew Latham	f9c3b26241	Add Function and Application Relationships to documentation ... Add and extend the see-also sections to the documentation for applications and functions in an effort to expand the online documentation of the wiki. Also check for and update any links to moved documentation in the doc folder. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@304913 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Matthew Nicholson	e706b5706e	According to section 19.1.2 of RFC 3261: ... For each component, the set of valid BNF expansions defines exactly which characters may appear unescaped. All other characters MUST be escaped. This patch modifies ast_uri_encode() to encode strings in line with this recommendation. This patch also adds an ast_escape_quoted() function which escapes '"' and '\' characters in quoted strings in accordance with section 25.1 of RFC 3261. The ast_uri_encode() function has also been modified to take an ast_flags struct describing the set of rules it should use when escaping characters to allow for it to escape SIP URIs in addition to HTTP URIs and other types of URIs or variations of those two URI types in the future. The ast_uri_decode() function has also been modified to accept an ast_flags struct describing the set of rules to use when decoding to enable decoding '+' as ' ' in legacy http URLs. The unit tests for these functions have also been updated. ABE-2705 Review: https://reviewboard.asterisk.org/r/1081/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@303509 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Tilghman Lesher	52dbebad8e	Add DB_KEYS. ... Discussion on #asterisk on 2011-01-19: (02:07:03 PM) boch: i wonder how to cycle all entries in a tree (02:07:11 PM) leifmadsen: use While() (02:07:17 PM) leifmadsen: you need to know the tree structure already though (02:07:36 PM) boch: what you mean? (02:09:02 PM) leifmadsen: you need to know the structure prior to looping, because you can't just return the structure from the dialplan (02:09:43 PM) leifmadsen: the only way I can think of doing that is via something like writing the output of: asterisk -rx "database show" to a file, then looping through that to know the structure of the database and check everything (02:09:59 PM) leifmadsen: but at that point you're better off just using either a relational database or an external script (02:10:13 PM) boch: for example i need to know all entries in the tree (02:10:15 PM) boch: got it (02:10:20 PM) leifmadsen: exactly (02:10:22 PM) leifmadsen: that's the problem (02:10:22 PM) boch: thank you (02:13:09 PM) mateu: yeah, i'm surprised there isn't something from the dialplan like 'database show family' so one can get all keys in a family to loop over. (02:15:35 PM) leifmadsen: database shows everything (02:16:22 PM) mateu: i mean something from the dial plan that mimics 'database show <family>' (02:16:41 PM) leifmadsen: guess no one has found that important enough to program :) (02:16:52 PM) leifmadsen: at that point you should probably just use a relational database... (02:17:10 PM) mateu: i dunno (02:17:16 PM) mateu: seems pretty basic to me. (02:17:16 PM) leifmadsen: me either (02:17:19 PM) leifmadsen: sure does (02:17:24 PM) leifmadsen: no one has programmed it though (02:17:28 PM) ***leifmadsen shrugs (02:17:43 PM) mateu: ok, well at least we know how it currently stands. thanks leifmadsen (02:28:52 PM) Corydon76-home: leifmadsen: something like HASHKEYS() ? (02:30:11 PM) leifmadsen: Corydon76-home: ummm, I was thinking more like DUNDI_QUERY() and DUNDI_RESULT() (02:30:31 PM) leifmadsen: although HASHKEYS() might work (02:30:58 PM) leifmadsen: actually ya, looking at it, similar to HASHKEYS() (02:31:01 PM) leifmadsen: DBKEYS() I guess? (02:31:45 PM) Corydon76-home: So with no argument, retrieves families, with an argument, retrieves keys of that family? (02:34:02 PM) leifmadsen: ya (02:34:16 PM) leifmadsen: how would you iterate through layers of them? (02:34:30 PM) leifmadsen: i.e. family/key/key/key ? (02:34:43 PM) Corydon76-home: Essentially, yes git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@303198 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Andrew Latham	7cb1c06dd3	Add relationships to function documentation. ... Fix amatuer type mistake git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@301850 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Andrew Latham	ca8a5498b1	Add relationships to function documentation. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@301846 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Tilghman Lesher	b2a70b4065	Oops, missed the actual decoding part. ... (closes issue #18046) Reported by: wdoekes git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@301008 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Tilghman Lesher	a58b2fb395	XML validation ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@300841 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Tilghman Lesher	473e176df8	Add a hashcompat mode called "legacy", which translates a literal plus sign to a space. ... (closes issue #18046) Reported by: wdoekes Patches: 20100930__issue18046.diff.txt uploaded by tilghman (license 14) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@300840 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Tilghman Lesher	ac77932bac	Merged revisions 298478 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r298478 | tilghman | 2010-12-16 02:56:13 -0600 (Thu, 16 Dec 2010) | 15 lines Merged revisions 298477 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r298477 | tilghman | 2010-12-16 02:54:23 -0600 (Thu, 16 Dec 2010) | 8 lines Eliminate duplicates from container. (closes issue #18091) Reported by: bunny Patches: 20101006__issue18091.diff.txt uploaded by tilghman (license 14) Tested by: bunny ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@298479 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Tilghman Lesher	53357354a4	Merged revisions 294989 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r294989 | tilghman | 2010-11-15 01:44:38 -0600 (Mon, 15 Nov 2010) | 15 lines Merged revisions 294988 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r294988 | tilghman | 2010-11-15 01:42:39 -0600 (Mon, 15 Nov 2010) | 8 lines It is possible to crash Asterisk by feeding the curl engine invalid data. (closes issue #18161) Reported by: wdoekes Patches: 20101029__issue18161.diff.txt uploaded by tilghman (license 14) Tested by: tilghman ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@294990 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Jeff Peeler	34c30c8ad3	Merged revisions 293159 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r293159 | jpeeler | 2010-10-28 11:11:08 -0500 (Thu, 28 Oct 2010) | 18 lines Merged revisions 293158 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r293158 | jpeeler | 2010-10-28 11:09:40 -0500 (Thu, 28 Oct 2010) | 11 lines Fix infinite loop in FILTER(). Specifically when you're using characters above \x7f or invalid character escapes (e.g. \xgg). (closes issue #18060) Reported by: wdoekes Patches: issue18060_func_strings_filter_infinite_loop.patch uploaded by wdoekes (license 717) Tested by: wdoekes ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@293160 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Tilghman Lesher	6d0e383321	Merged revisions 289543,289581 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r289543 | tilghman | 2010-09-30 12:50:52 -0500 (Thu, 30 Sep 2010) | 2 lines More Solaris compatibility fixes ........ r289581 | tilghman | 2010-09-30 15:23:10 -0500 (Thu, 30 Sep 2010) | 2 lines Solaris fixes. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@289588 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Tilghman Lesher	794ff358a3	Merged revisions 288713 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r288713 | tilghman | 2010-09-24 08:54:17 -0500 (Fri, 24 Sep 2010) | 12 lines Merged revisions 288712 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r288712 | tilghman | 2010-09-24 08:53:30 -0500 (Fri, 24 Sep 2010) | 5 lines Solaris won't printf a NULL. (closes issue #18041) Reported by: asgaroth ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@288714 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
David Vossel	2f3dee2379	Merged revisions 287647 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r287647 | dvossel | 2010-09-20 17:09:16 -0500 (Mon, 20 Sep 2010) | 21 lines Addition of the FrameHook API (AKA AwesomeHooks) So far all our tools for viewing and manipulating media streams within Asterisk have been entirely focused on audio. That made sense then, but is not scalable now. The FrameHook API lets us tap into and manipulate _ANY_ type of media or signaling passed on a channel present today or in the future. This tool is a step in the direction of expanding Asterisk's boundaries and will help generate some rather interesting applications in the future. In addition to the FrameHook API, a simple dialplan function exercising the api has been included as well. This function is called FRAME_TRACE(). FRAME_TRACE() allows for the internal ast_frames read and written to a channel to be output. Filters can be placed on this function to debug only certain types of frames. This function could be thought of as an internal way of doing ast_frame packet captures. Review: https://reviewboard.asterisk.org/r/925/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@287648 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Terry Wilson	d04046fbe7	Merged revisions 286189 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r286189 | twilson | 2010-09-10 17:04:53 -0500 (Fri, 10 Sep 2010) | 30 lines Merged revisions 286115 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r286115 | twilson | 2010-09-10 15:35:25 -0500 (Fri, 10 Sep 2010) | 23 lines Merged revisions 286059 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r286059 | twilson | 2010-09-10 14:25:08 -0500 (Fri, 10 Sep 2010) | 16 lines Inherit CHANNEL() writes to both sides of a Local channel Having Local (/n) channels as queue members and setting the language in the extension with Set(CHANNEL(language)=fr) sets the language on the Local/...,2 channel. Hold time report playbacks happen on the Local/...,1 channel and therefor do not play in the specified language. This patch modifies func_channel_write to call the setoption callback and pass the CHANNEL() write info to the callback. chan_local uses this information to look up the other side of the channel and apply the same changes to it. (closes issue #17673) Reported by: Guggemand Review: https://reviewboard.asterisk.org/r/903/ ........ ................ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@286190 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Tilghman Lesher	1c12ca0407	Merged revisions 285484 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r285484 | tilghman | 2010-09-08 02:14:17 -0500 (Wed, 08 Sep 2010) | 2 lines Documentation only ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@285485 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
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
Tilghman Lesher	8190e96fad	Merged revisions 284610 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r284610 | tilghman | 2010-09-02 00:20:59 -0500 (Thu, 02 Sep 2010) | 10 lines When optional_api is non-optional, force dependent modules to be loaded. (closes issue #17707) Reported by: ira Patches: 20100819__issue17707__asterisk1.8.diff.txt uploaded by tilghman (license 14) Tested by: tilghman Review: https://reviewboard.asterisk.org/r/876/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@284628 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Russell Bryant	2de5bbc89f	Merged revisions 283350 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r283350 | russell | 2010-08-24 07:49:41 -0500 (Tue, 24 Aug 2010) | 2 lines Don't attempt to release a NULL ODBC handle. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@283351 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Tilghman Lesher	42490d744b	Merged revisions 280809 via svnmerge from ... https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r280809 | tilghman | 2010-08-03 15:25:10 -0500 (Tue, 03 Aug 2010) | 12 lines Sneak FIELDNUM() into 1.8. Returns a 1-based index into a list of a specified item. Matches up with FIELDQTY() and CUT(). (closes issue #17713) Reported by: gareth Patches: svn-279754.diff uploaded by gareth (license 208) Tested by: gareth, tilghman Review: https://reviewboard.asterisk.org/r/810/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@280810 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Terry Wilson	d6e1c724e5	Remove built-in AES code and use optional_api instead ... Review: https://reviewboard.asterisk.org/r/793/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@278538 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Tilghman Lesher	b4e18d5660	Add load priority order, such that preload becomes unnecessary in most cases ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@278132 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	cf7bbcc4c6	Expand the caller ANI field to an ast_party_id ... Expand the ani field in ast_party_caller and ast_party_connected_line to an ast_party_id. This is an extension to the ast_callerid restructuring patch in review: https://reviewboard.asterisk.org/r/702/ Review: https://reviewboard.asterisk.org/r/744/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@276393 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	ec37ffbdaf	ast_callerid restructuring ... The purpose of this patch is to eliminate struct ast_callerid since it has turned into a miscellaneous collection of various party information. Eliminate struct ast_callerid and replace it with the following struct organization: struct ast_party_name { char *str; int char_set; int presentation; unsigned char valid; }; struct ast_party_number { char *str; int plan; int presentation; unsigned char valid; }; struct ast_party_subaddress { char *str; int type; unsigned char odd_even_indicator; unsigned char valid; }; struct ast_party_id { struct ast_party_name name; struct ast_party_number number; struct ast_party_subaddress subaddress; char *tag; }; struct ast_party_dialed { struct { char *str; int plan; } number; struct ast_party_subaddress subaddress; int transit_network_select; }; struct ast_party_caller { struct ast_party_id id; char *ani; int ani2; }; The new organization adds some new information as well. * The party name and number now have their own presentation value that can be manipulated independently. ISDN supplies the presentation value for the name and number at different times with the possibility that they could be different. * The party name and number now have a valid flag. Before this change the name or number string could be empty if the presentation were restricted. Most channel drivers assume that the name or number is then simply not available instead of indicating that the name or number was restricted. * The party name now has a character set value. SIP and Q.SIG have the ability to indicate what character set a name string is using so it could be presented properly. * The dialed party now has a numbering plan value that could be useful to have available. The various channel drivers will need to be updated to support the new core features as needed. They have simply been converted to supply current functionality at this time. The following items of note were either corrected or enhanced: * The CONNECTEDLINE() and REDIRECTING() dialplan functions were consolidated into func_callerid.c to share party id handling code. * CALLERPRES() is now deprecated because the name and number have their own presentation values. * Fixed app_alarmreceiver.c write_metadata(). The workstring[] could contain garbage. It also can only contain the caller id number so using ast_callerid_parse() on it is silly. There was also a typo in the CALLERNAME if test. * Fixed app_rpt.c using ast_callerid_parse() on the channel's caller id number string. ast_callerid_parse() alters the given buffer which in this case is the channel's caller id number string. Then using ast_shrink_phone_number() could alter it even more. * Fixed caller ID name and number memory leak in chan_usbradio.c. * Fixed uninitialized char arrays cid_num[] and cid_name[] in sig_analog.c. * Protected access to a caller channel with lock in chan_sip.c. * Clarified intent of code in app_meetme.c sla_ring_station() and dial_trunk(). Also made save all caller ID data instead of just the name and number strings. * Simplified cdr.c set_one_cid(). It hand coded the ast_callerid_merge() function. * Corrected some weirdness with app_privacy.c's use of caller presentation. Review: https://reviewboard.asterisk.org/r/702/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@276347 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Tilghman Lesher	0ae9097e3e	Oops, XML documentation fix. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@276122 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Tilghman Lesher	fc9efc4ff5	It really cannot fail in the places below, but the stupid compiler doesn't know that. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@276120 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Tilghman Lesher	e939dfea9d	Weird compiler error on Bamboo. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@276118 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Tilghman Lesher	50d5f134c8	FILE() now supports line-mode and writing (altering) files. ... (closes issue #16461) Reported by: skyman Patches: 20100622__issue16461.diff.txt uploaded by tilghman (license 14) Tested by: tilghman Review: https://reviewboard.asterisk.org/r/737/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@276114 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Tilghman Lesher	da8450323f	Kill some startup warnings and errors and make some messages more helpful in tracking down the source. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@275105 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Bradley Latus	4405813297	Add High Resolution Times to CDRs for Asterisk ... People expressed an interest in having access to the exact length of calls to a finer degree than seconds. See the CHANGES and UPGRADE.txt for usage also updated the sample configs to note the change. Patch by snuffy. (closes issue #16559) Reported by: cianmaher Tested by: cianmaher, snuffy Review: https://reviewboard.asterisk.org/r/461/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@269153 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
Tilghman Lesher	da0138932e	Handle OOM errors more gracefully. ... (closes issue #17084) Reported by: falves11 Patches: issue17084_162_A.diff uploaded by falves11 (license 374) Tested by: falves11 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@267669 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Tilghman Lesher	4eaea01cad	Needs to be wrapped in <para> ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@266522 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Tilghman Lesher	2da88f1977	Setup environment variables for the benefit of child processes and disallow changing them. ... (closes issue #14899) Reported by: jmls Patches: 20090916__issue14899.diff.txt uploaded by tilghman (license 14) Tested by: jmls git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@266385 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Mark Michelson	b5d5cc565f	Enhancements to connected line and redirecting work. ... From reviewboard: Digium has a commercial customer who has made extensive use of the connected party and redirecting information present in later versions of Asterisk Business Edition and which is to be in the upcoming 1.8 release. Through their use of the feature, new problems and solutions have come about. This patch adds several enhancements to maximize usage of the connected party and redirecting information functionality. First, Asterisk trunk already had connected line interception macros. These macros allow you to manipulate connected line information before it was sent out to its target. This patch adds the same feature except for redirecting information instead. Second, the ast_callerid and ast_party_id structures have been enhanced to provide a "tag." This tag can be set with func_callerid, func_connectedline, func_redirecting, and in the case of DAHDI, mISDN, and SIP channels, can be set in a configuration file. The idea behind the callerid tag is that it can be set to whatever value the administrator likes. Later, when running connected line and redirecting macros, the admin can read the tag off the appropriate structure to determine what action to take. You can think of this sort of like a channel variable, except that instead of having the variable associated with a channel, the variable is associated with a specific identity within Asterisk. Third, app_dial has two new options, s and u. The s option lets a dialplan writer force a specific caller ID tag to be placed on the outgoing channel. The u option allows the dialplan writer to force a specific calling presentation value on the outgoing channel. Fourth, there is a new control frame subclass called AST_CONTROL_READ_ACTION added. This was added to correct a very specific situation. In the case of SIP semi-attended (blond) transfers, the party being transferred would not have the opportunity to run a connected line interception macro to possibly alter the transfer target's connected line information. The issue here was that during a blond transfer, the SIP transfer code has no bridged channel on which to queue the connected line update. The way this was corrected was to add this new control frame subclass. Now, we queue an AST_CONTROL_READ_ACTION frame on the channel on which the connected line interception macro should be run. When ast_read is called to read the frame, ast_read responds by calling a callback function associated with the specific read action the control frame describes. In this case, the action taken is to run the connected line interception macro on the transferee's channel. Review: https://reviewboard.asterisk.org/r/652/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@263541 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Tilghman Lesher	03e1608c29	Double free crash ... (closes issue #17245) Reported by: thedavidfactor Patches: 20100426__issue17245.diff.txt uploaded by tilghman (license 14) Tested by: murraytm git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@261917 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Mark Michelson	693d1c44b1	Add small documentation update to func_callcompletion.c. ... This directs users to documents which can help explain the concepts and configuration options settable with the function. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@258345 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Mark Michelson	6640f309a9	Commit compromise I suggested on review 608. ... This allows for multiple SRV queries to be done from the dialplan for the same service on a single call while still allowing one to bypass the call to SRVQUERY if they so please. Taking action since no comments had been left for a while. This can easily be reverted if needed. External tests still pass. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@257851 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Mark Michelson	fb0a4e5bd0	Address Russell's comments on func_srv from reviewboard. ... * Change copyright date * Place channel in autoservice when doing SRV lookup * Get rid of trailing whitespace * Change logic in load_module function git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@257025 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Mark Michelson	ae7b76a1b9	Fix some compiler errors that popped up after the CCSS merge. ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@256529 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Mark Michelson	e24661fd18	Merge Call completion support into trunk. ... From Reviewboard: CCSS stands for Call Completion Supplementary Services. An admittedly out-of-date overview of the architecture can be found in the file doc/CCSS_architecture.pdf in the CCSS branch. Off the top of my head, the big differences between what is implemented and what is in the document are as follows: 1. We did not end up modifying the Hangup application at all. 2. The document states that a single call completion monitor may be used across multiple calls to the same device. This proved to not be such a good idea when implementing protocol-specific monitors, and so we ended up using one monitor per-device per-call. 3. There are some configuration options which were conceived after the document was written. These are documented in the ccss.conf.sample that is on this review request. For some basic understanding of terminology used throughout this code, see the ccss.tex document that is on this review. This implements CCBS and CCNR in several flavors. First up is a "generic" implementation, which can work over any channel technology provided that the channel technology can accurately report device state. Call completion is requested using the dialplan application CallCompletionRequest and can be canceled using CallCompletionCancel. Device state subscriptions are used in order to monitor the state of called parties. Next, there is a SIP-specific implementation of call completion. This method uses the methods outlined in draft-ietf-bliss-call-completion-06 to implement call completion using SIP signaling. There are a few things to note here: * The agent/monitor terminology used throughout Asterisk sometimes is the reverse of what is defined in the referenced draft. * Implementation of the draft required support for SIP PUBLISH. I attempted to write this in a generic-enough fashion such that if someone were to want to write PUBLISH support for other event packages, such as dialog-state or presence, most of the effort would be in writing callbacks specific to the event package. * A subportion of supporting PUBLISH reception was that we had to implement a PIDF parser. The PIDF support added is a bit minimal. I first wrote a validation routine to ensure that the PIDF document is formatted properly. The rest of the PIDF reading is done in-line in the call-completion-specific PUBLISH-handling code. In other words, while there is PIDF support here, it is not in any state where it could easily be applied to other event packages as is. Finally, there are a variety of ISDN-related call completion protocols supported. These were written by Richard Mudgett, and as such I can't really say much about their implementation. There are notes in the CHANGES file that indicate the ISDN protocols over which call completion is supported. Review: https://reviewboard.asterisk.org/r/523 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@256528 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Mark Michelson	6cad0f1602	func_srv and explicit specification of a remote IP for SIP. ... From Review Board: There are two interrelated changes here. First, there is the introduction of func_srv. This adds two new read-only dialplan functions, SRVQUERY and SRVRESULT. They work very similarly to the ENUMQUERY and ENUMRESULT functions, except that this allows one to query SRV records instead. In order to facilitate this work, I added a couple of new API calls to srv.h. ast_srv_get_record_count tells the number of records returned by an SRV lookup. This number is calculated at the time of the SRV lookup. ast_srv_get_nth_record allows one to get a numbered SRV record. Second, there is the modification to chan_sip that allows one to specify a hostname or IP address (along with a port) to send an outgoing INVITE to when dialing a SIP peer. This goes hand-in-hand with func_srv. You can query SRV records and then use the host and port from the results to dial via a specific host instead of what is configured in sip.conf. Review: https://reviewboard.asterisk.org/r/608 SWP-1200 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@256485 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	a5a0a5f867	Consolidate ast_channel.cid.cid_rdnis into ast_channel.redirecting.from.number. ... SWP-1229 ABE-2161 * Ensure chan_local.c:local_call() will not leak cid.cid_dnid when copying. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@256104 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Russell Bryant	008930a3f2	Fix memory corruption found by unit tests. ... ast_str_reset() was being called on a potentially uninitialized pointer. Valgrind is my hero, once again. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@253579 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Tilghman Lesher	afb6bac829	Hmmm, apparently needed to be fixed in trunk, too. ... (closes issue #16900) Reported by: bluecrow76 Patches: asterisk-1.6.2.4-func_strings.diff uploaded by bluecrow76 (license 270) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@251682 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Tilghman Lesher	dd3176cc91	It's amazing what writing a test will find. ... (issue #16900) Reported by: bluecrow76 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@251677 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Tilghman Lesher	e58fc610ae	Change needed to make Mac OS X 10.6 happy ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@251262 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
David Vossel	86a215c83e	fixes xml error in func_pitchshift ... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@251087 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
David Vossel	f468595789	PITCH_SHIFT dialplan function ... The PITCH_SHIFT function can be used on a channel to independently modify the pitch of both rx and tx audio streams. Now you can improve your conference calls by assigning a random pitch effect to everyone entering a meetme room, or just make your day more interesting by making your co-workers sound funny. These are just some of the numerious practical uses for this function. Enjoy! https://reviewboard.asterisk.org/r/526/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@251038 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