asterisk

Commit Graph

Author	SHA1	Message	Date
Matthew Jordan	944cdaa94d	Merged revisions 337118 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r337118 \| mjordan \| 2011-09-20 17:38:54 -0500 (Tue, 20 Sep 2011) \| 21 lines Fix for incorrect voicemail duration in external notifications This patch fixes an issue where the voicemail duration was being reported with a duration significantly less than the actual sound file duration. Voicemails that contained mostly silence were reporting the duration of only the sound in the file, as opposed to the duration of the file with the silence. This patch fixes this by having two durations reported in the __ast_play_and_record family of functions - the sound_duration and the actual duration of the file. The sound_duration, which is optional, now reports the duration of the sound in the file, while the actual full duration of the file is reported in the duration parameter. This allows the voicemail applications to use the sound_duration for minimum duration checking, while reporting the full duration to external parties if the voicemail is kept. (issue ASTERISK-2234) (closes issue ASTERISK-16981) Reported by: Mary Ciuciu, Byron Clark, Brad House, Karsten Wemheuer, KevinH Tested by: Matt Jordan Review: https://reviewboard.asterisk.org/r/1443 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@337120 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Matthew Jordan	be5c67401d	Merged revisions 332817 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r332817 \| mjordan \| 2011-08-22 13:15:51 -0500 (Mon, 22 Aug 2011) \| 4 lines Review: https://reviewboard.asterisk.org/r/1364/ This update adds a new AMI event, TestEvent, which is enabled when the TEST_FRAMEWORK compiler flag is defined. It also adds initial usage of this event to app_voicemail. The TestEvent AMI event is used extensively by the voicemail tests in the Asterisk Test Suite. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@332832 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Jonathan Rose	361d40e7fb	Merged revisions 329527 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r329527 \| jrose \| 2011-07-26 08:25:35 -0500 (Tue, 26 Jul 2011) \| 17 lines Fixes some voicemail forwarding behavior based around prepend mode. Formerly, prepend forwarding would have the user record a message with no useful prompt and an expectation for the user to push a button on the phone when finished recording. If a length of silence was detected instead, the recording would be canceled and the user would re-enter the voicemail forwarding menu. Subsequent time-outs in prepend recording would also bug out in the sense that they would write over the original message and get sent to the recipient regardless of whether they timed out or were accepted. This patch fixes this issue and adds a prompt which will be played after a timeout informing the user that they needed to press a button. Currently, the sound files that we have are somewhat inadquate for this, so after the call we simply have Allison say "Please try again. Then press pound." which actually relies on two separate sound files. Just one would be more appropriate. reporter: Vlad Povorozniuc Review: https://reviewboard.asterisk.org/r/1327/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/10@329528 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Leif Madsen	f2df0ed9f1	Increase prepend filename length. (closes issue #19238) Reported by: byronclark Patches: increase_prepend_filename_length.patch uploaded by byronclark (license 1200) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@318194 65c4cc65-6c06-0410-ace0-fbb531ad65f3	14 years ago
Paul Belanger	3556e4c2d4	Replace ast_log(LOG_DEBUG, ...) with ast_debug() (closes issue #18556) Reported by: kkm Review: https://reviewboard.asterisk.org/r/1071/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@306258 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
David Vossel	c26c190711	Asterisk media architecture conversion - no more format bitfields This patch is the foundation of an entire new way of looking at media in Asterisk. The code present in this patch is everything required to complete phase1 of my Media Architecture proposal. For more information about this project visit the link below. https://wiki.asterisk.org/wiki/display/AST/Media+Architecture+Proposal The primary function of this patch is to convert all the usages of format bitfields in Asterisk to use the new format and format_cap APIs. Functionally no change in behavior should be present in this patch. Thanks to twilson and russell for all the time they spent reviewing these changes. Review: https://reviewboard.asterisk.org/r/1083/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@306010 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Tilghman Lesher	f8bbf4bfa2	Merged revisions 302634 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r302634 \| tilghman \| 2011-01-19 14:24:57 -0600 (Wed, 19 Jan 2011) \| 22 lines Merged revisions 302599 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r302599 \| tilghman \| 2011-01-19 14:13:24 -0600 (Wed, 19 Jan 2011) \| 15 lines Kill zombies. When we ast_safe_fork() with a non-zero argument, we're expected to reap our own zombies. On a zero argument, however, the zombies are only reaped when there aren't any non-zero forked children alive. At other times, we accumulate zombies. This code is forward ported from res_agi in 1.4, so that forked children are always reaped, thus preventing an accumulation of zombie processes. (closes issue #18515) Reported by: ernied Patches: 20101221__issue18515.diff.txt uploaded by tilghman (license 14) Tested by: ernied ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@302644 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Russell Bryant	5d613c436b	Remove a trailing space. (testing something with bamboo ...) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@295125 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Olle Johansson	cc64448e2f	Whitespace cleanup git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@286342 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
Mark Michelson	cd4ebd336f	Add IPv6 to Asterisk. This adds a generic API for accommodating IPv6 and IPv4 addresses within Asterisk. While many files have been updated to make use of the API, chan_sip and the RTP code are the files which actually support IPv6 addresses at the time of this commit. The way has been paved for easier upgrading for other files in the near future, though. Big thanks go to Simon Perrault, Marc Blanchet, and Jean-Philippe Dionne for their hard work on this. (closes issue #17565) Reported by: russell Patches: asteriskv6-test-report.pdf uploaded by russell (license 2) Review: https://reviewboard.asterisk.org/r/743 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@274783 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
David Vossel	77a96c5a93	do all sip registry parsing before transmit_register This patch breaks up every part of the sip registry string during config parsing and removes all parsing from transmit_register(). Thanks to Nick_Lewis for contributing this patch! (closes issue #14331) Reported by: Nick_Lewis Patches: chan_sip.c-domparse.patch uploaded by Nick Lewis (license 657) chan_sip.c.patch uploaded by Nick Lewis (license 657) chan_sip.c.domainparse3.patch uploaded by Nick Lewis (license 657) chan_sip.c-domparse4.patch uploaded by Nick Lewis (license 657) chan_sip.c-domparse5.patch uploaded by Nick Lewis (license 657) nicklewispatch.diff uploaded by dvossel (license 671) Tested by: Nick_Lewis, dvossel Review: https://reviewboard.asterisk.org/r/628/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@266090 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Tilghman Lesher	8ced3317ed	Merged revisions 257544 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r257544 \| tilghman \| 2010-04-15 16:23:24 -0500 (Thu, 15 Apr 2010) \| 6 lines Allow application options with arguments to contain parentheses, through a variety of escaping techniques. Fixes SWP-1194 (ABE-2143). Review: https://reviewboard.asterisk.org/r/604/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@257560 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Russell Bryant	f72c83684b	Resolve unit test failure that occurred on Mac OSX. On Linux (glibc), regcomp() does not return an error for an empty string. However, the version on OSX will return an error. The test for channel group matching by regex now passes on the mac, as well. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@252241 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Jeff Peeler	27a4cda821	Add support for GROUP_MATCH_COUNT regex matching on category Current support for regex matching was previously only available on the group. Also, error reporting for regex failures has been added. In addition to this feature enhancement a unit test has been written to check the regular expression logic to ensure the count operation is working as expected. (closes issue #16642) Reported by: kobaz Patches: groupmatch2.patch uploaded by kobaz (license 834) Review: https://reviewboard.asterisk.org/r/503/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@247295 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Jeff Peeler	2414bc8005	Add audio announcement option to app_page As described in the CHANGES file: * MeetMe has a new option 'G' to play an announcement before joining a conference. * Page has a new option 'A(x)' which will playback an announcement simultaneously to all paged phones (and optionally excluding the caller's one using the new option 'n') before the call is bridged. To add the new option to meetme, the conference flag options had to be extended to 64 bits. (closes issue #14365) Reported by: dferrer Patches: page_announce.patch uploaded by dferrer (license 525) modified by me Review: https://reviewboard.asterisk.org/r/188/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@234173 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	cfd17ef0a6	Move implementation of closefrom(3) from app.c to strcompat.c git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@233358 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Matthew Nicholson	31848bcdd1	Merged revisions 231614 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r231614 \| mnicholson \| 2009-11-30 15:11:44 -0600 (Mon, 30 Nov 2009) \| 8 lines Remove duplicate entries from voicemail format lists. This prevents app_voicemail from entering an infinite loop when the same format is specified twice in the format list. (closes issue #15625) Reported by: Shagg63 Tested by: mnicholson Review: https://reviewboard.asterisk.org/r/429/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@231688 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Matthew Nicholson	936a2bd202	Reverted 231616 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@231637 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Matthew Nicholson	8d1f4fa5ea	Merged revisions 231614 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r231614 \| mnicholson \| 2009-11-30 15:11:44 -0600 (Mon, 30 Nov 2009) \| 8 lines Remove duplicate entries from voicemail format lists. This prevents app_voicemail from entering an infinite loop when the same format is specified twice in the format list. (closes issue #15625) Reported by: Shagg63 Tested by: mnicholson Review: https://reviewboard.asterisk.org/r/429/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@231616 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	d8e0c58437	Expand codec bitfield from 32 bits to 64 bits. Reviewboard: https://reviewboard.asterisk.org/r/416/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@227580 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	c80715706e	Remove unnecessary typedef git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@224403 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	c74a2d0b45	Create an API for adding an optional time unit onto the ends of time periods. Two examples of its use are included, and the usage could be expanded in some cases into certain configuration options where time periods are specified. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@224225 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	57a9927143	Close up to the soft open file limit (same on Linux, but varies drastically on OS X). Also, a Makefile fix for Darwin (OS X). (closes issue #14542) Reported by: jtodd Patches: 20090901__issue14542.diff.txt uploaded by tilghman (license 14) Tested by: jtodd, tilghman Change-type: bugfix git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@215567 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	a737df8603	Allow Gosub to recognize quote delimiters without consuming them. (closes issue #15557) Reported by: rain Patches: 20090723__issue15557.diff.txt uploaded by tilghman (license 14) Tested by: rain Review: https://reviewboard.asterisk.org/r/316/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@210908 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Russell Bryant	236c10a070	Only update total silence counter after a counter reset. (closes issue #2264) Reported by: pfn Patches: silent-vm-1.6.2-fix2.txt uploaded by pfn (license 810) Tested by: pfn git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@203962 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Russell Bryant	0bbd5c9424	Ignore voicemail messages that are just silence. (closes issue #2264) Reported by: pfn Patches: silent-vm-1.6.2.txt uploaded by pfn (license 810) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@202570 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 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	16 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	16 years ago
Jeff Peeler	11ac1f7e11	Fix building of chan_h323 with gcc-3.3 There seems to be a bug with old versions of g++ that doesn't allow a structure member to use the name list. Rename list member to group_list in ast_group_info and change the few places it is used. (closes issue #14790) Reported by: stuarth git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@190057 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	1030a25ac9	Modify headers and macros, according to Russell's suggestions on the -dev list git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@187599 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Russell Bryant	2a4f9f7181	Change global_app_buf to ast_str_thread_global_buf. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@184693 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	eb5bb03b82	Spacing changes only git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@180259 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
David Vossel	979eb709ae	app_read does not break from prompt loop with user terminated empty string In app.c, ast_app_getdata is called to stream the prompts and receive DTMF input. If ast_app_getdata() receives an empty string caused by the user inputing the end of string character, in this case '#', it should break from the prompt loop and return to app_read, but instead it cycles through all the prompts. I've added a return value for this special case in ast_readstring() which uses an enum I've delcared in apps.h. This enum is now used as a return value for ast_app_getdata(). (closes issue #14279) Reported by: Marquis Patches: fix_app_read.patch uploaded by Marquis (license 32) read-ampersanmd.patch2 uploaded by dvossel (license 671) Tested by: Marquis, dvossel Review: http://reviewboard.digium.com/r/177/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@180032 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	3252cd2e5b	Merged revisions 179468 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r179468 \| tilghman \| 2009-03-02 17:09:01 -0600 (Mon, 02 Mar 2009) \| 10 lines When ending a recording with silence detection, remember to reduce the duration. The end of the recording is correspondingly trimmed, but the duration was not trimmed by the number of seconds trimmed, so the saved duration was necessarily longer than the actual soundfile duration. (closes issue #14406) Reported by: sasargen Patches: 20090226__bug14406.diff.txt uploaded by tilghman (license 14) Tested by: sasargen ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@179469 65c4cc65-6c06-0410-ace0-fbb531ad65f3	16 years ago
Tilghman Lesher	3af1c558df	Allow semicolons to be escaped, when passing arguments to the System command. (closes issue #14231) Reported by: jcovert Patches: 20090113__bug14231__2.diff.txt uploaded by Corydon76 (license 14) corrected_20090113__bug14231__2.diff.txt uploaded by jcovert (license 551) Tested by: jcovert git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@177664 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
Tilghman Lesher	80f91f6ab7	Merged revisions 172438 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r172438 \| tilghman \| 2009-01-29 16:54:29 -0600 (Thu, 29 Jan 2009) \| 9 lines Lose the CAP_NET_ADMIN at every fork, instead of at startup. Otherwise, if Asterisk runs as a non-root user and the administrator does a 'restart now', Asterisk loses the ability to set QOS on packets. (closes issue #14004) Reported by: nemo Patches: 20090105__bug14004.diff.txt uploaded by Corydon76 (license 14) Tested by: Corydon76 ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@172441 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Olle Johansson	b79a12e929	- Make sure we set setvar= variables on outbound calls too, not only inbound calls. - Also, change a function in app.c to return a userful value instead of always returning 0. Patch by fnordian, changed by Corydon76 and myself. This does not close the bug report, as fnordian had an additional change we're still discussing. (related to issue #14059) Reported by: fnordian Patches: chan_sip_hfield.patch uploaded by fnordian (license 110) 20090116__bug14059.diff.txt uploaded by Corydon76 (license 14) Tested by: fnordian, Corydon76, oej git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@172268 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 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
Tilghman Lesher	fd3cb90841	Some platforms (notably, the BSDs) have a more efficient implementation called closefrom(3). git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@168522 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	c8223fc957	Merge ast_str_opaque branch (discontinue usage of ast_str internals) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@163991 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Russell Bryant	da0737c00c	Merged revisions 161948 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r161948 \| russell \| 2008-12-09 08:52:25 -0600 (Tue, 09 Dec 2008) \| 15 lines Fix a problem with GROUP() settings on a masquerade. The previous code carried over group settings from the old channel to the new one. However, it did nothing with the group settings that were already on the new channel. This patch removes all group settings that already existed on the new channel. I have a more complicated version of this patch which addresses only the most blatant problem with this, which is that a channel can end up with multiple group settings in the same category. However, I could not think of a use case for keeping any of the group settings from the old channel, so I went this route for now. (closes AST-152) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@161951 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	0d25ddd366	Add LISTFILTER dialplan function, along with supporting documentation. See documentation for more information on how to use it. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@154915 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	bbf8c3d7cc	When checking for an encoded character, make sure the string isn't blank, first. (Closes issue #13470) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@142748 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Sean Bright	b69c8e6ab5	Another big chunk of changes from the RSW branch. Bunch of stuff from main/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@137082 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Mark Michelson	8b310d67b1	Merged revisions 134475 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r134475 \| mmichelson \| 2008-07-30 13:31:47 -0500 (Wed, 30 Jul 2008) \| 4 lines Fix a spot where a function could return without bringing a channel out of autoservice. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@134476 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	885d17506b	Keep ast_app_inboxcount API compatible with 1.6.0. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@127609 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Tilghman Lesher	f5a6e39bf8	Merged revisions 124395 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r124395 \| tilghman \| 2008-06-20 17:02:55 -0500 (Fri, 20 Jun 2008) \| 3 lines If the last character in a string to be parsed is the delimiter, then we should count that final empty string as an additional argument. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@124396 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago
Michiel van Baak	f1e9371da8	- revert change to ast_queue_hangup and create ast_queue_hangup_with_cause - make data member of the ast_frame struct a named union instead of a void Recently the ast_queue_hangup function got a new parameter, the hangupcause Feedback came in that this is no good and that instead a new function should be created. This I did. The hangupcause was stored in the seqno member of the ast_frame struct. This is not very elegant, and since there's already a data member that one should be used. Problem is, this member was a void *. Now it's a named union so it can hold a pointer, an uint32 and there's a padding in case someone wants to store another type in there in the future. This commit is so massive, because all ast_frame.data uses have to be altered to ast_frame.data.data Thanks russellb and kpfleming for the feedback. (closes issue #12674) Reported by: mvanbaak git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@117802 65c4cc65-6c06-0410-ace0-fbb531ad65f3	17 years ago

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115 Commits (283f63c594d0c4327c51fdae0587a6a9f1cb15d3)