asterisk

Commit Graph

Author	SHA1	Message	Date
Richard Mudgett	eca6952995	Simplify locking code for REDIRECTING interception macro when forwarding a call. Simplified the locking code by using a local copy of the redirecting party information in app_dial.c:do_forward() and app_queue.c:wait_for_answer() for launching the REDIRECTING interception macro when a call is forwarded. Reduced the lock time of the 'o->chan' and 'in' channels. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@288080 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Brett Bryant	71cbbd60de	Merged revisions 287759 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r287759 \| bbryant \| 2010-09-20 19:58:26 -0400 (Mon, 20 Sep 2010) \| 23 lines Merged revisions 287758 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r287758 \| bbryant \| 2010-09-20 19:57:08 -0400 (Mon, 20 Sep 2010) \| 16 lines Fix misvalidation of meetme pins in conjunction with the 'a' MeetMe flag. When using the 'a' MeetMe flag and having a user and admin pin setup for your conference, using the user pin would gain you admin priviledges. Also, when no user pin was set, an admin pin was, the 'a' MeetMe flag wasn't used, and the user tried to enter a conference then they were still prompted for a pin and forced to hit #. (closes issue #17908) Reported by: kuj Patches: pins_2.patch uploaded by kuj (license 1111) Tested by: kuj Review: [full review board URL with trailing slash] ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@287760 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Tilghman Lesher	72718e1183	Merged revisions 287387 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r287387 \| tilghman \| 2010-09-17 16:08:00 -0500 (Fri, 17 Sep 2010) \| 14 lines Merged revisions 287386 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r287386 \| tilghman \| 2010-09-17 16:06:03 -0500 (Fri, 17 Sep 2010) \| 7 lines Blank columns should get set on reload, not ignored. (closes issue #16893) Reported by: haakon Patches: 20100818__issue16893.diff.txt uploaded by tilghman (license 14) ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@287388 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Russell Bryant	daf14509c5	Set the default for "autofill" and "shared_lastcall" to "yes" in queues.conf. Review: https://reviewboard.asterisk.org/r/922/ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@287193 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Jeff Peeler	149f98f25b	Merged revisions 286998 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r286998 \| jpeeler \| 2010-09-15 15:28:02 -0500 (Wed, 15 Sep 2010) \| 14 lines Merged revisions 286941 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r286941 \| jpeeler \| 2010-09-15 15:08:52 -0500 (Wed, 15 Sep 2010) \| 7 lines Ensure mailbox is not filled to capacity before doing message forwarding. Specifically, before prompting to record a prepended message the capacity is checked first. If the mailbox is full the extension will be reprompted. ABE-2517 ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@287015 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Brett Bryant	69e4f34914	Merged revisions 285532 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r285532 \| bbryant \| 2010-09-08 16:56:12 -0400 (Wed, 08 Sep 2010) \| 8 lines Fixes a bug with MeetMe where after announcing the amount of time left in a conference, if music on hold was playing, it doesn't restart. (closes issue #17408) Reported by: sysreq Patches: asterisk-issue-17408_fixed.patch uploaded by sysreq (license 1009) Tested by: sysreq ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@285533 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Brett Bryant	8131d12a71	Merged revisions 285196 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r285196 \| bbryant \| 2010-09-07 13:49:07 -0400 (Tue, 07 Sep 2010) \| 17 lines Merged revisions 285194 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r285194 \| bbryant \| 2010-09-07 13:45:41 -0400 (Tue, 07 Sep 2010) \| 10 lines Fixes voicemail.conf issues where mailboxes with passwords that don't precede a comma would throw unnecessary error messages. (closes issue #15726) Reported by: 298 Patches: M15726.diff uploaded by junky (license 177) Tested by: junky Review: [full review board URL with trailing slash] ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@285197 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Terry Wilson	2e701efc81	Merged revisions 284897 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r284897 \| twilson \| 2010-09-03 11:20:45 -0500 (Fri, 03 Sep 2010) \| 12 lines Merged revisions 284881 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r284881 \| twilson \| 2010-09-03 11:10:23 -0500 (Fri, 03 Sep 2010) \| 5 lines Properly detect when a sound file doesn't exist ast_fileexists returns -1 for error and 0 for a non-existant file. The existing code treated missing files as though they existed. ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@284921 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Tilghman Lesher	8bc90ad39a	Merged revisions 284631 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r284631 \| tilghman \| 2010-09-02 00:30:16 -0500 (Thu, 02 Sep 2010) \| 7 lines Don't reset queue stats on a module reload. (closes issue #17535) Reported by: raarts Patches: 20100819__issue17535.diff.txt uploaded by tilghman (license 14) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@284632 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Tilghman Lesher	7e3f95e00a	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/branches/1.8@284610 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Tilghman Lesher	96f182bec0	Merged revisions 284280 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r284280 \| tilghman \| 2010-08-30 17:27:06 -0500 (Mon, 30 Aug 2010) \| 11 lines Fix 3 coding errors: 1) After we close FD, we should not be trying to write to it. 2) Call _exit(0), not exit(0), to avoid running shutdown routines in a child. 3) Use endian, not processor, detection to ensure bytes are written in the correct order. (closes issue #15706) Reported by: modelnine Patches: asterisk-1.6.1.1-festival-debug.patch uploaded by modelnine (license 865) Tested by: gmartinez ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@284281 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Russell Bryant	c3ad0f569d	Add an argument missing from the CELGenUserEvent documentation. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@282979 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Tilghman Lesher	ddceaeeae9	Merged revisions 281722 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r281722 \| tilghman \| 2010-08-11 10:17:20 -0500 (Wed, 11 Aug 2010) \| 7 lines Only set status TIMEOUT, if we have no digits. (closes issue #15188) Reported by: jcovert Patches: app_readexten.c.patch-1.6.2.8-rc1 uploaded by jcovert (license 551) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@281723 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Russell Bryant	325a4cd89a	Merged revisions 281567 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r281567 \| russell \| 2010-08-10 12:47:13 -0500 (Tue, 10 Aug 2010) \| 15 lines Merged revisions 281566 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r281566 \| russell \| 2010-08-10 12:45:45 -0500 (Tue, 10 Aug 2010) \| 8 lines Reset visible indication after answer. (closes issue #17641) Reported by: klaus3000 Patches: ast1.6.2.9-app_dial-visible_indication.patch.txt uploaded by klaus3000 (license 65) Tested by: schmidts ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@281568 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
TransNexus OSP Development	cc67d321e8	Fixed the issue caused by EXTEN including user parameters. git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@281497 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Tilghman Lesher	0fed1acd2a	Merged revisions 280671 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r280671 \| tilghman \| 2010-08-02 16:26:11 -0500 (Mon, 02 Aug 2010) \| 2 lines Allow the pipe, but also allow the comma ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@280672 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Jean Galarneau	ae2e66e707	Merged revisions 280345 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r280345 \| jeang \| 2010-07-29 11:01:35 -0500 (Thu, 29 Jul 2010) \| 10 lines Merged revisions 280341 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r280341 \| jeang \| 2010-07-29 10:52:31 -0500 (Thu, 29 Jul 2010) \| 2 lines Fix a dsp structure leak occuring when a local channel is put into a meetme conference, then masquaraded away. ABE-2422 ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@280346 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Sean Bright	5b52f62450	Merged revisions 280160 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r280160 \| seanbright \| 2010-07-28 12:51:11 -0400 (Wed, 28 Jul 2010) \| 8 lines Plug a reference leak in app_queue when adding members dynamically. (closes issue #17738) Reported by: bobwienholt Patches: issue17738.patch uploaded by bobwienholt (license 950) Tested by: bobwienholt, seanbright ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@280161 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Richard Mudgett	6341d1b2ad	Merged revisions 279207 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r279207 \| rmudgett \| 2010-07-23 17:11:23 -0500 (Fri, 23 Jul 2010) \| 14 lines Merged revisions 279206 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r279206 \| rmudgett \| 2010-07-23 16:56:44 -0500 (Fri, 23 Jul 2010) \| 7 lines SIP promiscuous redirect could fail to dial the redirect. The ast_channel was created with one variable to ast_request() but the call to ast_call() that initiates the outgoing call was using a different variable. The two variables are not equivalent if the call_forward string included a channel technology specifier. e.g., SIP/200 ........ ................ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.8@279227 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Tilghman Lesher	9bb8dc67e7	Ensure realtime conferences are treated the same as static conferences when trying to find an empty one. Also, parse the useropts properly, when retrieving from realtime, and add them to the existing flags. (closes issue #17502) Reported by: kenji Patches: 20100720__issue17502.diff.txt uploaded by tilghman (license 14) Tested by: kenji git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@278463 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Tilghman Lesher	ebf651105e	Merged revisions 278261 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r278261 \| tilghman \| 2010-07-20 17:23:13 -0500 (Tue, 20 Jul 2010) \| 7 lines Delete IMAP messages in reverse order, to ensure reordering after each expunge does not cause deletion of the wrong message. (closes issue #16350) Reported by: noahisaac Patches: 20100623__issue16350.diff.txt uploaded by tilghman (license 14) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@278275 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
Jeff Peeler	5b8a8fc6c8	Fix reporting estimated queue hold time. Just say the number of seconds (after minutes) rather than doing some incorrect calculation with respect to minutes. (closes issue #17498) Reported by: corruptor Patches: holdesecs_bug.diff uploaded by corruptor (license 253) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@277488 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Jeff Peeler	b73c1377e5	Add missing handling for ringing state for use with queue empty options. (closes issue #17471) Reported by: jazzy Patches: app_queue.c.diff uploaded by jazzy (license 1056) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@277366 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Paul Belanger	8eb9e0b938	Merged revisions 277182 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r277182 \| pabelanger \| 2010-07-16 13:10:36 -0400 (Fri, 16 Jul 2010) \| 8 lines Total analysis time error with SIP and silence suppression When using app_amd with SIP providers that have silence suppression on, the iTotalTime count increases exponentially. (closes issue #17656) Reported by: juls ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@277183 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Olle Johansson	65203b12dd	Add a dialplan function to check if a queue exists: QUEUE_EXISTS Review: https://reviewboard.asterisk.org/r/777/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@276950 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
Jeff Peeler	6535a1d0ed	Merged revisions 275773 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r275773 \| jpeeler \| 2010-07-12 15:34:51 -0500 (Mon, 12 Jul 2010) \| 12 lines Make user removals and traversals thread safe in meetme. Race conditions present in meetme involving the user list where a lack of locking has the potential for a user to be removed during a traversal or as in the case of the reporter after checking if the list is empty could cause a crash. Fixing this was done by convering the userlist to an ao2 container. (closes issue #17390) Reported by: Vince Review: https://reviewboard.asterisk.org/r/746/ ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@276074 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
TransNexus OSP Development	f1df8ea2bf	Added support for indirect work mode. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@275551 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Eliel C. Sardanons	7eafb1a763	When creating a conference for a unit test, it is not mandatory to open a dahdi pseudo channel, so if we fail doing it, continue creating the conference. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@275509 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Tilghman Lesher	2fdf43f9fc	Get more information about the Bamboo test failures git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@275312 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Russell Bryant	c5476ecb69	Fix compile error. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@275309 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Paul Belanger	d348c9aa1e	Include rdnis in msgXXXX.txt file. (closes issue #17566) Reported by: outcast Patches: voicemail-rdnis.patch uploaded by outcast (license 1071) Tested by: outcast git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@275307 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Tilghman Lesher	d6011adab4	Weird, no output and Bamboo still fails... git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@275227 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Tilghman Lesher	384681e182	Add some diagnostic feedback to our data tests git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@275172 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
Matthew Nicholson	759872902a	Merged revisions 275027 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r275027 \| mnicholson \| 2010-07-09 11:04:21 -0500 (Fri, 09 Jul 2010) \| 8 lines Clear the AST_CDR_FLAG_DIALED flag for channels going into the pbx via the G option in app_dial (closes issue #17592) Reported by: jamicque Patches: G-flag-cdr-fix1.diff uploaded by mnicholson (license 96) Tested by: jamicque, mnicholson ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@275028 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
Eliel C. Sardanons	a1b89a6a50	Implement AstData API data providers as part of the GSOC 2010 project, midterm evaluation. Review: https://reviewboard.asterisk.org/r/757/ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@274727 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Tilghman Lesher	1eaa09a0a2	Also run the externnotify script when the pollmailboxes thread notices a change. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@274491 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Tilghman Lesher	45a4bf35c2	The switch fallthrough could create some errorneous situations, so best to force directly to the default case. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@273714 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Tzafrir Cohen	c613897d1c	Fix various typos reported by Lintian (Also fix the typos in the comments) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@273641 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Jeff Peeler	b840ef081e	Merged revisions 273474 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r273474 \| jpeeler \| 2010-07-01 15:19:16 -0500 (Thu, 01 Jul 2010) \| 14 lines Allow admin user to join conference without using admin mode and no user pin. Configuring the conference in meetme.conf like the following: conf => 2345,,6666 did not prompt for pin when used without admin mode. This meant that the conference could not be joined as an admin even if the user knew the correct pin. The original bug report was submitted claiming that the blank user pin should deny entry into the conference. I think a better way to handle this would be with a feature enhancement that used the following syntax: conf => 2345,X,6666 - where X denotes no acceptable pin allowed (closes issue #15704) Reported by: modelnine ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@273522 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Jeff Peeler	bd9ff2829e	Merged revisions 273354 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r273354 \| jpeeler \| 2010-07-01 10:05:43 -0500 (Thu, 01 Jul 2010) \| 12 lines Ensure channel placed in meetme in ringing state is properly hung up. An outgoing channel placed in meetme while still ringing which was then hung up would not exit meetme and the channel was not properly destroyed. Specifically checking for this scenario by looking at the appropriate control frames resolves the issue. (closes issue #15871) Reported by: Ivan Patches: meetme_congestion_trunk_v2.patch uploaded by Ivan (license 229) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@273355 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Matthew Nicholson	cb22af3ec5	Merged revisions 272367 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 This version of the patch only adds AgentComplete for attended transfers. It was already present for blind transfers. ........ r272367 \| mnicholson \| 2010-06-23 17:33:51 -0500 (Wed, 23 Jun 2010) \| 8 lines Send AgentComplete manager events in the event of blind and attended transfers. (closes issue #16819) Reported by: elbriga Patches: app_queue.diff uploaded by elbriga (license 482) ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@272368 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Paul Belanger	90c850b5b1	Fix previous merge. ast_test_flag != ast_test_flag64 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@272259 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Paul Belanger	affec518d6	Merged revisions 272255 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r272255 \| pabelanger \| 2010-06-23 16:57:01 -0400 (Wed, 23 Jun 2010) \| 12 lines First caller into a dynamic conference now enter pin once. If MeetMe is configured to use dynamic conference numbers, then the first caller (which creates the conference) had to enter the PIN number twice. (closes issue #15878) Reported by: shawkris Patches: issue15878.patch uploaded by pabelanger (license 224) Tested by: pabelanger ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@272257 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Terry Wilson	2bcef29e11	Don't start the sla thread unless we realy need it git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@272146 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago
Terry Wilson	7938510af9	Make sure reload updates SLA config Even if there are no stations or trunks defined, we need to start the sla thread to make sure we get the reload event. Also, when doing a reload we need to remove the existing trunks and stations or they end up hanging around. (closes issue #16818) Reported by: mbonin Patches: sla_reload.patch uploaded by twilson (license 396) Tested by: twilson git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@272109 65c4cc65-6c06-0410-ace0-fbb531ad65f3	15 years ago

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3807 Commits (a08abc1dca23b832b055f288797038a10d98e17f)