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2011-05-25Merged revisions 320823 via svnmerge from rmudgett1-1/+19
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r320823 | rmudgett | 2011-05-25 12:06:38 -0500 (Wed, 25 May 2011) | 18 lines The AMI Newstate event contains different information between v1.4 and v1.8. The addition of connected line support in v1.8 changes the behavior of the channel caller ID somewhat. The channel caller ID value no longer time shares with the connected line ID on outgoing call legs. The timing of some AMI events/responses output the connected line ID as caller ID. These party ID's are now separate. * The ConnectedLineNum and ConnectedLineName headers were added to many AMI events/responses if the CallerIDNum/CallerIDName headers were also present. (closes issue #18252) Reported by: gje Tested by: rmudgett Review: https://reviewboard.asterisk.org/r/1227/ ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@320825 f38db490-d61c-443f-a65b-d21fe96a405b
2011-05-06Merged revisions 317584 via svnmerge from twilson1-0/+11
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r317584 | twilson | 2011-05-06 01:18:53 -0700 (Fri, 06 May 2011) | 20 lines Merged revisions 317575 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r317575 | twilson | 2011-05-06 01:04:17 -0700 (Fri, 06 May 2011) | 13 lines Merged revisions 317574 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r317574 | twilson | 2011-05-06 00:55:21 -0700 (Fri, 06 May 2011) | 6 lines Re-fix queue round-robin This part of the change for r315596 was incorrect. No bridge occurs when doing a roundrobin dial and no one answers, so this code shouldn't have been removed. ........ ................ ................ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@317596 f38db490-d61c-443f-a65b-d21fe96a405b
2011-05-05Merged revisions 317336 via svnmerge from russell1-1/+1
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r317336 | russell | 2011-05-05 14:55:58 -0500 (Thu, 05 May 2011) | 7 lines Increase buffer size to be PATH_MAX for a path. (closes issue #19239) Reported by: byronclark Patches: queue_announce_length.patch uploaded by byronclark (license 1200) ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@317337 f38db490-d61c-443f-a65b-d21fe96a405b
2011-05-03Merged revisions 316265 via svnmerge from russell1-3/+2
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r316265 | russell | 2011-05-03 14:55:49 -0500 (Tue, 03 May 2011) | 5 lines Fix a bunch of compiler warnings generated by gcc 4.6.0. Most of these are -Wunused-but-set-variable, but there were a few others mixed in here, as well. ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@316293 f38db490-d61c-443f-a65b-d21fe96a405b
2011-04-26Merged revisions 315644 via svnmerge from twilson1-11/+0
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r315644 | twilson | 2011-04-26 14:39:01 -0700 (Tue, 26 Apr 2011) | 32 lines Merged revisions 315643 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r315643 | twilson | 2011-04-26 14:27:44 -0700 (Tue, 26 Apr 2011) | 25 lines Merged revisions 315596 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r315596 | twilson | 2011-04-26 14:16:10 -0700 (Tue, 26 Apr 2011) | 18 lines Allow transfer loops without allowing forwarding loops We try to avoid the situation where two phones may be forwarded to each other causing an infinite loop by storing each dialed interface in a channel datastore and checking the list before dialing out. This works, but currently breaks situations like A calls B, A transfers B to C, B transfers C to A, and A transfers C to B. Since human interaction is happening here and not an automated forwarding loop, it should be allowed. This patch removes the dialed_interfaces datastore when a call is bridged (a suggestion from the brilliant mmichelson). If a call is being bridged, it should be safe to assume that we aren't stuck in a loop. Since we are now handling this is the bridge code, the previous attempts at handling it in app_dial and app_queue are removed. Review: https://reviewboard.asterisk.org/r/1195/ ........ ................ ................ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@315670 f38db490-d61c-443f-a65b-d21fe96a405b
2011-02-15Merged revisions 308010 via svnmerge from qwell1-1/+0
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r308010 | qwell | 2011-02-15 17:34:03 -0600 (Tue, 15 Feb 2011) | 24 lines Merged revisions 308007 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r308007 | qwell | 2011-02-15 17:33:24 -0600 (Tue, 15 Feb 2011) | 17 lines Merged revisions 308002 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r308002 | qwell | 2011-02-15 17:32:20 -0600 (Tue, 15 Feb 2011) | 10 lines Fix regression that changed behavior of queues when ringing a queue member. This reverts r298596, which was to fix a highly bizarre and contrived issue with a queue member that called into his own queue being transferred back into his own queue. I couldn't reproduce that issue in any way. I think one of the other recent transfer fixes actually fixed this. (closes issue #18747) Reported by: vrban ........ ................ ................ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@308013 f38db490-d61c-443f-a65b-d21fe96a405b
2011-02-14Merged revisions 307750 via svnmerge from tilghman1-2/+12
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r307750 | tilghman | 2011-02-14 00:50:23 -0600 (Mon, 14 Feb 2011) | 23 lines Calling a gosub routine defined in AEL from Dial/Queue ceased to work. A bug in AEL did not distinguish between the "s" extension generated by AEL and an "s" extension that was required to exist by the chan_dahdi (or another channel) that was not supplied with a starting extension. Therefore, AEL made incorrect assumptions about what commands were permissable in the context. This was fixed by making AEL generate a different extension name. However, Dial and Queue make additional assumptions about the name of the default gosub extension. Therefore, they needed to be brought into line with a "macro" rendered by AEL (as a gosub), without breaking traditional dialplans written without the aid of AEL. Related to (issue #18480) Reported by: nivek (closes issue #18729) Reported by: kkm Patches: 20110209__issue18729.diff.txt uploaded by tilghman (license 14) 018729-dial-queue-gosub-try3.patch uploaded by kkm (license 888) Tested by: kkm ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@307751 f38db490-d61c-443f-a65b-d21fe96a405b
2011-02-04Merged revisions 306356 via svnmerge from qwell1-0/+1
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r306356 | qwell | 2011-02-04 13:24:29 -0600 (Fri, 04 Feb 2011) | 16 lines Merged revisions 306346 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ........ r306346 | qwell | 2011-02-04 13:21:43 -0600 (Fri, 04 Feb 2011) | 9 lines Don't fallthrough to 'unknown' in the 'ringing' case. This could cause improper exits from the queue. (closes issue #18499) Reported by: zaltar Patches: app_queue.patch uploaded by zaltar (license 1148) ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@306359 f38db490-d61c-443f-a65b-d21fe96a405b
2011-02-04Merged revisions 306324 via svnmerge from rmudgett1-1/+1
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r306324 | rmudgett | 2011-02-04 12:53:06 -0600 (Fri, 04 Feb 2011) | 9 lines Don't send redirecting updates to the caller if the dialplan forked the call. Each fork in the dial could be redirected and confuse the caller. For ISDN the DivLeg1 and DivLeg3 messages would get confused because ISDN redirects calls in sequence not in parallel. * Also fixed a formatting inconsistency in app_dial.c and make a warning message more useful about what frame type could not be written. ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@306325 f38db490-d61c-443f-a65b-d21fe96a405b
2011-01-30Add Function and Application Relationships to documentationlathama1-3/+161
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: http://svn.digium.com/svn/asterisk/trunk@304913 f38db490-d61c-443f-a65b-d21fe96a405b
2011-01-20Merged revisions 303009 via svnmerge from jpeeler1-4/+8
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r303009 | jpeeler | 2011-01-20 11:10:32 -0600 (Thu, 20 Jan 2011) | 21 lines Merged revisions 303008 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r303008 | jpeeler | 2011-01-20 11:07:44 -0600 (Thu, 20 Jan 2011) | 14 lines Merged revisions 303007 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r303007 | jpeeler | 2011-01-20 11:04:08 -0600 (Thu, 20 Jan 2011) | 8 lines Add new queue strategy to preserve behavior for when queue members moved to ao2. Add queue strategy called "rrordered" to mimic old behavior from when queue members were stored in a linked list. ABE-2707 ........ ................ ................ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@303011 f38db490-d61c-443f-a65b-d21fe96a405b
2010-12-16Merged revisions 298598 via svnmerge from jpeeler1-0/+1
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r298598 | jpeeler | 2010-12-16 14:51:44 -0600 (Thu, 16 Dec 2010) | 21 lines Merged revisions 298597 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.6.2 ................ r298597 | jpeeler | 2010-12-16 14:49:33 -0600 (Thu, 16 Dec 2010) | 14 lines Merged revisions 298596 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r298596 | jpeeler | 2010-12-16 14:46:52 -0600 (Thu, 16 Dec 2010) | 7 lines Fix improper hangup when doing an attended transfer to queue. Had to indicate ringing in wait_for_answer so the attended transfer code would not try and hang up the local channel it created, which would kill the call. ABE-2624 ........ ................ ................ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@298599 f38db490-d61c-443f-a65b-d21fe96a405b
2010-11-19Merged revisions 295670 via svnmerge from bbryant1-4/+9
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r295670 | bbryant | 2010-11-19 16:40:21 -0500 (Fri, 19 Nov 2010) | 8 lines Patch for deadlock from ordering issue between channel/queue locks in app_queue (set_queue_variables). (closes issue #18031) Reported by: rain Review: https://reviewboard.asterisk.org/r/1018/ ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@295671 f38db490-d61c-443f-a65b-d21fe96a405b
2010-09-21Merged revisions 288079-288080 via svnmerge from rmudgett1-9/+8
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r288079 | rmudgett | 2010-09-21 15:29:51 -0500 (Tue, 21 Sep 2010) | 2 lines Protect channel access in CONNECTED_LINE and REDIRECTING interception macro launch code. ........ r288080 | rmudgett | 2010-09-21 15:29:59 -0500 (Tue, 21 Sep 2010) | 8 lines 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: http://svn.digium.com/svn/asterisk/trunk@288081 f38db490-d61c-443f-a65b-d21fe96a405b
2010-09-17Merged revisions 287388 via svnmerge from tilghman1-4/+4
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r287388 | tilghman | 2010-09-17 16:08:54 -0500 (Fri, 17 Sep 2010) | 21 lines 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: http://svn.digium.com/svn/asterisk/trunk@287389 f38db490-d61c-443f-a65b-d21fe96a405b
2010-09-16Merged revisions 287193 via svnmerge from russell1-2/+2
https://origsvn.digium.com/svn/asterisk/branches/1.8 ........ r287193 | russell | 2010-09-16 16:57:51 -0500 (Thu, 16 Sep 2010) | 4 lines Set the default for "autofill" and "shared_lastcall" to "yes" in queues.conf. Review: https://reviewboard.asterisk.org/r/922/ ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@287194 f38db490-d61c-443f-a65b-d21fe96a405b
2010-09-02Merged revisions 284632 via svnmerge from tilghman1-1/+1
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r284632 | tilghman | 2010-09-02 00:31:02 -0500 (Thu, 02 Sep 2010) | 14 lines 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: http://svn.digium.com/svn/asterisk/trunk@284633 f38db490-d61c-443f-a65b-d21fe96a405b
2010-09-02Merged revisions 284610 via svnmerge from tilghman1-1/+2
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: http://svn.digium.com/svn/asterisk/trunk@284628 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-28Merged revisions 280161 via svnmerge from seanbright1-0/+1
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r280161 | seanbright | 2010-07-28 12:52:12 -0400 (Wed, 28 Jul 2010) | 15 lines 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: http://svn.digium.com/svn/asterisk/trunk@280162 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-23Merged revisions 279227 via svnmerge from rmudgett1-3/+6
https://origsvn.digium.com/svn/asterisk/branches/1.8 ................ r279227 | rmudgett | 2010-07-23 17:20:47 -0500 (Fri, 23 Jul 2010) | 21 lines 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: http://svn.digium.com/svn/asterisk/trunk@279245 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-20Add load priority order, such that preload becomes unnecessary in most casestilghman1-1/+2
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@278132 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-16Fix reporting estimated queue hold time.jpeeler1-1/+1
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: http://svn.digium.com/svn/asterisk/trunk@277488 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-16Add missing handling for ringing state for use with queue empty options.jpeeler1-0/+5
(closes issue #17471) Reported by: jazzy Patches: app_queue.c.diff uploaded by jazzy (license 1056) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@277366 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-16Add a dialplan function to check if a queue exists: QUEUE_EXISTSoej1-0/+41
Review: https://reviewboard.asterisk.org/r/777/ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@276950 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-14ast_callerid restructuringrmudgett1-46/+56
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: http://svn.digium.com/svn/asterisk/trunk@276347 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-08Implement AstData API data providers as part of the GSOC 2010 project,eliel1-49/+40
midterm evaluation. Review: https://reviewboard.asterisk.org/r/757/ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@274727 f38db490-d61c-443f-a65b-d21fe96a405b
2010-07-02The switch fallthrough could create some errorneous situations, so best to ↵tilghman1-0/+5
force directly to the default case. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@273714 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-23Merged revisions 272367 via svnmerge from mnicholson1-1/+5
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: http://svn.digium.com/svn/asterisk/trunk@272368 f38db490-d61c-443f-a65b-d21fe96a405b
2010-06-02Generic Advice of Charge.rmudgett1-0/+30
Asterisk Generic AOC Representation - Generic AOC encode/decode routines. (Generic AOC must be encoded to be passed on the wire in the AST_CONTROL_AOC frame) - AST_CONTROL_AOC frame type to represent generic encoded AOC data - Manager events for AOC-S, AOC-D, and AOC-E messages Asterisk App Support - app_dial AOC-S pass-through support on call setup - app_queue AOC-S pass-through support on call setup AOC Unit Tests - AOC Unit Tests for encode/decode routines - AOC Unit Test for manager event representation. SIP AOC Support - Pass-through of generic AOC-D and AOC-E messages to snom phones via the snom AOC specification. - Creation of chan_sip page3 flags for the addition of the new 'snom_aoc_enabled' sip.conf option. IAX AOC Support - Natively supports AOC pass-through through the use of the new AST_CONTROL_AOC frame type DAHDI AOC Support - ETSI PRI full AOC Pass-through support - 'aoc_enable' chan_dahdi.conf option for independently enabling pass-through of AOC-S, AOC-D, AOC-E. - 'aoce_delayhangup' option for retrieving AOC-E on disconnect. - DAHDI A() dial string option for requesting AOC services. example usage: ;requests AOC-S, AOC-D, and AOC-E on call setup exten=>1111,1,Dial(DAHDI/g1/1112/A(s,d,e)) Review: https://reviewboard.asterisk.org/r/552/ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@267096 f38db490-d61c-443f-a65b-d21fe96a405b
2010-05-25Merged revisions 265610 via svnmerge from mnicholson1-7/+0
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r265610 | mnicholson | 2010-05-25 11:48:19 -0500 (Tue, 25 May 2010) | 8 lines Don't mark the cdr records of unanswered queue calls with "NOANSWER". This restores the behavior prior to r258670. (closes issue #17334) Reported by: jvandal Patches: queue-cdr-fixes1.diff uploaded by mnicholson (license 96) Tested by: aragon, jvandal ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@265611 f38db490-d61c-443f-a65b-d21fe96a405b
2010-05-21Merged revisions 265089 via svnmerge from mmichelson1-0/+4
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r265089 | mmichelson | 2010-05-21 15:59:14 -0500 (Fri, 21 May 2010) | 8 lines Don't hang up on a queue caller if the file we attempt to play does not exist. This also fixes a documentation mistake in file.h that made my original attempt to correct this problem not work correctly. (closes issue #17061) Reported by: RoadKill ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@265090 f38db490-d61c-443f-a65b-d21fe96a405b
2010-05-20Dial and queue connected line update macro not always run when expected.rmudgett1-29/+49
The connected line update macro would not get run if the connected line number string was empty. The number could be empty if the connected line update did not update a number but the name. It should be run if there was an AST_CONTROL_CONNECTED_LINE frame received for pending dials and queues. Renamed and added some more comments for some confusing identifiers directly connected to the related code. Also fixed a memory leak in app_queue. Review: https://reviewboard.asterisk.org/r/669/ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@264669 f38db490-d61c-443f-a65b-d21fe96a405b
2010-05-17Enhancements to connected line and redirecting work.mmichelson1-2/+19
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: http://svn.digium.com/svn/asterisk/trunk@263541 f38db490-d61c-443f-a65b-d21fe96a405b
2010-05-05'queue reset stats' erroneously clears wrapuptime configuration.pabelanger1-1/+1
Resets each member's lastcall to 0 now. (closes issue #17262) Reported by: rain Patches: wrapuptime_reset_fix.diff uploaded by rain (license 327) Tested by: rain git-svn-id: http://svn.digium.com/svn/asterisk/trunk@261232 f38db490-d61c-443f-a65b-d21fe96a405b
2010-05-04Add new possible value to autopause option to allow members to be autopaused ↵mmichelson1-5/+54
in all queues. See the CHANGES file and queues.conf.sample for more details. (closes issue #17008) Reported by: jlpedrosa Patches: queues.autopause_en_review.diff uploaded by jlpedrosa (license 1002) Review: https://reviewboard.asterisk.org/r/581/ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@261051 f38db490-d61c-443f-a65b-d21fe96a405b
2010-04-30Fix logic reversal error when queue callers join the queue.mmichelson1-1/+1
When a specific position is specified for the queue, the idea was that the caller cannot be placed ahead of higher-priority callers. Unfortunately, the logic was reversed so that the caller could ONLY be placed ahead of higher priority callers. Discovered while writing a unit test. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@260344 f38db490-d61c-443f-a65b-d21fe96a405b
2010-04-22Asterisk data retrieval API.eliel1-0/+280
This module implements an abstraction for retrieving and exporting asterisk data. Developed by: Brett Bryant <brettbryant@gmail.com> Eliel C. Sardanons (LU1ALY) <eliels@gmail.com> For the Google Summer of code 2009 Project. Documentation can be found in doxygen format and inside the header include/asterisk/data.h Review: https://reviewboard.asterisk.org/r/275/ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@258517 f38db490-d61c-443f-a65b-d21fe96a405b
2010-04-03Consolidate ast_channel.cid.cid_rdnis into ast_channel.redirecting.from.number.rmudgett1-16/+12
SWP-1229 ABE-2161 * Ensure chan_local.c:local_call() will not leak cid.cid_dnid when copying. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@256104 f38db490-d61c-443f-a65b-d21fe96a405b
2010-03-23Remove unused structure member in app_queue.seanbright1-2/+0
(closes issue #15494) Reported by: makoto git-svn-id: http://svn.digium.com/svn/asterisk/trunk@254045 f38db490-d61c-443f-a65b-d21fe96a405b
2010-03-03Removed cdrflags from ast_channel structure.rmudgett1-1/+0
Only chan_dahdi set a value in cdrflags. Everyone else just copied it around the system. Noone cared about any value it may have contained. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@250565 f38db490-d61c-443f-a65b-d21fe96a405b
2010-02-18fixes Queue with C option crashdvossel1-1/+1
(closes issue #16475) Reported by: okrief Patches: queue_crash.diff uploaded by dvossel (license 671) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@247736 f38db490-d61c-443f-a65b-d21fe96a405b
2010-02-17Merged revisions 247168 via svnmerge from mmichelson1-3/+5
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r247168 | mmichelson | 2010-02-17 10:24:17 -0600 (Wed, 17 Feb 2010) | 3 lines Make sure that when autofill is disabled that callers not in the front of the queue cannot place calls. ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@247169 f38db490-d61c-443f-a65b-d21fe96a405b
2010-02-10Merged revisions 246115 via svnmerge from dvossel1-6/+7
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r246115 | dvossel | 2010-02-10 11:44:20 -0600 (Wed, 10 Feb 2010) | 8 lines fixes random deadlock in app_queue with use_weight during reload (closes issue #16677) Reported by: tim_ringenbach Patches: app_queue_use_weight_deadlock.diff uploaded by tim ringenbach (license 540) ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@246116 f38db490-d61c-443f-a65b-d21fe96a405b
2010-01-27Merged revisions 243691 via svnmerge from jpeeler1-1/+1
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r243691 | jpeeler | 2010-01-27 14:35:56 -0600 (Wed, 27 Jan 2010) | 5 lines Revert 243570, I should have looked at this closer. Will reopen the issue, but am leaving the review closed as the change was pointless. (issue #16488) ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@243693 f38db490-d61c-443f-a65b-d21fe96a405b
2010-01-27Merged revisions 243570 via svnmerge from jpeeler1-1/+1
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r243570 | jpeeler | 2010-01-27 12:47:34 -0600 (Wed, 27 Jan 2010) | 9 lines Extend announcement URL used with Queue from 80 chars to PATH_MAX. (closes issue #16488) Reported by: syspert Patches: soundfilelen.pacth-2 uploaded by syspert (license 938) Review: https://reviewboard.asterisk.org/r/475/ ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@243571 f38db490-d61c-443f-a65b-d21fe96a405b
2010-01-18fixes spelling error. s/memeber/memberdvossel1-2/+2
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@240842 f38db490-d61c-443f-a65b-d21fe96a405b
2010-01-07cli 'queue show' formatting fix. queue name was truncated over 12 charactersdvossel1-1/+1
(closes issue #16078) Reported by: RoadKill Patches: quequename_limit.patch uploaded by ppyy (license 906) Tested by: dvossel git-svn-id: http://svn.digium.com/svn/asterisk/trunk@238361 f38db490-d61c-443f-a65b-d21fe96a405b
2010-01-05fixes holdtime playback issue in app_queuedvossel1-5/+2
When reporting hold time, the number of seconds should be mod 60. Otherwise audio playback could be something like "2 minutes 123 seconds" rather than "2 minutes 3 seconds". Also, the "minute" sound file is missing, so for the moment until that file can be created the "minutes" file is used instead. (closes issue #16168) Reported by: nickilo Patches: patch-unified-trunk-rev-222176 uploaded by nickilo (license ) Tested by: nickilo, wonderg git-svn-id: http://svn.digium.com/svn/asterisk/trunk@237920 f38db490-d61c-443f-a65b-d21fe96a405b
2010-01-04app_queue segfaults if realtime field uniqueid is NULLdvossel1-0/+5
(closes issue #16385) Reported by: haakon Patches: app_queue.c.patch uploaded by haakon (license 880) app_queue.c.patch_v2 uploaded by dvossel (license 671) Tested by: haakon git-svn-id: http://svn.digium.com/svn/asterisk/trunk@237327 f38db490-d61c-443f-a65b-d21fe96a405b
2009-12-23QUEUE_MEMBER(..., ready) counts only ready agents, not free agents wrapping updvossel1-3/+19
The QUEUE_MEMBER dialplan function can return total members, logged-in members and "free" members count. A member is counted as "free" immediately after his call ends, even though its wrap-up time, if specified in queues.conf, has not yet expired, and the queue will not actually route a call to it. This Patch introduces a new "ready" option that only counts free agents no longer in the wrap up time period. (closes issue #16240) Reported by: kkm Patches: appqueue-memberfun-readyoption-trunk.diff uploaded by kkm (license 888) Tested by: kkm, dvossel git-svn-id: http://svn.digium.com/svn/asterisk/trunk@236308 f38db490-d61c-443f-a65b-d21fe96a405b
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