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2008-06-02Merged revisions 119531 via svnmerge from russell1-1/+1
https://origsvn.digium.com/svn/asterisk/trunk ................ r119531 | russell | 2008-06-01 20:04:01 -0500 (Sun, 01 Jun 2008) | 10 lines Merged revisions 119530 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r119530 | russell | 2008-06-01 20:03:22 -0500 (Sun, 01 Jun 2008) | 2 lines Fix another typo in documentation ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@119532 f38db490-d61c-443f-a65b-d21fe96a405b
2008-06-01Merged revisions 119479 via svnmerge from mvanbaak1-1/+1
https://origsvn.digium.com/svn/asterisk/trunk ................ r119479 | mvanbaak | 2008-06-01 23:06:27 +0200 (Sun, 01 Jun 2008) | 10 lines Merged revisions 119478 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r119478 | mvanbaak | 2008-06-01 22:47:55 +0200 (Sun, 01 Jun 2008) | 2 lines small typo fix 'retires' => 'retries' ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@119529 f38db490-d61c-443f-a65b-d21fe96a405b
2008-05-30Merged revisions 119296 via svnmerge from tilghman1-4/+4
https://origsvn.digium.com/svn/asterisk/trunk ........ r119296 | tilghman | 2008-05-30 11:10:46 -0500 (Fri, 30 May 2008) | 8 lines Add native AGI command GOSUB, as invoking Gosub with EXEC does not work properly. (closes issue #12760) Reported by: Corydon76 Patches: 20080530__bug12760.diff.txt uploaded by Corydon76 (license 14) Tested by: tim_ringenbach, Corydon76 ........ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@119297 f38db490-d61c-443f-a65b-d21fe96a405b
2008-04-14Merged revisions 114113 via svnmerge from mmichelson1-2/+8
https://origsvn.digium.com/svn/asterisk/trunk ................ r114113 | mmichelson | 2008-04-14 11:25:09 -0500 (Mon, 14 Apr 2008) | 17 lines Merged revisions 114112 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r114112 | mmichelson | 2008-04-14 11:24:22 -0500 (Mon, 14 Apr 2008) | 9 lines If the datastore has been moved to another channel due to a masquerade, then freeing the datastore here causes an eventual double free when the new channel hangs up. We should only free the datastore if we were able to successfully remove it from the channel we are referencing (i.e. the datastore was not moved). (closes issue #12359) Reported by: pguido ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@114114 f38db490-d61c-443f-a65b-d21fe96a405b
2008-03-10Merged revisions 107017 via svnmerge from file1-17/+0
https://origsvn.digium.com/svn/asterisk/trunk ................ r107017 | file | 2008-03-10 11:36:16 -0300 (Mon, 10 Mar 2008) | 15 lines Merged revisions 107016 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r107016 | file | 2008-03-10 11:33:02 -0300 (Mon, 10 Mar 2008) | 7 lines Move where unanswered CDRs are dropped to the CDR core, not everything uses app_dial. (closes issue #11516) Reported by: ys Patches: branch_1.4_cdr.diff uploaded by ys (license 281) Tested by: anest, jcapp, dartvader ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@107018 f38db490-d61c-443f-a65b-d21fe96a405b
2008-03-06Merged revisions 106239 via svnmerge from russell1-1/+6
https://origsvn.digium.com/svn/asterisk/trunk ................ r106239 | file | 2008-03-05 16:43:22 -0600 (Wed, 05 Mar 2008) | 12 lines Merged revisions 106235 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r106235 | file | 2008-03-05 18:32:10 -0400 (Wed, 05 Mar 2008) | 4 lines Add a control frame to indicate the source of media has changed. Depending on the underlying technology it may need to change some things. (closes issue #12148) Reported by: jcomellas ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/branches/1.6.0@106318 f38db490-d61c-443f-a65b-d21fe96a405b
2008-03-01Asterisk, when parking can drop rights a caller when a parking timeout ↵twilson1-0/+64
occurs. Also, when doing built-in attended transfers, sometimes incorrectly passes rights from the transferrer to the transferee. This patch tries to fixes the parking issue and lays some groundwork for later fixing the transfer issue. (closes issue #11520) Reported by: pliew Tested by: otherwiseguy git-svn-id: http://svn.digium.com/svn/asterisk/trunk@105477 f38db490-d61c-443f-a65b-d21fe96a405b
2008-02-09whitespace fixes only.mvanbaak1-179/+170
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@103249 f38db490-d61c-443f-a65b-d21fe96a405b
2008-02-05Get rid of any remaining ast_verbose calls in the code in favor of mmichelson1-4/+2
ast_verb (closes issue #11934) Reported by: mvanbaak Patches: 20080205_astverb-2.diff.txt uploaded by mvanbaak (license 7) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@102525 f38db490-d61c-443f-a65b-d21fe96a405b
2008-01-22Merged revisions 99592 via svnmerge from oej1-0/+5
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r99592 | oej | 2008-01-22 18:31:17 +0100 (Tis, 22 Jan 2008) | 5 lines Add dependency on chan_local to app_dial. Dial still runs without chan_local, but will be missing forwarding functionality. ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@99595 f38db490-d61c-443f-a65b-d21fe96a405b
2007-12-14Convert ast_verbose to ast_verb.tilghman1-14/+9
Reported by: snuffy Patch by: snuffy (Closes issue #11547) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@92913 f38db490-d61c-443f-a65b-d21fe96a405b
2007-12-12Conversions of free to ast_free, where applicable, and several other ↵tilghman1-81/+83
formatting fixes. Reported by: eliel Patch by: eliel,tilghman (Closes issue #11209) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@92594 f38db490-d61c-443f-a65b-d21fe96a405b
2007-12-07Merged revisions 91783 via svnmerge from russell1-26/+43
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r91783 | russell | 2007-12-07 10:38:48 -0600 (Fri, 07 Dec 2007) | 6 lines * Add channel locking around datastore operations that expect the channel to be locked. * Document why we don't record Local channels in the dialed interfaces list. * Remove the dialed variable as it isn't needed. * Restructure some code for clarity and coding guidelines stuff ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@91784 f38db490-d61c-443f-a65b-d21fe96a405b
2007-12-07Merged revisions 91693 via svnmerge from russell1-1/+1
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r91693 | russell | 2007-12-06 20:51:22 -0600 (Thu, 06 Dec 2007) | 2 lines Don't unlock the dialed_interfaces list until we're done messing with the iterator. ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@91700 f38db490-d61c-443f-a65b-d21fe96a405b
2007-12-07Merged revisions 91677 via svnmerge from russell1-1/+1
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r91677 | russell | 2007-12-06 20:38:40 -0600 (Thu, 06 Dec 2007) | 4 lines Allow dialing local channels from Queue() and Dial() again. There was a slight flaw in the code to prevent call forwards from looping that caused this problem. (related to issue #11486) ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@91678 f38db490-d61c-443f-a65b-d21fe96a405b
2007-12-06- Dial eventoej1-8/+10
- Event Dial has new headers, to comply with other events - Source -> Channel Channel name (caller) - SrcUniqueID -> UniqueID Uniqueid (new) -> Dialstring Dialstring in app data (moremanager) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@91407 f38db490-d61c-443f-a65b-d21fe96a405b
2007-12-05Merged revisions 91273 via svnmerge from mmichelson1-2/+4
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r91273 | mmichelson | 2007-12-05 16:35:52 -0600 (Wed, 05 Dec 2007) | 11 lines The 'G' option for Dial() did not properly handle the case where only a label was provided. This was due to the fact that the answering channel did not have an extension set, so ast_parseable_goto would fail. This fix eliminates the call to ast_parseable_goto on the answering channel since it is a wasteful call. The answering channel and the calling channel are both directed to the same extension and context, just different priorities, so we can just copy the values from the calling channel to the answering channel and increment the answering channel's priority. (closes issue #11382, reported by jon, patch by me with correction by jon) ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@91291 f38db490-d61c-443f-a65b-d21fe96a405b
2007-12-04Fix build in trunk. This was fixed in 1.4, but blocked in trunk since this ↵qwell1-1/+1
hadn't been merged yet. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@90877 f38db490-d61c-443f-a65b-d21fe96a405b
2007-12-04Merged revisions 90735 via svnmerge from mmichelson1-69/+75
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r90735 | mmichelson | 2007-12-03 17:12:17 -0600 (Mon, 03 Dec 2007) | 22 lines A big one... This is the merge of the forward-loop branch. The main change here is that call-forwards can no longer loop. This is accomplished by creating a datastore on the calling channel which has a linked list of all devices dialed. If a forward happens, then the local channel which is created inherits the datastore. If, through this progression of forwards and datastore inheritance, a device is attempted to be dialed a second time, it will simply be skipped and a warning message will be printed to the CLI. After the dialing has been completed, the datastore is detached from the channel and destroyed. This change also introduces some side effects to the code which I shall enumerate here: 1. Datastore inheritance has been backported from trunk into 1.4 2. A large chunk of code has been removed from app_dial. This chunk is the section of code which handles the call forward case after the channel has been requested but before it has been called. This was removed because call-forwarding still works fine without it, it makes the code less error-prone should it need changing, and it made this set of changes much less painful to just have the forwarding handled in one place in each module. 3. Two new files, global_datastores.h and .c have been added. These are necessary since the datastore which is attached to the channel may be created and attached in either app_dial or app_queue, so they need a common place to find the datastore info. This approach was taken in case similar datastores are needed in the future, there will be a common place to add them. ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@90873 f38db490-d61c-443f-a65b-d21fe96a405b
2007-12-03Remove the file descriptors from the main poll channel when the channel is ↵file1-2/+7
hung up during the dialing attempt, and make sure a channel exists before trying to remove it at the end. (closes issue #11441) Reported by: blitzrage git-svn-id: http://svn.digium.com/svn/asterisk/trunk@90508 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-30Adding support for the "automixmonitor" dial and queue options.mmichelson1-4/+19
This works in much the same way as the automonitor, except that instead of using the monitor app, it uses the mixmonitor app. By providing an 'x' or 'X' as a dial or queue option, a DTMF sequence may be entered (as defined in features.conf) to start the one-touch mixmonitor. This patch also introduces some new API calls to the audiohooks code for searching for an audiohook by type and for searching for a running audiohook by type. Big thanks to joetester for writing the initial patch, testing it and patiently waiting for it to be committed. (closes issue #10185, reported and patched by xmarksthespot) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@90388 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-27Merged revisions 89622 via svnmerge from murf1-1/+21
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r89622 | murf | 2007-11-26 23:24:02 -0700 (Mon, 26 Nov 2007) | 1 line closes issue #11379; OK, this is an attempt to make both sides happy. To the cdr.conf file, I added the option 'unanswered', which defaults to 'no'. In this mode, you will see a cdr for a call, whether it was answered or not. The disposition will be NO ANSWER or ANSWERED, as appropriate. The src is as you'd expect, the destination channel will be one of the channels from the Dial() call, usually the last in the list if more than one chan was specified. With unanswered set to 'yes', you will still see this cdr entry in both cases. But in the case where the dial timed out, you will also see a cdr for each line attempted, marked NO ANSWER, with no destination channel name. The new option defaults to 'no', so you don't see the pesky extra cdr's by default, and you will not see the irritating 'not posted' messages. ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89623 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-21closes issue #11285, where an unload of a module that creates a dialplan ↵murf1-1/+4
context, causes a crash when you do a 'dialplan show' of that context. This is because the registrar string is defined in the module, and the stale pointer is traversed. The reporter offered a patch that would always strdup the registrar string, which is practical, but I preferred to destroy the created contexts in each module where one is created. That seemed more symmetric. There were only 6 place in asterisk where this is done: chan_sip, chan_iax2, chan_skinny, res_features, app_dial, and app_queue. The two apps destroyed the context, but left the contexts. All is fixed now and unloads should be dialplan friendly. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89513 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-21remove another set of redundant #include "asterisk/options.h"rizzo1-1/+0
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89512 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-20move asterisk/paths.h outside asterisk.h and into those filesrizzo1-0/+1
who really need it. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89466 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-19include "logger.h" and errno.h from asterisk.h - usage shows that theyrizzo1-2/+0
were included almost everywhere. Remove some of the instances. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89424 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-16Start untangling header inclusion in a way that does not affectrizzo1-4/+0
build times - tested, there is no measureable difference before and after this commit. In this change: use asterisk/compat.h to include a small set of system headers: inttypes.h, unistd.h, stddef.h, stddint.h, sys/types.h, stdarg.h, stdlib.h, alloca.h, stdio.h Where available, the inclusion is conditional on HAVE_FOO_H as determined by autoconf. Normally, source files should not include any of the above system headers, and instead use either "asterisk.h" or "asterisk/compat.h" which does it better. For the time being I have left alone second-level directories (main/db1-ast, etc.). git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89333 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-14Use BEGIN_OPTIONS / END_OPTIONS to make the syntax highlighting in my editor ↵russell1-2/+2
happy git-svn-id: http://svn.digium.com/svn/asterisk/trunk@89264 f38db490-d61c-443f-a65b-d21fe96a405b
2007-11-01This commits the performance mods that give the priority processing engine ↵murf1-7/+7
in the pbx, a 25-30% speed boost. The two updates used, are, first, to merge the ast_exists_extension() and the ast_spawn_extension() where they are called sequentially in a loop in the code, into a slightly upgraded version of ast_spawn_extension(), with a few extra args; and, second, I modified the substitute_variables_helper_full, so it zeroes out the byte after the evaluated string instead of demanding you pre-zero the buffer; I also went thru the code and removed the code that zeroed this buffer before every call to the substitute_variables_helper_full. The first fix provides about a 9% speedup, and the second the rest. These figures come from the 'PIPS' benchmark I describe in blogs, conf. reports, etc. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@88166 f38db490-d61c-443f-a65b-d21fe96a405b
2007-10-11Make sure we propogate ANI2 to the outbound channelmattf1-0/+1
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@85499 f38db490-d61c-443f-a65b-d21fe96a405b
2007-10-09Remove redundant includes (patch by snuffy) (Closes issue #10922)tilghman1-1/+0
git-svn-id: http://svn.digium.com/svn/asterisk/trunk@85140 f38db490-d61c-443f-a65b-d21fe96a405b
2007-10-01Merged revisions 84166 via svnmerge from russell1-1/+3
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r84166 | russell | 2007-10-01 09:24:49 -0500 (Mon, 01 Oct 2007) | 2 lines Simplify the CAN_EARLY_BRIDGE macro a bit. ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@84167 f38db490-d61c-443f-a65b-d21fe96a405b
2007-10-01Merged revisions 84158 via svnmerge from file1-5/+8
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r84158 | file | 2007-10-01 10:49:36 -0300 (Mon, 01 Oct 2007) | 4 lines Only attempt early bridging if the options given to Dial() permit it. (closes issue #10861) Reported by: peekyb ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@84159 f38db490-d61c-443f-a65b-d21fe96a405b
2007-09-17Make the MALLOC_DEBUG output for free() useful again. After changing calls torussell1-1/+1
free to be ast_free, astmm said all calls to free were coming from utils.h git-svn-id: http://svn.digium.com/svn/asterisk/trunk@82628 f38db490-d61c-443f-a65b-d21fe96a405b
2007-08-31Merged revisions 81412 via svnmerge from qwell1-1/+3
https://origsvn.digium.com/svn/asterisk/branches/1.4 (closes issue #10621) ........ r81412 | qwell | 2007-08-31 13:44:44 -0500 (Fri, 31 Aug 2007) | 4 lines Re-order dial options to be in line with the existing alpha order. Issue 10621, initial patch by junky ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@81413 f38db490-d61c-443f-a65b-d21fe96a405b
2007-08-08Add support for using epoll instead of poll. This should increase ↵file1-1/+13
scalability and is done in such a way that we should be able to add support for other poll() replacements. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@78683 f38db490-d61c-443f-a65b-d21fe96a405b
2007-08-06Extend the ast_senddigit and ast_dtmf_stream API calls to allow the duration ↵file1-3/+3
of the DTMF digit(s) to be specified and make the SendDTMF application have the capability to use it. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@78278 f38db490-d61c-443f-a65b-d21fe96a405b
2007-08-01Convert code that checks the _softhangup member of ast_channel directory to userussell1-2/+2
the ast_check_hangup() funciton. This function takes scheduled hangups into account. (closes issue #10230, patch by Juggie) git-svn-id: http://svn.digium.com/svn/asterisk/trunk@77858 f38db490-d61c-443f-a65b-d21fe96a405b
2007-07-27These fixes take care of two problems: a complaint in asterisk-dev that ↵murf1-0/+5
goto's aren't working in trunk, a side effect of the move to commas as arg seps in apps and funcs; and a problem I spotted myself with dial's 'e' option, where gotos were off by one, because I forgot to set the AUTOLOOP flag in the peer channel. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@77520 f38db490-d61c-443f-a65b-d21fe96a405b
2007-07-26Do a massive conversion for using the ast_verb() macrorussell1-80/+48
(closes issue #10277, patches by mvanbaak) Basically, this changes ... if (option_verbose > 2) ast_verbose(VERBOSE_PREFIX_3, "Something\n"); to ... ast_verb(3, "Something\n"); git-svn-id: http://svn.digium.com/svn/asterisk/trunk@77299 f38db490-d61c-443f-a65b-d21fe96a405b
2007-07-23Merge the dialplan_aesthetics branch. Most of this patch simply converts ↵tilghman1-38/+30
applications using old methods of parsing arguments to using the standard macros. However, the big change is that the really old way of specifying application and arguments separated by a comma will no longer work (e.g. NoOp,foo|bar). Instead, the way that has been recommended since long before 1.0 will become the only method available (e.g. NoOp(foo,bar). git-svn-id: http://svn.digium.com/svn/asterisk/trunk@76703 f38db490-d61c-443f-a65b-d21fe96a405b
2007-07-19After some study, thought, comparing, etc. I've backed out the previous ↵murf1-89/+89
universal mod to make ast_flags a 64 bit thing. Instead, I added a 64-bit version of ast_flags (ast_flags64), and 64-bit versions of the test-flag, set-flag, etc. macros, and an app_parse_options64 routine, and I use these in app_dial alone, to eliminate the 30-option limit it had grown to meet. There is room now for 32 more options and flags. I was heavily tempted to implement some of the other ideas that were presented, but this solution does not intro any new versions of dial, doesn't have a different API, has a minimal/zero impact on code outside of dial, and doesn't seriously (I hope) affect the code structure of dial. It's the best I can think of right now. My goal was NOT to rewrite dial. I leave that to a future, coordinated effort. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@75983 f38db490-d61c-443f-a65b-d21fe96a405b
2007-07-17Merged revisions 75405 via svnmerge from mmichelson1-2/+2
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r75405 | mmichelson | 2007-07-17 15:03:48 -0500 (Tue, 17 Jul 2007) | 6 lines Fixing an error I made earlier. ast_fileexists can return -1 on failure, so I need to be sure that we only enter the if statement if it is successful. Related to my fix to issue #10186 ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@75406 f38db490-d61c-443f-a65b-d21fe96a405b
2007-07-17via 10206, I have added an option (e) to Dial to allow the h exten to get ↵murf1-5/+30
run on peer. Had to upgrade ast_flag stuff to 64 bits to do this. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@75400 f38db490-d61c-443f-a65b-d21fe96a405b
2007-07-17Fix an incorrect parenthesization (TODO: Find a better word) in app_dialqwell1-1/+1
Pointed out by Fanzhou Zhao Closes issue #10216 git-svn-id: http://svn.digium.com/svn/asterisk/trunk@75351 f38db490-d61c-443f-a65b-d21fe96a405b
2007-07-16Merged revisions 75253 via svnmerge from mmichelson1-5/+16
https://origsvn.digium.com/svn/asterisk/branches/1.4 ........ r75253 | mmichelson | 2007-07-16 13:16:15 -0500 (Mon, 16 Jul 2007) | 8 lines Restoring functionality from 1.2 wherein Retrydial will not exit if there is no announce file specified. This change makes it so that if there is no announce file specified, the application will continue until finished (or caller hangs up). If a bogus announce file is specified, then a warning message will be printed saying that the file could not be found, but execution will still continue. (closes issue #10186, reported by jon, patched by me) ........ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@75254 f38db490-d61c-443f-a65b-d21fe96a405b
2007-07-16Applications no longer need to call ast_module_user_add and ↵file1-10/+1
ast_module_user_remove. This is now taken care of in the pbx_exec function outside of the application. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@75200 f38db490-d61c-443f-a65b-d21fe96a405b
2007-07-16It is no longer required for each module that deals with a channel to call ↵file1-2/+0
ast_module_user_hangup_all in it's unload function. The loader will automatically perform this action for it. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@75183 f38db490-d61c-443f-a65b-d21fe96a405b
2007-07-09Implementation of a feature that will disable "missed calls" counters on SIP ↵oej1-5/+13
phones. If the call is answered by another phone, other phones won't display the call as "missed". You can also add an option to the dial command so that you can have a "followme" scenario and not count the calls as "missed" when you cancel the call. Thanks to Ramon and Frank for feedback on this feature. git-svn-id: http://svn.digium.com/svn/asterisk/trunk@74024 f38db490-d61c-443f-a65b-d21fe96a405b
2007-07-03Merged revisions 73053 via svnmerge from tilghman1-2/+2
https://origsvn.digium.com/svn/asterisk/branches/1.4 ................ r73053 | tilghman | 2007-07-03 07:38:53 -0500 (Tue, 03 Jul 2007) | 10 lines Merged revisions 73052 via svnmerge from https://origsvn.digium.com/svn/asterisk/branches/1.2 ........ r73052 | tilghman | 2007-07-03 07:34:14 -0500 (Tue, 03 Jul 2007) | 2 lines RetryDial should accept a 0 argument, but it does not, because atoi does not distinguish between 0 and error (closes issue #10106) ........ ................ git-svn-id: http://svn.digium.com/svn/asterisk/trunk@73054 f38db490-d61c-443f-a65b-d21fe96a405b
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