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${ noResults }
355 Commits (b148e15762f39bded9b10aee463cdf906efa1e8c)
Author | SHA1 | Message | Date |
---|---|---|---|
|
f089bbf663 |
Goodbye Zaptel, hello DAHDI. Removes Zaptel driver support with DAHDI. Configuration file and dialplan backwards compatability has been put in place where appropiate. Release announcement to follow.
git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@123332 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
3a1b062a16 |
Merged revisions 123165 via svnmerge from
https://origsvn.digium.com/svn/asterisk/trunk ........ r123165 | murf | 2008-06-16 14:43:46 -0600 (Mon, 16 Jun 2008) | 19 lines (closes issue #12689) Reported by: ys Many thanks to ys for doing the research on this problem. I didn't think it would be best to unlock the contexts and then relock them after the remove_extension2() call, so I added an extra arg to remove_extension2() and set it appropriately in each call. There were not that many. I considered forcing the code to lock the contexts before the call to remove_extension2(), but that would require a slightly greater degree of changes, especially since the find_context_locked is local to pbx.c I did a simple sanity test to make sure the code doesn't mess things up in general. ........ git-svn-id: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@123173 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
e60f5750ad |
Merged revisions 119531 via svnmerge from
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: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@119532 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
726f081141 |
Merged revisions 119479 via svnmerge from
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: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@119529 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
caab83a81b |
Merged revisions 119296 via svnmerge from
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: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@119297 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
|
96e74f94f0 |
Merged revisions 114113 via svnmerge from
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: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@114114 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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1b56c4ce2d |
Merged revisions 107017 via svnmerge from
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: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@107018 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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90046983f7 |
Merged revisions 106239 via svnmerge from
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: https://origsvn.digium.com/svn/asterisk/branches/1.6.0@106318 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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7d1891d5c3 |
Asterisk, when parking can drop rights a caller when a parking timeout 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: https://origsvn.digium.com/svn/asterisk/trunk@105477 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
17 years ago |
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4dccb58fb7 |
whitespace fixes only.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@103249 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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fe9821cc10 |
Get rid of any remaining ast_verbose calls in the code in favor of
ast_verb (closes issue #11934) Reported by: mvanbaak Patches: 20080205_astverb-2.diff.txt uploaded by mvanbaak (license 7) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@102525 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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cc648a40ae |
Merged revisions 99592 via svnmerge from
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: https://origsvn.digium.com/svn/asterisk/trunk@99595 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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d5b454bf8d |
Convert ast_verbose to ast_verb.
Reported by: snuffy Patch by: snuffy (Closes issue #11547) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@92913 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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99308dfb4e |
Conversions of free to ast_free, where applicable, and several other formatting fixes.
Reported by: eliel Patch by: eliel,tilghman (Closes issue #11209) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@92594 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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3a4d1c852b |
Merged revisions 91783 via svnmerge from
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: https://origsvn.digium.com/svn/asterisk/trunk@91784 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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547083e21a |
Merged revisions 91693 via svnmerge from
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: https://origsvn.digium.com/svn/asterisk/trunk@91700 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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c72fa81580 |
Merged revisions 91677 via svnmerge from
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: https://origsvn.digium.com/svn/asterisk/trunk@91678 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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807d5e1ef7 |
- Dial event
- 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: https://origsvn.digium.com/svn/asterisk/trunk@91407 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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b32e39cbda |
Merged revisions 91273 via svnmerge from
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: https://origsvn.digium.com/svn/asterisk/trunk@91291 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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814a7f66c0 |
Fix build in trunk. This was fixed in 1.4, but blocked in trunk since this hadn't been merged yet.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@90877 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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c52d8a1cd5 |
Merged revisions 90735 via svnmerge from
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: https://origsvn.digium.com/svn/asterisk/trunk@90873 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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4201a5af8b |
Remove the file descriptors from the main poll channel when the channel is 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: https://origsvn.digium.com/svn/asterisk/trunk@90508 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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6b08c442c7 |
Adding support for the "automixmonitor" dial and queue options.
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: https://origsvn.digium.com/svn/asterisk/trunk@90388 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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4d8932a6dc |
Merged revisions 89622 via svnmerge from
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: https://origsvn.digium.com/svn/asterisk/trunk@89623 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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86476c607f |
closes issue #11285, where an unload of a module that creates a dialplan 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: https://origsvn.digium.com/svn/asterisk/trunk@89513 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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7e8835e0d7 |
remove another set of redundant #include "asterisk/options.h"
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89512 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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a23c055c3d |
move asterisk/paths.h outside asterisk.h and into those files
who really need it. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89466 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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0595b5e2aa |
include "logger.h" and errno.h from asterisk.h - usage shows that they
were included almost everywhere. Remove some of the instances. git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89424 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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fdb7f7ba3d |
Start untangling header inclusion in a way that does not affect
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: https://origsvn.digium.com/svn/asterisk/trunk@89333 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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0df5e50e97 |
Use BEGIN_OPTIONS / END_OPTIONS to make the syntax highlighting in my editor happy
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@89264 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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63f2f04cf4 |
This commits the performance mods that give the priority processing engine 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: https://origsvn.digium.com/svn/asterisk/trunk@88166 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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a4be521c89 |
Make sure we propogate ANI2 to the outbound channel
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@85499 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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7adbd6bb16 |
Remove redundant includes (patch by snuffy) (Closes issue #10922)
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@85140 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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bff784d509 |
Merged revisions 84166 via svnmerge from
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: https://origsvn.digium.com/svn/asterisk/trunk@84167 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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3ed4d505b7 |
Merged revisions 84158 via svnmerge from
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: https://origsvn.digium.com/svn/asterisk/trunk@84159 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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9388173f85 |
Make the MALLOC_DEBUG output for free() useful again. After changing calls to
free to be ast_free, astmm said all calls to free were coming from utils.h git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@82628 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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836c550ce3 |
Merged revisions 81412 via svnmerge from
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: https://origsvn.digium.com/svn/asterisk/trunk@81413 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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22114b509d |
Add support for using epoll instead of poll. This should increase scalability and is done in such a way that we should be able to add support for other poll() replacements.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@78683 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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9ef1b0a974 |
Extend the ast_senddigit and ast_dtmf_stream API calls to allow the duration of the DTMF digit(s) to be specified and make the SendDTMF application have the capability to use it.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@78278 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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4e0947c5f1 |
Convert code that checks the _softhangup member of ast_channel directory to use
the ast_check_hangup() funciton. This function takes scheduled hangups into account. (closes issue #10230, patch by Juggie) git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@77858 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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ceca4d97e1 |
These fixes take care of two problems: a complaint in asterisk-dev that 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: https://origsvn.digium.com/svn/asterisk/trunk@77520 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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f8483a0d04 |
Do a massive conversion for using the ast_verb() macro
(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: https://origsvn.digium.com/svn/asterisk/trunk@77299 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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55b1ee298e |
Merge the dialplan_aesthetics branch. Most of this patch simply converts 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: https://origsvn.digium.com/svn/asterisk/trunk@76703 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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0e969271ae |
After some study, thought, comparing, etc. I've backed out the previous 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: https://origsvn.digium.com/svn/asterisk/trunk@75983 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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ee6d59eef2 |
Merged revisions 75405 via svnmerge from
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: https://origsvn.digium.com/svn/asterisk/trunk@75406 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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8a7732f067 |
via 10206, I have added an option (e) to Dial to allow the h exten to get run on peer. Had to upgrade ast_flag stuff to 64 bits to do this.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@75400 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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766121a5bc |
Fix an incorrect parenthesization (TODO: Find a better word) in app_dial
Pointed out by Fanzhou Zhao Closes issue #10216 git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@75351 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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ce8f95d750 |
Merged revisions 75253 via svnmerge from
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: https://origsvn.digium.com/svn/asterisk/trunk@75254 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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b8cd949cce |
Applications no longer need to call ast_module_user_add and ast_module_user_remove. This is now taken care of in the pbx_exec function outside of the application.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@75200 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |
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96a646734f |
It is no longer required for each module that deals with a channel to call ast_module_user_hangup_all in it's unload function. The loader will automatically perform this action for it.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@75183 65c4cc65-6c06-0410-ace0-fbb531ad65f3 |
18 years ago |