Steve Murphy
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.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@88166 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-11-01 22:26:51 +00:00
Matthew Fredrickson
a4be521c89
Make sure we propogate ANI2 to the outbound channel
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@85499 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-10-11 22:42:44 +00:00
Tilghman Lesher
7adbd6bb16
Remove redundant includes (patch by snuffy) (Closes issue #10922 )
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@85140 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-10-09 16:04:41 +00:00
Russell Bryant
bff784d509
Merged revisions 84166 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r84166 | russell | 2007-10-01 09:24:49 -0500 (Mon, 01 Oct 2007) | 2 lines
Simplify the CAN_EARLY_BRIDGE macro a bit.
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2007-10-01 14:27:02 +00:00
Joshua Colp
3ed4d505b7
Merged revisions 84158 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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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
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2007-10-01 13:53:09 +00:00
Russell Bryant
9388173f85
Make the MALLOC_DEBUG output for free() useful again. After changing calls to
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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
2007-09-17 18:57:56 +00:00
Jason Parker
836c550ce3
Merged revisions 81412 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4
(closes issue #10621 )
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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
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2007-08-31 18:46:02 +00:00
Joshua Colp
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.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@78683 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-08-08 21:44:58 +00:00
Joshua Colp
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.
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2007-08-06 21:52:30 +00:00
Russell Bryant
4e0947c5f1
Convert code that checks the _softhangup member of ast_channel directory to use
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the ast_check_hangup() funciton. This function takes scheduled hangups into
account.
(closes issue #10230 , patch by Juggie)
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2007-08-01 15:39:54 +00:00
Steve Murphy
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.
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2007-07-27 15:46:20 +00:00
Russell Bryant
f8483a0d04
Do a massive conversion for using the ast_verb() macro
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(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");
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2007-07-26 15:49:18 +00:00
Tilghman Lesher
55b1ee298e
Merge the dialplan_aesthetics branch. Most of this patch simply converts applications
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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
2007-07-23 19:51:41 +00:00
Steve Murphy
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.
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2007-07-19 23:24:27 +00:00
Mark Michelson
ee6d59eef2
Merged revisions 75405 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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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
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2007-07-17 20:05:19 +00:00
Steve Murphy
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.
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2007-07-17 19:40:29 +00:00
Jason Parker
766121a5bc
Fix an incorrect parenthesization (TODO: Find a better word) in app_dial
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Pointed out by Fanzhou Zhao
Closes issue #10216
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2007-07-17 12:01:05 +00:00
Mark Michelson
ce8f95d750
Merged revisions 75253 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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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)
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2007-07-16 18:18:19 +00:00
Joshua Colp
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.
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2007-07-16 14:39:29 +00:00
Joshua Colp
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.
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2007-07-16 13:35:20 +00:00
Olle Johansson
a1b9cbcd31
Implementation of a feature that will disable "missed calls" counters on SIP phones.
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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: https://origsvn.digium.com/svn/asterisk/trunk@74024 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-07-09 08:27:37 +00:00
Tilghman Lesher
8b93f50dfc
Merged revisions 73053 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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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
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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 )
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2007-07-03 12:40:26 +00:00
Tilghman Lesher
a1bc823136
Issue 9990 - New API ast_mkdir, which creates parent directories as necessary (and is faster than an outcall to mkdir -p)
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2007-06-22 04:35:12 +00:00
Steve Murphy
2462d5ab4f
Cleaning up a small disaster I created earlier
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2007-06-20 23:26:07 +00:00
Steve Murphy
57526b35cc
As per 9228, now app_queue should have the proper machinery to do gosubs.
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2007-06-20 21:38:49 +00:00
Tilghman Lesher
ce2c52d519
Merged revisions 70445 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r70445 | tilghman | 2007-06-20 14:29:23 -0500 (Wed, 20 Jun 2007) | 10 lines
Merged revisions 70444 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r70444 | tilghman | 2007-06-20 14:25:54 -0500 (Wed, 20 Jun 2007) | 2 lines
Issue 9997 - Timelimit times out the wrong channel
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2007-06-20 19:30:31 +00:00
Tilghman Lesher
704c756c4a
Merge work to make U(...) option work for Dial
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2007-06-20 17:35:08 +00:00
Steve Murphy
866bbaa515
Via bug9228, no way to create macros via AEL, and some of the apps allow you to call macros..., I modded the apps that allow macro calls to allow gosubs calls also, to make them AEL compliant.
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2007-06-19 23:36:34 +00:00
Russell Bryant
055d82cbce
Add a massive set of changes for converting to use the ast_debug() macro.
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(issue #9957 , patches from mvanbaak, caio1982, critch, and dimas)
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2007-06-14 19:39:12 +00:00
Russell Bryant
9e0458e9f1
Completely remove all of the code related to jumping to priority n + 101. yay!
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(issue #9926 , caio1982)
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2007-06-12 15:58:28 +00:00
Joshua Colp
2492bf26fb
Merged revisions 68071 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r68071 | file | 2007-06-07 10:21:59 -0400 (Thu, 07 Jun 2007) | 10 lines
Merged revisions 68070 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r68070 | file | 2007-06-07 10:19:40 -0400 (Thu, 07 Jun 2007) | 2 lines
Allow the 'g' option to work if used with the 'S' option. (issue #9888 reported by gasparz)
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2007-06-07 14:23:21 +00:00
Tilghman Lesher
9d05ff8ed5
Issue 9869 - replace malloc and memset with ast_calloc, and other coding guidelines changes
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@67864 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-06-06 21:20:11 +00:00
Joshua Colp
bae04fd90e
Merged revisions 67066 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r67066 | file | 2007-06-04 13:59:14 -0400 (Mon, 04 Jun 2007) | 2 lines
Initialize cidname variable to nothing since it may be used without having been touched. (issue #9661 reported by dimas)
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2007-06-04 18:00:24 +00:00
Steve Murphy
4572edae31
Merged revisions 65200 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r65200 | murf | 2007-05-18 16:06:27 -0600 (Fri, 18 May 2007) | 9 lines
Merged revisions 65172 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r65172 | murf | 2007-05-18 14:56:20 -0600 (Fri, 18 May 2007) | 1 line
This update will fix the situation that occurs as described by 9717, where when several targets are specified for a dial, if any one them reports FAIL, the whole call gets FAIL, even though others were ringing OK. I rearranged the priorities, so that a new disposition, NULL, is at the lowest level, and the disposition get init'd to NULL. Then, next up is FAIL, and next up is BUSY, then NOANSWER, then ANSWERED. All the related set routines will only do so if the disposition value to be set to is greater than what's already there. This gives the intended effect. So, if all the targets are busy, you'd get BUSY for the call disposition. If all get BUSY, but one, and that one rings is not answered, you get NOANSWER. If by some freak of nature, the NULL value doesn't get overridden, then the disp2str routine will report NOANSWER as before.
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2007-05-18 22:33:51 +00:00
Russell Bryant
83d0b0417c
Merged revisions 64756 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r64756 | russell | 2007-05-17 11:47:29 -0500 (Thu, 17 May 2007) | 3 lines
Increase the size of a buffer to support longer dial strings for channels.
(issue #9291 , reported and fix suggested by meni)
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2007-05-17 16:49:50 +00:00
Joshua Colp
e2871220e2
Merged revisions 61656 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r61656 | file | 2007-04-13 15:17:08 -0400 (Fri, 13 Apr 2007) | 10 lines
Merged revisions 61655 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r61655 | file | 2007-04-13 15:15:12 -0400 (Fri, 13 Apr 2007) | 2 lines
Add OUTBOUND_GROUP_ONCE variable to app_dial. This behaves the same as OUTBOUND_GROUP except it will get unset after use so it won't get accidentally inherited. (issue #BE-140)
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2007-04-13 19:18:46 +00:00
Russell Bryant
be874b92d3
Remove unused instances of unnamed enums.
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2007-04-09 22:49:32 +00:00
Joshua Colp
840f1e61e0
Merged revisions 60798 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r60798 | file | 2007-04-08 21:03:14 -0400 (Sun, 08 Apr 2007) | 10 lines
Merged revisions 60797 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r60797 | file | 2007-04-08 20:59:29 -0400 (Sun, 08 Apr 2007) | 2 lines
When calling a device that then forwards us elsewhere... we have to make our channels compatible if it is the only channel being dialed. (issue #9445 reported by marcelbarbulescu)
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2007-04-09 01:06:56 +00:00
Joshua Colp
726928cb60
Properly hangup the original dialed channel, not the new channel that appeared from the forwarding. (issue #9161 reported by PhilSmith)
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2007-02-27 22:17:42 +00:00
Joshua Colp
fb9e48ca9e
Merged revisions 55154 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r55154 | file | 2007-02-16 22:55:30 -0500 (Fri, 16 Feb 2007) | 10 lines
Merged revisions 55153 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r55153 | file | 2007-02-16 22:53:45 -0500 (Fri, 16 Feb 2007) | 2 lines
Answer the channel before recording privacy information. (issue #8926 reported by lmamane)
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2007-02-17 03:57:23 +00:00
Joshua Colp
8e92c73a1d
Merged revisions 54924 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r54924 | file | 2007-02-16 13:51:15 -0500 (Fri, 16 Feb 2007) | 2 lines
Need to check macro extension as well as macro context for directed pickup.
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2007-02-16 18:53:17 +00:00
Joshua Colp
821c941976
Merged revisions 54884 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r54884 | file | 2007-02-16 12:02:35 -0500 (Fri, 16 Feb 2007) | 2 lines
Allow directed pickup to pick up the real context instead of the macro context if a Macro is used. (issue #8984 reported by jamesb63)
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2007-02-16 17:07:10 +00:00
Joshua Colp
a11b56a8e5
Merged revisions 54623 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r54623 | file | 2007-02-15 11:19:39 -0500 (Thu, 15 Feb 2007) | 10 lines
Merged revisions 54622 via svnmerge from
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r54622 | file | 2007-02-15 11:14:40 -0500 (Thu, 15 Feb 2007) | 2 lines
Use a separate variable to indicate execution should continue instead of the return value. (issue #8842 reported by pluto70)
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2007-02-15 16:24:13 +00:00
Joshua Colp
68a66656e6
Merged revisions 54481 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r54481 | file | 2007-02-14 16:07:23 -0500 (Wed, 14 Feb 2007) | 2 lines
Forward begin DTMF frames as well as end. (issue #9068 reported by mhardeman)
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2007-02-14 21:10:53 +00:00
Joshua Colp
32cd307d6f
Merged revisions 53749 via svnmerge from
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r53749 | file | 2007-02-09 14:33:31 -0500 (Fri, 09 Feb 2007) | 2 lines
Temporarily change musicclass on channel to one specified in Dial so that the 'm' option functions properly. (issue #8969 reported by christianbee)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@53750 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-02-09 19:39:26 +00:00
Russell Bryant
ce321f87e9
Merged revisions 53136 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r53136 | russell | 2007-02-03 14:44:20 -0600 (Sat, 03 Feb 2007) | 12 lines
Merged revisions 53133 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r53133 | russell | 2007-02-03 14:38:13 -0600 (Sat, 03 Feb 2007) | 4 lines
set the DIALSTATUS variable to contain "INVALIDARGS" when the dial application
exits early because of invalid arguments instead of just leaving it empty.
(issue #8975 )
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@53137 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-02-03 20:46:36 +00:00
Joshua Colp
a378353ecc
Merged revisions 50298 via svnmerge from
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https://origsvn.digium.com/svn/asterisk/branches/1.4
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r50298 | file | 2007-01-09 23:55:13 -0500 (Tue, 09 Jan 2007) | 10 lines
Merged revisions 50295 via svnmerge from
https://origsvn.digium.com/svn/asterisk/branches/1.2
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r50295 | file | 2007-01-09 23:51:06 -0500 (Tue, 09 Jan 2007) | 2 lines
Add another return value to dial_exec_full that indicates execution is going to continuing at a new extension/context/priority and to just let it slide. (issue #8598 reported by jon)
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@50302 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2007-01-10 04:56:48 +00:00
Luigi Rizzo
83f52ed5e2
better name for struct dial_localuser.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@48581 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-12-19 16:36:45 +00:00
Luigi Rizzo
a755ec928a
introduce a temporary variable for tmp->chan to shorten expressions.
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git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@48575 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-12-19 09:58:40 +00:00
Luigi Rizzo
9c81431c93
stop what i think is a memory leak in case Dial fails to
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connect to a channel.
Before committing to 1.4 i would like some other people to
review and test this fix - thanks.
git-svn-id: https://origsvn.digium.com/svn/asterisk/trunk@48574 65c4cc65-6c06-0410-ace0-fbb531ad65f3
2006-12-19 09:33:57 +00:00